Initial code drop of WebRTC r1804
diff --git a/.gitignore b/.gitignore
new file mode 100644
index 0000000..90ec22b
--- /dev/null
+++ b/.gitignore
@@ -0,0 +1 @@
+.svn
diff --git a/trunk/.gitignore b/trunk/.gitignore
new file mode 100644
index 0000000..c27b166
--- /dev/null
+++ b/trunk/.gitignore
@@ -0,0 +1,65 @@
+*.target.mk
+*.Makefile
+*.ncb
+*.ninja
+*.props
+*.pyc
+*.rules
+*.scons
+*.sdf
+*.sln
+*.suo
+*.targets
+*.user
+*.vcproj
+*.vcxproj
+*.vcxproj.filters
+*.vpj
+*.vpw
+*.vpwhistu
+*.vtg
+*.xcodeproj
+*~
+.*.sw?
+.DS_Store
+.cproject
+.gdb_history
+.gdbinit
+.metadata
+.project
+.pydevproject
+.settings
+/build
+/chromium_deps
+/gyp-mac-tool
+/Makefile
+/out
+/resources
+/src/supplement.gypi
+/testing
+/third_party/asan
+/third_party/cygwin
+/third_party/expat
+/third_party/gaeunit
+/third_party/gold
+/third_party/google-gflags/src
+/third_party/google-visualization-python
+/third_party/jsoncpp
+/third_party/libjingle
+/third_party/libjpeg
+/third_party/libjpeg_turbo
+/third_party/libsrtp
+/third_party/libvpx
+/third_party/libyuv
+/third_party/llvm-build
+/third_party/oauth2
+/third_party/protobuf
+/third_party/valgrind
+/third_party/yasm
+/tools/clang
+/tools/gyp
+/tools/python
+/tools/valgrind
+/tools/win
+/x86-generic_out/
+/xcodebuild
diff --git a/trunk/AUTHORS b/trunk/AUTHORS
new file mode 100644
index 0000000..c3d75d6
--- /dev/null
+++ b/trunk/AUTHORS
@@ -0,0 +1,6 @@
+# Names should be added to this file like so:
+# Name or Organization <email address>
+
+Google Inc.
+Mozilla Foundation
+Ben Strong <bstrong@gmail.com>
\ No newline at end of file
diff --git a/trunk/Android.mk b/trunk/Android.mk
new file mode 100644
index 0000000..b7c28e6
--- /dev/null
+++ b/trunk/Android.mk
@@ -0,0 +1,167 @@
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+MY_WEBRTC_ROOT_PATH := $(call my-dir)
+
+# voice
+include $(MY_WEBRTC_ROOT_PATH)/src/common_audio/resampler/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/common_audio/signal_processing/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/common_audio/vad/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_coding/neteq/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_coding/codecs/cng/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_coding/codecs/g711/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_coding/codecs/g722/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_coding/codecs/pcm16b/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_coding/codecs/ilbc/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_coding/codecs/iSAC/fix/source/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_coding/codecs/iSAC/main/source/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_coding/main/source/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_conference_mixer/source/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_device/main/source/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_processing/aec/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_processing/aecm/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_processing/agc/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_processing/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_processing/ns/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_processing/utility/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/media_file/source/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/rtp_rtcp/source/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/udp_transport/source/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/utility/source/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/system_wrappers/source/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/voice_engine/main/source/Android.mk
+
+# video
+include $(MY_WEBRTC_ROOT_PATH)/src/common_video/jpeg/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/common_video/libyuv/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/video_capture/main/source/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/video_coding/codecs/i420/main/source/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/video_coding/codecs/vp8/main/source/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/video_coding/main/source/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/video_processing/main/source/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/modules/video_render/main/source/Android.mk
+include $(MY_WEBRTC_ROOT_PATH)/src/video_engine/Android.mk
+
+# third party
+include $(MY_WEBRTC_ROOT_PATH)/libvpx.mk
+
+# build .so
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+include $(LOCAL_PATH)/../../external/webrtc/android-webrtc.mk
+
+LOCAL_ARM_MODE := arm
+LOCAL_MODULE := libwebrtc_audio_preprocessing
+LOCAL_MODULE_TAGS := optional
+
+LOCAL_WHOLE_STATIC_LIBRARIES := \
+    libwebrtc_spl \
+    libwebrtc_resampler \
+    libwebrtc_apm \
+    libwebrtc_apm_utility \
+    libwebrtc_vad \
+    libwebrtc_ns \
+    libwebrtc_agc \
+    libwebrtc_aec \
+    libwebrtc_aecm \
+    libwebrtc_system_wrappers
+
+# Add Neon libraries.
+ifeq ($(WEBRTC_BUILD_NEON_LIBS),true)
+LOCAL_WHOLE_STATIC_LIBRARIES += \
+    libwebrtc_aecm_neon \
+    libwebrtc_ns_neon
+endif
+
+LOCAL_STATIC_LIBRARIES := \
+    libprotobuf-cpp-2.3.0-lite
+
+LOCAL_SHARED_LIBRARIES := \
+    libcutils \
+    libdl \
+    libstlport
+
+LOCAL_PRELINK_MODULE := false
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_SHARED_LIBRARY)
+
+###
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+LOCAL_ARM_MODE := arm
+LOCAL_MODULE := libwebrtc
+LOCAL_MODULE_TAGS := optional
+
+LOCAL_WHOLE_STATIC_LIBRARIES := \
+    libwebrtc_system_wrappers \
+    libwebrtc_audio_device \
+    libwebrtc_pcm16b \
+    libwebrtc_cng \
+    libwebrtc_audio_coding \
+    libwebrtc_rtp_rtcp \
+    libwebrtc_media_file \
+    libwebrtc_udp_transport \
+    libwebrtc_utility \
+    libwebrtc_neteq \
+    libwebrtc_audio_conference_mixer \
+    libwebrtc_isac \
+    libwebrtc_ilbc \
+    libwebrtc_isacfix \
+    libwebrtc_g722 \
+    libwebrtc_g711 \
+    libwebrtc_voe_core \
+    libwebrtc_video_render \
+    libwebrtc_video_capture \
+    libwebrtc_i420 \
+    libwebrtc_video_coding \
+    libwebrtc_video_processing \
+    libwebrtc_vp8 \
+    libwebrtc_vie_core \
+    libwebrtc_yuv \
+    libwebrtc_jpeg \
+    libwebrtc_vpx
+
+# Add Neon libraries.
+ifeq ($(WEBRTC_BUILD_NEON_LIBS),true)
+LOCAL_WHOLE_STATIC_LIBRARIES += \
+    libwebrtc_isacfix_neon
+endif
+
+LOCAL_STATIC_LIBRARIES := \
+    libyuv_static
+
+LOCAL_SHARED_LIBRARIES := \
+    libcutils \
+    libdl \
+    libstlport \
+    libjpeg \
+    libGLESv2 \
+    libOpenSLES \
+    libwebrtc_audio_preprocessing
+
+LOCAL_PRELINK_MODULE := false
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_SHARED_LIBRARY)
+
+# test apps, they're for test only; all these test apps have LOCAL_MODULE_TAGS:=tests
+# voice engine test apps
+include $(MY_WEBRTC_ROOT_PATH)/src/voice_engine/main/test/cmd_test/Android.mk
+#include $(MY_WEBRTC_ROOT_PATH)/src/voice_engine/main/test/auto_test/Android.mk
+# video engien test apps
+include $(MY_WEBRTC_ROOT_PATH)/src/video_engine/main/test/android_test/Android.mk
+#include $(MY_WEBRTC_ROOT_PATH)/src/video_engine/test/auto_test/android/Android.mk
diff --git a/trunk/DEPS b/trunk/DEPS
new file mode 100644
index 0000000..0c8892e
--- /dev/null
+++ b/trunk/DEPS
@@ -0,0 +1,130 @@
+vars = {
+  # Use this googlecode_url variable only if there is an internal mirror for it.
+  # If you do not know, use the full path while defining your new deps entry.
+  "googlecode_url": "http://%s.googlecode.com/svn",
+  "chromium_trunk" : "http://src.chromium.org/svn/trunk",
+  "chromium_revision": "122775",
+
+  # External resources like video and audio files used for testing purposes.
+  # Downloaded on demand when needed.
+  "webrtc_resources_revision": "8",
+}
+
+# NOTE: Prefer revision numbers to tags for svn deps. Use http rather than
+# https; the latter can cause problems for users behind proxies.
+deps = {
+  "trunk/chromium_deps":
+    File(Var("chromium_trunk") + "/src/DEPS@" + Var("chromium_revision")),
+
+  "trunk/build":
+    Var("chromium_trunk") + "/src/build@" + Var("chromium_revision"),
+
+  "trunk/testing":
+    Var("chromium_trunk") + "/src/testing@" + Var("chromium_revision"),
+
+  "trunk/testing/gmock":
+    From("trunk/chromium_deps", "src/testing/gmock"),
+
+  "trunk/testing/gtest":
+    From("trunk/chromium_deps", "src/testing/gtest"),
+
+  "trunk/third_party/expat":
+    Var("chromium_trunk") + "/src/third_party/expat@" + Var("chromium_revision"),
+
+  # Used by tools/quality_tracking.
+  "trunk/third_party/gaeunit":
+    "http://code.google.com/p/gaeunit.git@e16d5bd4",
+
+  "trunk/third_party/google-gflags/src":
+    (Var("googlecode_url") % "google-gflags") + "/trunk/src@45",
+
+  # Used by tools/quality_tracking/dashboard and tools/python_charts.
+  "trunk/third_party/google-visualization-python":
+    (Var("googlecode_url") % "google-visualization-python") + "/trunk@15",
+
+  "trunk/third_party/libjpeg":
+    Var("chromium_trunk") + "/src/third_party/libjpeg@" + Var("chromium_revision"),
+
+  "trunk/third_party/libjpeg_turbo":
+    From("trunk/chromium_deps", "src/third_party/libjpeg_turbo"),
+
+  "trunk/third_party/libvpx/source/libvpx":
+    "http://git.chromium.org/webm/libvpx.git@v1.0.0",
+
+  "trunk/third_party/libyuv":
+    (Var("googlecode_url") % "libyuv") + "/trunk@190",
+
+  "trunk/third_party/protobuf":
+    Var("chromium_trunk") + "/src/third_party/protobuf@" + Var("chromium_revision"),
+
+  # Used by tools/quality_tracking.
+  "trunk/third_party/oauth2":
+    "http://github.com/simplegeo/python-oauth2.git@a83f4a29",
+
+  "trunk/third_party/yasm":
+    Var("chromium_trunk") + "/src/third_party/yasm@" + Var("chromium_revision"),
+
+  "trunk/third_party/yasm/source/patched-yasm":
+    From("trunk/chromium_deps", "src/third_party/yasm/source/patched-yasm"),
+
+  "trunk/tools/clang":
+    Var("chromium_trunk") + "/src/tools/clang@" + Var("chromium_revision"),
+
+  "trunk/tools/gyp":
+    From("trunk/chromium_deps", "src/tools/gyp"),
+
+  "trunk/tools/python":
+    Var("chromium_trunk") + "/src/tools/python@" + Var("chromium_revision"),
+
+  "trunk/tools/valgrind":
+    Var("chromium_trunk") + "/src/tools/valgrind@" + Var("chromium_revision"),
+
+  # Needed by build/common.gypi.
+  "trunk/tools/win/supalink":
+    Var("chromium_trunk") + "/src/tools/win/supalink@" + Var("chromium_revision"),
+}
+
+deps_os = {
+  "win": {
+    "trunk/third_party/cygwin":
+      Var("chromium_trunk") + "/deps/third_party/cygwin@66844",
+
+    # Used by libjpeg-turbo.
+    "trunk/third_party/yasm/binaries":
+      From("trunk/chromium_deps", "src/third_party/yasm/binaries"),
+  },
+  "unix": {
+    "trunk/third_party/gold":
+      From("trunk/chromium_deps", "src/third_party/gold"),
+  },
+}
+
+hooks = [
+  {
+    # Create a supplement.gypi file under trunk/.  This file will be picked up
+    # by gyp and we use it to set Chromium related variables (inside_chromium_build)
+    # to 0 and enable the standalone build.
+    "pattern": ".",
+    "action": ["python", "trunk/tools/create_supplement_gypi.py", "trunk/src/supplement.gypi"],
+  },
+  {
+    # Pull clang on mac. If nothing changed, or on non-mac platforms, this takes
+    # zero seconds to run. If something changed, it downloads a prebuilt clang.
+    "pattern": ".",
+    "action": ["python", "trunk/tools/clang/scripts/update.py", "--mac-only"],
+  },
+  {
+    # Download test resources, i.e. video and audio files. If the latest
+    # version is already downloaded, this takes zero seconds to run.
+    # If a newer version or no current download exists, it will download
+    # the resources and extract them.
+    "pattern": ".",
+    "action": ["python", "trunk/tools/resources/update.py"],
+  },
+  {
+    # A change to a .gyp, .gypi, or to GYP itself should run the generator.
+    "pattern": ".",
+    "action": ["python", "trunk/build/gyp_chromium", "--depth=trunk", "trunk/webrtc.gyp"],
+  },
+]
+
diff --git a/trunk/LICENSE b/trunk/LICENSE
new file mode 100644
index 0000000..dd4a345
--- /dev/null
+++ b/trunk/LICENSE
@@ -0,0 +1 @@
+Refer to src/LICENSE.
diff --git a/trunk/LICENSE_THIRD_PARTY b/trunk/LICENSE_THIRD_PARTY
new file mode 100644
index 0000000..d47c055
--- /dev/null
+++ b/trunk/LICENSE_THIRD_PARTY
@@ -0,0 +1 @@
+Refer to src/LICENSE_THIRD_PARTY.
diff --git a/trunk/OWNERS b/trunk/OWNERS
new file mode 100644
index 0000000..b110a52
--- /dev/null
+++ b/trunk/OWNERS
@@ -0,0 +1,5 @@
+henrika@webrtc.org

+niklas.enbom@webrtc.org

+andrew@webrtc.org

+tina.legrand@webrtc.org

+tommi@webrtc.org
\ No newline at end of file
diff --git a/trunk/PATENTS b/trunk/PATENTS
new file mode 100644
index 0000000..5cb83ec
--- /dev/null
+++ b/trunk/PATENTS
@@ -0,0 +1 @@
+Refer to src/PATENTS.
diff --git a/trunk/PRESUBMIT.py b/trunk/PRESUBMIT.py
new file mode 100644
index 0000000..56c8d5b
--- /dev/null
+++ b/trunk/PRESUBMIT.py
@@ -0,0 +1,144 @@
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+def _LicenseHeader(input_api):
+  """Returns the license header regexp."""
+  license_header = (
+      r'.*? Copyright \(c\) %(year)s The WebRTC project authors\. '
+        r'All Rights Reserved\.\n'
+      r'.*?\n'
+      r'.*? Use of this source code is governed by a BSD-style license\n'
+      r'.*? that can be found in the LICENSE file in the root of the source\n'
+      r'.*? tree\. An additional intellectual property rights grant can be '
+        r'found\n'
+      r'.*? in the file PATENTS\.  All contributing project authors may\n'
+      r'.*? be found in the AUTHORS file in the root of the source tree\.\n'
+  ) % {
+      'year': input_api.time.strftime('%Y'),
+  }
+  return license_header
+
+def _CheckNoIOStreamInHeaders(input_api, output_api):
+  """Checks to make sure no .h files include <iostream>."""
+  files = []
+  pattern = input_api.re.compile(r'^#include\s*<iostream>',
+                                 input_api.re.MULTILINE)
+  for f in input_api.AffectedSourceFiles(input_api.FilterSourceFile):
+    if not f.LocalPath().endswith('.h'):
+      continue
+    contents = input_api.ReadFile(f)
+    if pattern.search(contents):
+      files.append(f)
+
+  if len(files):
+    return [ output_api.PresubmitError(
+        'Do not #include <iostream> in header files, since it inserts static ' +
+        'initialization into every file including the header. Instead, ' +
+        '#include <ostream>. See http://crbug.com/94794',
+        files) ]
+  return []
+
+def _CheckNoFRIEND_TEST(input_api, output_api):
+  """Make sure that gtest's FRIEND_TEST() macro is not used, the
+  FRIEND_TEST_ALL_PREFIXES() macro from testsupport/gtest_prod_util.h should be
+  used instead since that allows for FLAKY_, FAILS_ and DISABLED_ prefixes."""
+  problems = []
+
+  file_filter = lambda f: f.LocalPath().endswith(('.cc', '.h'))
+  for f in input_api.AffectedFiles(file_filter=file_filter):
+    for line_num, line in f.ChangedContents():
+      if 'FRIEND_TEST(' in line:
+        problems.append('    %s:%d' % (f.LocalPath(), line_num))
+
+  if not problems:
+    return []
+  return [output_api.PresubmitPromptWarning('WebRTC\'s code should not use '
+      'gtest\'s FRIEND_TEST() macro. Include testsupport/gtest_prod_util.h and '
+      'use FRIEND_TEST_ALL_PREFIXES() instead.\n' + '\n'.join(problems))]
+
+def _CheckNewFilesLintClean(input_api, output_api, source_file_filter=None):
+  """Checks that all NEW '.cc' and '.h' files pass cpplint.py.
+  This check is based on _CheckChangeLintsClean in
+  depot_tools/presubmit_canned_checks.py but has less filters and only checks
+  added files."""
+  result = []
+
+  # Initialize cpplint.
+  import cpplint
+  # Access to a protected member _XX of a client class
+  # pylint: disable=W0212
+  cpplint._cpplint_state.ResetErrorCounts()
+
+  # Justifications for each filter:
+  #
+  # - build/header_guard  : WebRTC coding style says they should be prefixed
+  #                         with WEBRTC_, which is not possible to configure in
+  #                         cpplint.py.
+  cpplint._SetFilters('-build/header_guard')
+
+  # Use the strictest verbosity level for cpplint.py (level 1) which is the
+  # default when running cpplint.py from command line.
+  # To make it possible to work with not-yet-converted code, we're only applying
+  # it to new (or moved/renamed) files.
+  verbosity_level = 1
+  files = []
+  for f in input_api.AffectedSourceFiles(source_file_filter):
+    # Note that moved/renamed files also count as added for svn.
+    if (f.Action() == 'A'):
+      files.append(f.AbsoluteLocalPath())
+  for file_name in files:
+    cpplint.ProcessFile(file_name, verbosity_level)
+
+  if cpplint._cpplint_state.error_count > 0:
+    if input_api.is_committing:
+      # TODO(kjellander): Change back to PresubmitError below when we're
+      # confident with the lint settings.
+      res_type = output_api.PresubmitPromptWarning
+    else:
+      res_type = output_api.PresubmitPromptWarning
+    result = [res_type('Changelist failed cpplint.py check.')]
+
+  return result
+
+def _CommonChecks(input_api, output_api):
+  """Checks common to both upload and commit."""
+  # TODO(kjellander): Use presubmit_canned_checks.PanProjectChecks too.
+  results = []
+  results.extend(input_api.canned_checks.CheckLongLines(
+      input_api, output_api))
+  results.extend(input_api.canned_checks.CheckChangeHasNoTabs(
+      input_api, output_api))
+  results.extend(input_api.canned_checks.CheckChangeHasNoStrayWhitespace(
+      input_api, output_api))
+  results.extend(input_api.canned_checks.CheckChangeTodoHasOwner(
+      input_api, output_api))
+  results.extend(_CheckNewFilesLintClean(input_api, output_api))
+  results.extend(input_api.canned_checks.CheckLicense(
+      input_api, output_api, _LicenseHeader(input_api)))
+  results.extend(_CheckNoIOStreamInHeaders(input_api, output_api))
+  results.extend(_CheckNoFRIEND_TEST(input_api, output_api))
+  return results
+
+def CheckChangeOnUpload(input_api, output_api):
+  results = []
+  results.extend(_CommonChecks(input_api, output_api))
+  return results
+
+def CheckChangeOnCommit(input_api, output_api):
+  results = []
+  results.extend(_CommonChecks(input_api, output_api))
+  results.extend(input_api.canned_checks.CheckOwners(input_api, output_api))
+  results.extend(input_api.canned_checks.CheckChangeWasUploaded(
+      input_api, output_api))
+  results.extend(input_api.canned_checks.CheckChangeHasDescription(
+      input_api, output_api))
+  results.extend(input_api.canned_checks.CheckChangeHasBugField(
+      input_api, output_api))
+  results.extend(input_api.canned_checks.CheckChangeHasTestField(
+      input_api, output_api))
+  return results
diff --git a/trunk/WATCHLISTS b/trunk/WATCHLISTS
new file mode 100644
index 0000000..1b592f5
--- /dev/null
+++ b/trunk/WATCHLISTS
@@ -0,0 +1,112 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+# Inspired by chromium.org:
+# http://dev.chromium.org/developers/contributing-code/watchlists
+
+{
+  'WATCHLIST_DEFINITIONS': {
+    'this_file': {
+      'filepath': '^WATCHLISTS$',
+    },
+    'all_src': {
+      'filepath': 'src/.*',
+    },
+    'root_files': {
+      # src/build/ and non-recursive contents of ./ and src/
+      'filepath': '^[^/]*$|src/[^/]*$|src/build/.*',
+    },
+    'documented_interfaces': {
+      'filepath': 'src/[^/]*\.h$|'\
+                  'src/video_engine/main/interface/.*|'\
+                  'src/voice_engine/main/interface/.*',
+    },
+    'build_files': {
+      'filepath': '\.gyp$|\.gypi$|Android\.mk$',
+    },
+    'java_files': {
+      'filepath': '\.java$|\.xml$',
+    },
+    'video_engine': {
+      'filepath': 'src/video_engine/.*',
+    },
+    'voice_engine': {
+      'filepath': 'src/voice_engine/.*',
+    },
+    'common_audio': {
+      'filepath': 'src/common_audio/.*',
+    },
+    'video_capture': {
+      'filepath': 'src/modules/video_capture/.*',
+    },
+    'video_render': {
+      'filepath': 'src/modules/video_render/.*',
+    },
+    'audio_device': {
+      'filepath': 'src/modules/audio_device/.*',
+    },
+    'audio_coding': {
+      'filepath': 'src/modules/audio_coding/.*',
+    },
+    'neteq': {
+      'filepath': 'src/modules/audio_coding/neteq/.*',
+    },
+    'audio_processing': {
+      'filepath': 'src/modules/audio_processing/.*',
+    },
+    'video_codecs': {
+      'filepath': 'src/modules/video_coding/codecs/.*',
+    },
+    'video_coding': {
+      'filepath': 'src/modules/video_coding/.*',
+    },
+    'rtp_rtcp': {
+      'filepath': 'src/modules/rtp_rtcp/.*'
+    },
+    'system_wrappers': {
+      'filepath': 'src/system_wrappers/.*',
+    },
+  },
+
+  'WATCHLISTS': {
+    'this_file': [''],
+    'all_src': ['tterriberry@mozilla.com',
+                'giles@mozilla.com'],
+    'root_files': ['andrew@webrtc.org',
+                   'niklas.enbom@webrtc.org'],
+    'documented_interfaces': ['interface-changes@webrtc.org',
+                              'rwolff@gocast.it'],
+    'build_files': ['leozwang@webrtc.org'],
+    'java_files': ['leozwang@webrtc.org'],
+    'common_audio': ['bjornv@webrtc.org',
+                     'andrew@webrtc.org'],
+    'video_engine': ['mflodman@webrtc.org',
+                     'perkj@webrtc.org'],
+    'voice_engine': ['henrika@webrtc.org'],
+    'video_capture': ['mflodman@webrtc.org',
+                      'perkj@webrtc.org',
+                      'leozwang@webrtc.org'],
+    'video_render': ['mflodman@webrtc.org',
+                     'perkj@webrtc.org',
+                     'leozwang@webrtc.org'],
+    'audio_device': ['henrika@webrtc.org',
+                     'leozwang@webrtc.org'],
+    'audio_coding': ['tina.legrand@webrtc.org'],
+    'neteq': ['henrik.lundin@webrtc.org'],
+    'audio_processing': ['andrew@webrtc.org',
+                         'bjornv@webrtc.org'],
+    'video_codecs': ['henrik.lundin@webrtc.org',
+                     'pwestin@webrtc.org'],
+    'video_coding': ['stefan@webrtc.org'],
+    'rtp_rtcp': ['mflodman@webrtc.org',
+                 'pwestin@webrtc.org'],
+    'system_wrappers': ['mflodman@webrtc.org',
+                        'henrika@webrtc.org',
+                        'andrew@webrtc.org'],
+  },
+}
diff --git a/trunk/android-webrtc.mk b/trunk/android-webrtc.mk
new file mode 100644
index 0000000..cd495b0
--- /dev/null
+++ b/trunk/android-webrtc.mk
@@ -0,0 +1,46 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+# These defines will apply to all source files
+# Think again before changing it
+MY_WEBRTC_COMMON_DEFS := \
+    '-DWEBRTC_TARGET_PC' \
+    '-DWEBRTC_LINUX' \
+    '-DWEBRTC_THREAD_RR' \
+    '-DWEBRTC_CLOCK_TYPE_REALTIME' \
+    '-DWEBRTC_ANDROID'
+#    The following macros are used by modules,
+#    we might need to re-organize them
+#    '-DWEBRTC_ANDROID_OPENSLES' [module audio_device]
+#    '-DNETEQ_VOICEENGINE_CODECS' [module audio_coding neteq]
+#    '-DWEBRTC_MODULE_UTILITY_VIDEO' [module media_file] [module utility]
+ifeq ($(TARGET_ARCH),arm)
+MY_WEBRTC_COMMON_DEFS += \
+    '-DWEBRTC_ARCH_ARM'
+#    '-DWEBRTC_DETECT_ARM_NEON' # only used in a build configuration without Neon
+# TODO(kma): figure out if the above define could be moved to NDK build only.
+
+# TODO(kma): test if the code under next two macros works with generic GCC compilers
+ifeq ($(ARCH_ARM_HAVE_NEON),true)
+MY_WEBRTC_COMMON_DEFS += \
+    '-DWEBRTC_ARCH_ARM_NEON'
+MY_ARM_CFLAGS_NEON := \
+    -flax-vector-conversions
+endif
+
+ifneq (,$(filter '-DWEBRTC_DETECT_ARM_NEON' '-DWEBRTC_ARCH_ARM_NEON', \
+    $(MY_WEBRTC_COMMON_DEFS)))
+WEBRTC_BUILD_NEON_LIBS := true
+endif
+
+ifeq ($(ARCH_ARM_HAVE_ARMV7A),true)
+MY_WEBRTC_COMMON_DEFS += \
+    '-DWEBRTC_ARCH_ARM_V7A'
+endif
+
+endif # ifeq ($(TARGET_ARCH),arm)
\ No newline at end of file
diff --git a/trunk/codereview.settings b/trunk/codereview.settings
new file mode 100644
index 0000000..b70057e
--- /dev/null
+++ b/trunk/codereview.settings
@@ -0,0 +1,9 @@
+# This file is used by gcl to get repository specific information.

+CODE_REVIEW_SERVER: webrtc-codereview.appspot.com

+#CC_LIST:

+#VIEW_VC:

+#STATUS: 

+TRY_ON_UPLOAD: False

+#TRYSERVER_SVN_URL: 

+#GITCL_PREUPLOAD: 

+#GITCL_PREDCOMMIT: 

diff --git a/trunk/libvpx.mk b/trunk/libvpx.mk
new file mode 100644
index 0000000..07c04dc
--- /dev/null
+++ b/trunk/libvpx.mk
@@ -0,0 +1,107 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+LOCAL_PATH := $(call my-dir)
+include $(CLEAR_VARS)
+
+MY_LIBVPX_PATH = ../libvpx
+
+LOCAL_SRC_FILES = \
+     $(MY_LIBVPX_PATH)/vp8/common/alloccommon.c \
+     $(MY_LIBVPX_PATH)/vp8/common/blockd.c \
+     $(MY_LIBVPX_PATH)/vp8/common/debugmodes.c \
+     $(MY_LIBVPX_PATH)/vp8/common/entropy.c \
+     $(MY_LIBVPX_PATH)/vp8/common/entropymode.c \
+     $(MY_LIBVPX_PATH)/vp8/common/entropymv.c \
+     $(MY_LIBVPX_PATH)/vp8/common/extend.c \
+     $(MY_LIBVPX_PATH)/vp8/common/filter.c \
+     $(MY_LIBVPX_PATH)/vp8/common/findnearmv.c \
+     $(MY_LIBVPX_PATH)/vp8/common/generic/systemdependent.c \
+     $(MY_LIBVPX_PATH)/vp8/common/idctllm.c \
+     $(MY_LIBVPX_PATH)/vp8/common/invtrans.c \
+     $(MY_LIBVPX_PATH)/vp8/common/loopfilter.c \
+     $(MY_LIBVPX_PATH)/vp8/common/loopfilter_filters.c \
+     $(MY_LIBVPX_PATH)/vp8/common/mbpitch.c \
+     $(MY_LIBVPX_PATH)/vp8/common/modecont.c \
+     $(MY_LIBVPX_PATH)/vp8/common/modecontext.c \
+     $(MY_LIBVPX_PATH)/vp8/common/quant_common.c \
+     $(MY_LIBVPX_PATH)/vp8/common/recon.c \
+     $(MY_LIBVPX_PATH)/vp8/common/reconinter.c \
+     $(MY_LIBVPX_PATH)/vp8/common/reconintra.c \
+     $(MY_LIBVPX_PATH)/vp8/common/reconintra4x4.c \
+     $(MY_LIBVPX_PATH)/vp8/common/setupintrarecon.c \
+     $(MY_LIBVPX_PATH)/vp8/common/swapyv12buffer.c \
+     $(MY_LIBVPX_PATH)/vp8/common/textblit.c \
+     $(MY_LIBVPX_PATH)/vp8/common/treecoder.c \
+     $(MY_LIBVPX_PATH)/vp8/vp8_cx_iface.c \
+     $(MY_LIBVPX_PATH)/vp8/vp8_dx_iface.c \
+     $(MY_LIBVPX_PATH)/vpx_config.c \
+     $(MY_LIBVPX_PATH)/vpx/src/vpx_codec.c \
+     $(MY_LIBVPX_PATH)/vpx/src/vpx_decoder.c \
+     $(MY_LIBVPX_PATH)/vpx/src/vpx_image.c \
+     $(MY_LIBVPX_PATH)/vpx_mem/vpx_mem.c \
+     $(MY_LIBVPX_PATH)/vpx_scale/generic/vpxscale.c \
+     $(MY_LIBVPX_PATH)/vpx_scale/generic/yv12config.c \
+     $(MY_LIBVPX_PATH)/vpx_scale/generic/yv12extend.c \
+     $(MY_LIBVPX_PATH)/vpx_scale/generic/gen_scalers.c \
+     $(MY_LIBVPX_PATH)/vpx_scale/generic/scalesystemdependent.c \
+     $(MY_LIBVPX_PATH)/vpx/src/vpx_encoder.c \
+     $(MY_LIBVPX_PATH)/vp8/encoder/bitstream.c \
+     $(MY_LIBVPX_PATH)/vp8/encoder/boolhuff.c \
+     $(MY_LIBVPX_PATH)/vp8/encoder/dct.c \
+     $(MY_LIBVPX_PATH)/vp8/encoder/encodeframe.c \
+     $(MY_LIBVPX_PATH)/vp8/encoder/encodeintra.c \
+     $(MY_LIBVPX_PATH)/vp8/encoder/encodemb.c \
+     $(MY_LIBVPX_PATH)/vp8/encoder/encodemv.c \
+     $(MY_LIBVPX_PATH)/vp8/encoder/ethreading.c \
+     $(MY_LIBVPX_PATH)/vp8/encoder/firstpass.c \
+     $(MY_LIBVPX_PATH)/vp8/encoder/generic/csystemdependent.c \
+     $(MY_LIBVPX_PATH)/vp8/encoder/mcomp.c \
+     $(MY_LIBVPX_PATH)/vp8/encoder/modecosts.c \
+     $(MY_LIBVPX_PATH)/vp8/encoder/pickinter.c \
+     $(MY_LIBVPX_PATH)/vp8/encoder/picklpf.c \
+     $(MY_LIBVPX_PATH)/vp8/encoder/psnr.c \
+     $(MY_LIBVPX_PATH)/vp8/encoder/quantize.c \
+     $(MY_LIBVPX_PATH)/vp8/encoder/ratectrl.c \
+     $(MY_LIBVPX_PATH)/vp8/encoder/rdopt.c \
+     $(MY_LIBVPX_PATH)/vp8/encoder/sad_c.c \
+     $(MY_LIBVPX_PATH)/vp8/encoder/segmentation.c \
+     $(MY_LIBVPX_PATH)/vp8/encoder/tokenize.c \
+     $(MY_LIBVPX_PATH)/vp8/encoder/treewriter.c \
+     $(MY_LIBVPX_PATH)/vp8/encoder/onyx_if.c \
+     $(MY_LIBVPX_PATH)/vp8/encoder/temporal_filter.c \
+     $(MY_LIBVPX_PATH)/vp8/encoder/variance_c.c \
+     $(MY_LIBVPX_PATH)/vp8/decoder/dboolhuff.c \
+     $(MY_LIBVPX_PATH)/vp8/decoder/decodemv.c \
+     $(MY_LIBVPX_PATH)/vp8/decoder/decodframe.c \
+     $(MY_LIBVPX_PATH)/vp8/decoder/dequantize.c \
+     $(MY_LIBVPX_PATH)/vp8/decoder/detokenize.c \
+     $(MY_LIBVPX_PATH)/vp8/decoder/generic/dsystemdependent.c \
+     $(MY_LIBVPX_PATH)/vp8/decoder/onyxd_if.c \
+     $(MY_LIBVPX_PATH)/vp8/decoder/reconintra_mt.c \
+     $(MY_LIBVPX_PATH)/vp8/decoder/threading.c \
+     $(MY_LIBVPX_PATH)/vp8/decoder/idct_blk.c \
+     $(MY_LIBVPX_PATH)/vp8/common/arm/arm_systemdependent.c \
+     $(MY_LIBVPX_PATH)/vp8/encoder/arm/arm_csystemdependent.c \
+     $(MY_LIBVPX_PATH)/vp8/decoder/arm/arm_dsystemdependent.c \
+
+LOCAL_CFLAGS := \
+    -DHAVE_CONFIG_H=vpx_config.h \
+    -include $(LOCAL_PATH)/third_party/libvpx/source/config/android/vpx_config.h
+
+LOCAL_MODULE := libwebrtc_vpx
+
+LOCAL_C_INCLUDES := \
+    external/libvpx \
+    external/libvpx/vpx_ports \
+    external/libvpx/vp8/common \
+    external/libvpx/vp8/encoder \
+    external/libvpx/vp8 \
+    external/libvpx/vpx_codec 
+
+include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/license_template.txt b/trunk/license_template.txt
new file mode 100644
index 0000000..5a3e653
--- /dev/null
+++ b/trunk/license_template.txt
@@ -0,0 +1,10 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
diff --git a/trunk/peerconnection/DEPS b/trunk/peerconnection/DEPS
new file mode 100644
index 0000000..7f1e4fe
--- /dev/null
+++ b/trunk/peerconnection/DEPS
@@ -0,0 +1,132 @@
+vars = {
+  # Use this googlecode_url variable only if there is an internal mirror for it.
+  # If you do not know, use the full path while defining your new deps entry.
+  "googlecode_url": "http://%s.googlecode.com/svn",
+  "chromium_trunk" : "http://src.chromium.org/svn/trunk",
+  "chromium_revision": "120526",
+  "webrtc_revision": "1538",
+  "libjingle_revision": "115",
+}
+
+# NOTE: Prefer revision numbers to tags for svn deps. Use http rather than
+# https; the latter can cause problems for users behind proxies.
+deps = {
+  # WebRTC deps.
+  "trunk/src":
+    (Var("googlecode_url") % "webrtc") + "/trunk/src@" + Var("webrtc_revision"),
+
+  "trunk/tools":
+    (Var("googlecode_url") % "webrtc") + "/trunk/tools@" + Var("webrtc_revision"),
+
+  "trunk/test":
+    (Var("googlecode_url") % "webrtc") + "/trunk/test@" + Var("webrtc_revision"),
+
+  "trunk/third_party/google-gflags":
+    (Var("googlecode_url") % "webrtc") + "/trunk/third_party/google-gflags@" + Var("webrtc_revision"),
+
+  "trunk/third_party/libvpx":
+    (Var("googlecode_url") % "webrtc") + "/trunk/third_party/libvpx@" + Var("webrtc_revision"),
+
+  "trunk/build":
+    Var("chromium_trunk") + "/src/build@" + Var("chromium_revision"),
+
+  "trunk/testing":
+    Var("chromium_trunk") + "/src/testing@" + Var("chromium_revision"),
+
+  "trunk/testing/gtest":
+    (Var("googlecode_url") % "googletest") + "/trunk@573",
+
+  "trunk/testing/gmock":
+    (Var("googlecode_url") % "googlemock") + "/trunk@386",
+
+  "trunk/tools/gyp":
+    (Var("googlecode_url") % "gyp") + "/trunk@1187",
+
+  # Needed by build/common.gypi.
+  "trunk/tools/win/supalink":
+    Var("chromium_trunk") + "/src/tools/win/supalink@" + Var("chromium_revision"),
+
+  "trunk/tools/clang/scripts":
+    Var("chromium_trunk") + "/src/tools/clang/scripts@" + Var("chromium_revision"),
+
+  "trunk/tools/python":
+    Var("chromium_trunk") + "/src/tools/python@" + Var("chromium_revision"),
+
+  "trunk/tools/valgrind":
+    Var("chromium_trunk") + "/src/tools/valgrind@" + Var("chromium_revision"),
+
+  "trunk/third_party/protobuf/":
+    Var("chromium_trunk") + "/src/third_party/protobuf@" + Var("chromium_revision"),
+
+  "trunk/third_party/libvpx/source/libvpx":
+    "http://git.chromium.org/webm/libvpx.git@e479379a",
+
+  "trunk/third_party/libjpeg_turbo/":
+    Var("chromium_trunk") + "/deps/third_party/libjpeg_turbo@119959",
+
+  "trunk/third_party/libjpeg/":
+    Var("chromium_trunk") + "/src/third_party/libjpeg@" + Var("chromium_revision"),
+
+  "trunk/third_party/yasm/":
+    Var("chromium_trunk") + "/src/third_party/yasm@" + Var("chromium_revision"),
+
+  "trunk/third_party/expat/":
+    Var("chromium_trunk") + "/src/third_party/expat@" + Var("chromium_revision"),
+
+  "trunk/third_party/google-gflags/src":
+    (Var("googlecode_url") % "google-gflags") + "/trunk/src@45",
+
+  "trunk/third_party/yasm/source/patched-yasm":
+    Var("chromium_trunk") + "/deps/third_party/yasm/patched-yasm@73761",
+
+  # Used by libjpeg-turbo
+  "trunk/third_party/yasm/binaries":
+    Var("chromium_trunk") + "/deps/third_party/yasm/binaries@74228",
+
+  # TODO(andrew): roll to 164 after fixing:
+  # http://code.google.com/p/webrtc/issues/detail?id=267
+  "trunk/third_party/libyuv":
+    (Var("googlecode_url") % "libyuv") + "/trunk@121",
+
+  # libjingle deps.
+  "trunk/third_party/libjingle/source":
+    (Var("googlecode_url") % "libjingle") + "/trunk/@" + Var("libjingle_revision"),
+
+  "trunk/third_party/libsrtp/":
+    Var("chromium_trunk") + "/deps/third_party/libsrtp@119742",
+
+  "trunk/third_party/jsoncpp/":
+    Var("chromium_trunk") + "/src/third_party/jsoncpp@" + Var("chromium_revision"),
+
+  "trunk/third_party/jsoncpp/source":
+    "http://jsoncpp.svn.sourceforge.net/svnroot/jsoncpp/trunk/jsoncpp@248",
+}
+
+deps_os = {
+  "win": {
+    "trunk/third_party/cygwin/":
+      Var("chromium_trunk") + "/deps/third_party/cygwin@66844",
+  }
+}
+
+hooks = [
+  {
+    # Create a supplement.gypi file under trunk/.  This file will be picked up
+    # by gyp and we use it to set Chromium related variables (inside_chromium_build)
+    # to 0 and enable the standalone build.
+    "pattern": ".",
+    "action": ["python", "trunk/tools/create_supplement_gypi.py", "trunk/src/supplement.gypi"],
+  },
+  {
+    # Pull clang on mac. If nothing changed, or on non-mac platforms, this takes
+    # zero seconds to run. If something changed, it downloads a prebuilt clang.
+    "pattern": ".",
+    "action": ["python", "trunk/tools/clang/scripts/update.py", "--mac-only"],
+  },
+  {
+    # A change to a .gyp, .gypi, or to GYP itself should run the generator.
+    "pattern": ".",
+    "action": ["python", "trunk/build/gyp_chromium", "--depth=trunk", "trunk/peerconnection.gyp"],
+  },
+]
+
diff --git a/trunk/peerconnection/OWNERS b/trunk/peerconnection/OWNERS
new file mode 100644
index 0000000..1527445
--- /dev/null
+++ b/trunk/peerconnection/OWNERS
@@ -0,0 +1,5 @@
+henrike@webrtc.org
+mallinath@webrtc.org
+perkj@webrtc.org
+wu@webrtc.org
+tommi@webrtc.org
diff --git a/trunk/peerconnection/README b/trunk/peerconnection/README
new file mode 100644
index 0000000..bfe1234
--- /dev/null
+++ b/trunk/peerconnection/README
@@ -0,0 +1,10 @@
+This folder can be used to pull together the chromium version of webrtc
+and libjingle, and build the peerconnection sample client and server.
+
+Follow the instructions for WebRTC here:
+http://www.webrtc.org/reference/getting-started
+
+but substitute this "gclient config" command:
+gclient config --name trunk
+http://webrtc.googlecode.com/svn/trunk/peerconnection
+
diff --git a/trunk/peerconnection/peerconnection.gyp b/trunk/peerconnection/peerconnection.gyp
new file mode 100644
index 0000000..e928194
--- /dev/null
+++ b/trunk/peerconnection/peerconnection.gyp
@@ -0,0 +1,113 @@
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+  'includes': [ 'src/build/common.gypi', ],
+  'variables': {
+    'peerconnection_sample': 'third_party/libjingle/source/talk/examples/peerconnection',
+  },  
+
+  'targets': [
+    {
+      'target_name': 'peerconnection_server',
+      'type': 'executable',
+      'sources': [
+        '<(peerconnection_sample)/server/data_socket.cc',
+        '<(peerconnection_sample)/server/data_socket.h',
+        '<(peerconnection_sample)/server/main.cc',
+        '<(peerconnection_sample)/server/peer_channel.cc',
+        '<(peerconnection_sample)/server/peer_channel.h',
+        '<(peerconnection_sample)/server/utils.cc',
+        '<(peerconnection_sample)/server/utils.h',
+      ],
+      'include_dirs': [
+        'third_party/libjingle/source',
+      ],
+    },
+  ],
+  'conditions': [
+    ['OS=="win"', {
+      'targets': [
+        {
+          'target_name': 'peerconnection_client',
+          'type': 'executable',
+          'sources': [
+            '<(peerconnection_sample)/client/conductor.cc',
+            '<(peerconnection_sample)/client/conductor.h',
+            '<(peerconnection_sample)/client/defaults.cc',
+            '<(peerconnection_sample)/client/defaults.h',
+            '<(peerconnection_sample)/client/main.cc',
+            '<(peerconnection_sample)/client/main_wnd.cc',
+            '<(peerconnection_sample)/client/main_wnd.h',
+            '<(peerconnection_sample)/client/peer_connection_client.cc',
+            '<(peerconnection_sample)/client/peer_connection_client.h',
+            'third_party/libjingle/source/talk/base/win32socketinit.cc',
+            'third_party/libjingle/source/talk/base/win32socketserver.cc',
+          ],
+          'msvs_settings': {
+            'VCLinkerTool': {
+             'SubSystem': '2',  # Windows
+            },
+          },
+          'dependencies': [
+            'third_party/libjingle/libjingle.gyp:libjingle_app',
+          ],
+          'include_dirs': [
+            'src',
+            'src/modules/interface',
+            'third_party/libjingle/source',
+          ],
+        },
+      ],  # targets
+    }, ],  # OS="win"
+    ['OS=="linux"', {
+      'targets': [
+        {
+          'target_name': 'peerconnection_client',
+          'type': 'executable',
+          'sources': [
+            '<(peerconnection_sample)/client/conductor.cc',
+            '<(peerconnection_sample)/client/conductor.h',
+            '<(peerconnection_sample)/client/defaults.cc',
+            '<(peerconnection_sample)/client/defaults.h',
+            '<(peerconnection_sample)/client/linux/main.cc',
+            '<(peerconnection_sample)/client/linux/main_wnd.cc',
+            '<(peerconnection_sample)/client/linux/main_wnd.h',
+            '<(peerconnection_sample)/client/peer_connection_client.cc',
+            '<(peerconnection_sample)/client/peer_connection_client.h',
+          ],
+          'dependencies': [
+            'third_party/libjingle/libjingle.gyp:libjingle_app',
+            # TODO(tommi): Switch to this and remove specific gtk dependency
+            # sections below for cflags and link_settings.
+            # '<(DEPTH)/build/linux/system.gyp:gtk',
+          ],
+          'include_dirs': [
+            'src',
+            'src/modules/interface',
+            'third_party/libjingle/source',
+          ],
+          'cflags': [
+            '<!@(pkg-config --cflags gtk+-2.0)',
+          ],
+          'link_settings': {
+            'ldflags': [
+              '<!@(pkg-config --libs-only-L --libs-only-other gtk+-2.0 gthread-2.0)',
+            ],
+            'libraries': [
+              '<!@(pkg-config --libs-only-l gtk+-2.0 gthread-2.0)',
+              '-lX11',
+              '-lXext',
+            ],
+          },
+        },
+      ],  # targets
+    }, ],  # OS="linux"
+  ],
+
+}
diff --git a/trunk/peerconnection/third_party/libjingle/libjingle.gyp b/trunk/peerconnection/third_party/libjingle/libjingle.gyp
new file mode 100644
index 0000000..48939bf
--- /dev/null
+++ b/trunk/peerconnection/third_party/libjingle/libjingle.gyp
@@ -0,0 +1,698 @@
+# Copyright (c) 2012 The Chromium Authors. All rights reserved.
+# Use of this source code is governed by a BSD-style license that can be
+# found in the LICENSE file.
+
+{
+  # TODO(wu): Use the libjingle.gyp from chromium and get rid of this file.
+  'variables': {
+    'no_libjingle_logging%': 0,
+    'conditions': [
+      ['inside_chromium_build==1', {
+        'overrides': 'overrides',
+      },{
+        'overrides': 'source',
+      }],
+    ],
+  },
+  'target_defaults': {
+    'defines': [
+      'FEATURE_ENABLE_SSL',
+      'FEATURE_ENABLE_VOICEMAIL',  # TODO(ncarter): Do we really need this?
+      '_USE_32BIT_TIME_T',
+      'SAFE_TO_DEFINE_TALK_BASE_LOGGING_MACROS',
+      'EXPAT_RELATIVE_PATH',
+      'JSONCPP_RELATIVE_PATH',
+      'WEBRTC_RELATIVE_PATH',
+      'HAVE_WEBRTC',
+      'HAVE_WEBRTC_VIDEO',
+      'HAVE_WEBRTC_VOICE',
+      'HAVE_SRTP',
+      'SRTP_RELATIVE_PATH',
+    ],
+    'configurations': {
+      'Debug': {
+        'defines': [
+          # TODO(sergeyu): Fix libjingle to use NDEBUG instead of
+          # _DEBUG and remove this define. See below as well.
+          '_DEBUG',
+        ],
+      }
+    },
+    'dependencies': [
+      '../expat/expat.gyp:expat',
+    ],
+    'export_dependent_settings': [
+      '../expat/expat.gyp:expat',
+    ],
+    'include_dirs': [
+      '../../third_party/libyuv/include/',
+    ],
+    'direct_dependent_settings': {
+      'defines': [
+        'FEATURE_ENABLE_SSL',
+        'FEATURE_ENABLE_VOICEMAIL',
+        'EXPAT_RELATIVE_PATH',
+        'WEBRTC_RELATIVE_PATH',
+      ],
+      'conditions': [
+        ['OS=="win"', {
+          'link_settings': {
+            'libraries': [
+              '-lsecur32.lib',
+              '-lcrypt32.lib',
+              '-liphlpapi.lib',
+            ],
+          },
+        }],
+        ['OS=="win"', {
+          'include_dirs': [
+            '../third_party/platformsdk_win7/files/Include',
+          ],
+          'defines': [
+              '_CRT_SECURE_NO_WARNINGS',  # Suppres warnings about _vsnprinf
+          ],
+        }],
+        ['OS=="linux"', {
+          'defines': [
+            'LINUX',
+          ],
+        }],
+        ['OS=="mac"', {
+          'defines': [
+            'OSX',
+          ],
+        }],
+        ['os_posix == 1', {
+          'defines': [
+            'POSIX',
+          ],
+        }],
+        ['OS=="openbsd" or OS=="freebsd"', {
+          'defines': [
+            'BSD',
+          ],
+        }],
+        ['no_libjingle_logging==1', {
+          'defines': [
+            'NO_LIBJINGLE_LOGGING',
+          ],
+        }],
+      ],
+    },
+    'all_dependent_settings': {
+      'configurations': {
+        'Debug': {
+          'defines': [
+            # TODO(sergeyu): Fix libjingle to use NDEBUG instead of
+            # _DEBUG and remove this define. See above as well.
+            '_DEBUG',
+          ],
+        }
+      },
+    },
+    'conditions': [
+      ['inside_chromium_build==1', {
+        'defines': [
+          'NO_SOUND_SYSTEM',
+        ],
+        'include_dirs': [
+          '<(overrides)',
+          'source',
+          '../..',  # the third_party folder for webrtc includes
+          '../../third_party/expat/files',
+        ],
+        'direct_dependent_settings': {
+          'defines': [
+            'NO_SOUND_SYSTEM',
+          ],
+          'include_dirs': [
+            '<(overrides)',
+            'source',
+            '../../third_party/expat/files'
+          ],
+        },
+        'dependencies': [
+          '../../base/base.gyp:base',
+          '../../net/net.gyp:net',
+        ],
+      },{
+        'include_dirs': [
+          # the third_party folder for webrtc/ includes (non-chromium).
+          '../../src',
+          'source',
+          '../../third_party/expat/files',
+        ],
+      }],
+      ['OS=="win"', {
+        'include_dirs': [
+          '../third_party/platformsdk_win7/files/Include',
+        ],
+      }],
+      ['OS=="linux"', {
+        'defines': [
+          'LINUX',
+        ],
+      }],
+      ['OS=="mac"', {
+        'defines': [
+          'OSX',
+        ],
+      }],
+      ['os_posix == 1', {
+        'defines': [
+          'POSIX',
+        ],
+      }],
+      ['OS=="openbsd" or OS=="freebsd"', {
+        'defines': [
+          'BSD',
+        ],
+      }],
+    ],
+  },
+  'targets': [
+    {
+      'target_name': 'libjingle',
+      'type': 'static_library',
+      'sources': [
+        '<(overrides)/talk/base/basictypes.h',
+        '<(overrides)/talk/base/constructormagic.h',
+
+        # Need to override logging.h because we need
+        # SAFE_TO_DEFINE_TALK_BASE_LOGGING_MACROS to work.
+        # TODO(sergeyu): push SAFE_TO_DEFINE_TALK_BASE_LOGGING_MACROS to
+        # libjingle and remove this override.
+        '<(overrides)/talk/base/logging.h',
+
+        '<(overrides)/talk/base/scoped_ptr.h',
+
+        # Libjingle's QName is not threadsafe, so we need to use our own version
+        # here.
+        # TODO(sergeyu): Fix QName in Libjingle.
+        '<(overrides)/talk/xmllite/qname.cc',
+        '<(overrides)/talk/xmllite/qname.h',
+
+        'source/talk/base/Equifax_Secure_Global_eBusiness_CA-1.h',
+        'source/talk/base/asyncfile.cc',
+        'source/talk/base/asyncfile.h',
+        'source/talk/base/asynchttprequest.cc',
+        'source/talk/base/asynchttprequest.h',
+        'source/talk/base/asyncpacketsocket.h',
+        'source/talk/base/asyncsocket.cc',
+        'source/talk/base/asyncsocket.h',
+        'source/talk/base/asynctcpsocket.cc',
+        'source/talk/base/asynctcpsocket.h',
+        'source/talk/base/asyncudpsocket.cc',
+        'source/talk/base/asyncudpsocket.h',
+        'source/talk/base/autodetectproxy.cc',
+        'source/talk/base/autodetectproxy.h',
+        'source/talk/base/base64.cc',
+        'source/talk/base/base64.h',
+        'source/talk/base/basicdefs.h',
+        'source/talk/base/basicpacketsocketfactory.cc',
+        'source/talk/base/basicpacketsocketfactory.h',
+        'source/talk/base/bytebuffer.cc',
+        'source/talk/base/bytebuffer.h',
+        'source/talk/base/byteorder.h',
+        'source/talk/base/checks.cc',
+        'source/talk/base/checks.h',
+        'source/talk/base/common.cc',
+        'source/talk/base/common.h',
+        'source/talk/base/criticalsection.h',
+        'source/talk/base/cryptstring.h',
+        'source/talk/base/diskcache.cc',
+        'source/talk/base/diskcache.h',
+        'source/talk/base/event.cc',
+        'source/talk/base/event.h',
+        'source/talk/base/fileutils.cc',
+        'source/talk/base/fileutils.h',
+        'source/talk/base/firewallsocketserver.cc',
+        'source/talk/base/firewallsocketserver.h',
+        'source/talk/base/flags.cc',
+        'source/talk/base/flags.h',
+        'source/talk/base/helpers.cc',
+        'source/talk/base/helpers.h',
+        'source/talk/base/host.cc',
+        'source/talk/base/host.h',
+        'source/talk/base/httpbase.cc',
+        'source/talk/base/httpbase.h',
+        'source/talk/base/httpclient.h',
+        'source/talk/base/httpclient.cc',
+        'source/talk/base/httpcommon-inl.h',
+        'source/talk/base/httpcommon.cc',
+        'source/talk/base/httpcommon.h',
+        'source/talk/base/httprequest.cc',
+        'source/talk/base/httprequest.h',
+        'source/talk/base/ipaddress.cc',
+        'source/talk/base/ipaddress.h',
+        'source/talk/base/json.cc',
+        'source/talk/base/json.h',
+        'source/talk/base/linked_ptr.h',
+        'source/talk/base/logging.cc',
+        'source/talk/base/md5.h',
+        'source/talk/base/md5c.c',
+        'source/talk/base/messagehandler.cc',
+        'source/talk/base/messagehandler.h',
+        'source/talk/base/messagequeue.cc',
+        'source/talk/base/messagequeue.h',
+        'source/talk/base/nethelpers.cc',
+        'source/talk/base/nethelpers.h',
+        'source/talk/base/network.cc',
+        'source/talk/base/network.h',
+        'source/talk/base/pathutils.cc',
+        'source/talk/base/pathutils.h',
+        'source/talk/base/physicalsocketserver.cc',
+        'source/talk/base/physicalsocketserver.h',
+        'source/talk/base/proxydetect.cc',
+        'source/talk/base/proxydetect.h',
+        'source/talk/base/proxyinfo.cc',
+        'source/talk/base/proxyinfo.h',
+        'source/talk/base/ratetracker.cc',
+        'source/talk/base/ratetracker.h',
+        'source/talk/base/sec_buffer.h',
+        'source/talk/base/signalthread.cc',
+        'source/talk/base/signalthread.h',
+        'source/talk/base/sigslot.h',
+        'source/talk/base/sigslotrepeater.h',
+        'source/talk/base/socket.h',
+        'source/talk/base/socketadapters.cc',
+        'source/talk/base/socketadapters.h',
+        'source/talk/base/socketaddress.cc',
+        'source/talk/base/socketaddress.h',
+        'source/talk/base/socketaddresspair.cc',
+        'source/talk/base/socketaddresspair.h',
+        'source/talk/base/socketfactory.h',
+        'source/talk/base/socketpool.cc',
+        'source/talk/base/socketpool.h',
+        'source/talk/base/socketserver.h',
+        'source/talk/base/socketstream.cc',
+        'source/talk/base/socketstream.h',
+        'source/talk/base/ssladapter.cc',
+        'source/talk/base/ssladapter.h',
+        'source/talk/base/sslsocketfactory.cc',
+        'source/talk/base/sslsocketfactory.h',
+        'source/talk/base/stream.cc',
+        'source/talk/base/stream.h',
+        'source/talk/base/stringdigest.cc',
+        'source/talk/base/stringdigest.h',
+        'source/talk/base/stringencode.cc',
+        'source/talk/base/stringencode.h',
+        'source/talk/base/stringutils.cc',
+        'source/talk/base/stringutils.h',
+        'source/talk/base/task.cc',
+        'source/talk/base/task.h',
+        'source/talk/base/taskparent.cc',
+        'source/talk/base/taskparent.h',
+        'source/talk/base/taskrunner.cc',
+        'source/talk/base/taskrunner.h',
+        'source/talk/base/thread.cc',
+        'source/talk/base/thread.h',
+        'source/talk/base/time.cc',
+        'source/talk/base/time.h',
+        'source/talk/base/urlencode.cc',
+        'source/talk/base/urlencode.h',
+        'source/talk/base/worker.cc', 
+        'source/talk/base/worker.h', 
+        'source/talk/xmllite/xmlbuilder.cc',
+        'source/talk/xmllite/xmlbuilder.h',
+        'source/talk/xmllite/xmlconstants.cc',
+        'source/talk/xmllite/xmlconstants.h',
+        'source/talk/xmllite/xmlelement.cc',
+        'source/talk/xmllite/xmlelement.h',
+        'source/talk/xmllite/xmlnsstack.cc',
+        'source/talk/xmllite/xmlnsstack.h',
+        'source/talk/xmllite/xmlparser.cc',
+        'source/talk/xmllite/xmlparser.h',
+        'source/talk/xmllite/xmlprinter.cc',
+        'source/talk/xmllite/xmlprinter.h',
+        'source/talk/xmpp/asyncsocket.h',
+        'source/talk/xmpp/constants.cc',
+        'source/talk/xmpp/constants.h',
+        'source/talk/xmpp/jid.cc',
+        'source/talk/xmpp/jid.h',
+        'source/talk/xmpp/plainsaslhandler.h',
+        'source/talk/xmpp/prexmppauth.h',
+        'source/talk/xmpp/ratelimitmanager.cc',
+        'source/talk/xmpp/ratelimitmanager.h',
+        'source/talk/xmpp/saslcookiemechanism.h',
+        'source/talk/xmpp/saslhandler.h',
+        'source/talk/xmpp/saslmechanism.cc',
+        'source/talk/xmpp/saslmechanism.h',
+        'source/talk/xmpp/saslplainmechanism.h',
+        'source/talk/xmpp/xmppclient.cc',
+        'source/talk/xmpp/xmppclient.h',
+        'source/talk/xmpp/xmppclientsettings.h',
+        'source/talk/xmpp/xmppengine.h',
+        'source/talk/xmpp/xmppengineimpl.cc',
+        'source/talk/xmpp/xmppengineimpl.h',
+        'source/talk/xmpp/xmppengineimpl_iq.cc',
+        'source/talk/xmpp/xmpplogintask.cc',
+        'source/talk/xmpp/xmpplogintask.h',
+        'source/talk/xmpp/xmppstanzaparser.cc',
+        'source/talk/xmpp/xmppstanzaparser.h',
+        'source/talk/xmpp/xmpptask.cc',
+        'source/talk/xmpp/xmpptask.h',
+      ],
+      'conditions': [
+        ['inside_chromium_build==0', {
+          'sources': [ 
+            'source/talk/sound/automaticallychosensoundsystem.h', 
+            'source/talk/sound/platformsoundsystem.cc', 
+            'source/talk/sound/platformsoundsystem.h', 
+            'source/talk/sound/platformsoundsystemfactory.cc', 
+            'source/talk/sound/platformsoundsystemfactory.h', 
+            'source/talk/sound/sounddevicelocator.h', 
+            'source/talk/sound/soundinputstreaminterface.h', 
+            'source/talk/sound/soundoutputstreaminterface.h', 
+            'source/talk/sound/soundsystemfactory.h', 
+            'source/talk/sound/soundsysteminterface.cc', 
+            'source/talk/sound/soundsysteminterface.h', 
+            'source/talk/sound/soundsystemproxy.cc', 
+            'source/talk/sound/soundsystemproxy.h',
+          ],
+          'conditions' : [ 
+            ['OS=="linux"', {
+              'sources': [
+                'source/talk/sound/alsasoundsystem.cc',
+                'source/talk/sound/alsasoundsystem.h',
+                'source/talk/sound/alsasymboltable.cc',
+                'source/talk/sound/alsasymboltable.h',
+                'source/talk/sound/linuxsoundsystem.cc',
+                'source/talk/sound/linuxsoundsystem.h',
+                'source/talk/sound/pulseaudiosoundsystem.cc',
+                'source/talk/sound/pulseaudiosoundsystem.h',
+                'source/talk/sound/pulseaudiosymboltable.cc',
+                'source/talk/sound/pulseaudiosymboltable.h',
+              ],
+            }],
+          ],
+        }],
+        ['OS=="win"', {
+          'sources': [
+            '<(overrides)/talk/base/win32socketinit.cc',
+            'source/talk/base/schanneladapter.cc',
+            'source/talk/base/schanneladapter.h',
+            'source/talk/base/win32.h',
+            'source/talk/base/win32.cc',
+            'source/talk/base/win32filesystem.cc',
+            'source/talk/base/win32filesystem.h',
+            'source/talk/base/win32window.h',
+            'source/talk/base/win32window.cc',
+            'source/talk/base/win32securityerrors.cc',
+            'source/talk/base/winfirewall.cc',
+            'source/talk/base/winfirewall.h',
+            'source/talk/base/winping.cc',
+            'source/talk/base/winping.h',
+          ],
+        }],
+        ['os_posix == 1', {
+          'sources': [
+            'source/talk/base/latebindingsymboltable.cc',
+            'source/talk/base/latebindingsymboltable.h',
+            'source/talk/base/sslstreamadapter.cc',
+            'source/talk/base/sslstreamadapter.h',
+            'source/talk/base/unixfilesystem.cc',
+            'source/talk/base/unixfilesystem.h',
+          ],
+        }],
+        ['OS=="linux"', {
+          'sources': [
+            'source/talk/base/linux.cc',
+            'source/talk/base/linux.h',
+          ],
+        }],
+        ['OS=="mac"', {
+          'sources': [
+            'source/talk/base/macconversion.cc',
+            'source/talk/base/macconversion.h',
+            'source/talk/base/macutils.cc',
+            'source/talk/base/macutils.h',
+          ],
+        }],
+      ],
+      'dependencies': [
+        '<(DEPTH)/third_party/jsoncpp/jsoncpp.gyp:jsoncpp',
+      ],
+    },
+    # This has to be is a separate project due to a bug in MSVS:
+    # https://connect.microsoft.com/VisualStudio/feedback/details/368272/duplicate-cpp-filename-in-c-project-visual-studio-2008
+    # We have two files named "constants.cc" and MSVS doesn't handle this
+    # properly.
+    {
+      'target_name': 'libjingle_p2p',
+      'type': 'static_library',
+      'sources': [
+        'source/talk/p2p/base/candidate.h',
+        'source/talk/p2p/base/common.h',
+        'source/talk/p2p/base/constants.cc',
+        'source/talk/p2p/base/constants.h',
+        'source/talk/p2p/base/p2ptransport.cc',
+        'source/talk/p2p/base/p2ptransport.h',
+        'source/talk/p2p/base/p2ptransportchannel.cc',
+        'source/talk/p2p/base/p2ptransportchannel.h',
+        'source/talk/p2p/base/port.cc',
+        'source/talk/p2p/base/port.h',
+        'source/talk/p2p/base/portallocator.cc',
+        'source/talk/p2p/base/portallocator.h',
+        'source/talk/p2p/base/portallocatorsessionproxy.cc',
+        'source/talk/p2p/base/portallocatorsessionproxy.h',
+        'source/talk/p2p/base/portproxy.cc',
+        'source/talk/p2p/base/portproxy.h',
+        'source/talk/p2p/base/pseudotcp.cc',
+        'source/talk/p2p/base/pseudotcp.h',
+        'source/talk/p2p/base/rawtransport.cc',
+        'source/talk/p2p/base/rawtransport.h',
+        'source/talk/p2p/base/rawtransportchannel.cc',
+        'source/talk/p2p/base/rawtransportchannel.h',
+        'source/talk/p2p/base/relayport.cc',
+        'source/talk/p2p/base/relayport.h',
+        'source/talk/p2p/base/session.cc',
+        'source/talk/p2p/base/session.h',
+        'source/talk/p2p/base/sessionclient.h',
+        'source/talk/p2p/base/sessiondescription.cc',
+        'source/talk/p2p/base/sessiondescription.h',
+        'source/talk/p2p/base/sessionid.h',
+        'source/talk/p2p/base/sessionmanager.cc',
+        'source/talk/p2p/base/sessionmanager.h',
+        'source/talk/p2p/base/sessionmessages.cc',
+        'source/talk/p2p/base/sessionmessages.h',
+        'source/talk/p2p/base/parsing.cc',
+        'source/talk/p2p/base/parsing.h',
+        'source/talk/p2p/base/stun.cc',
+        'source/talk/p2p/base/stun.h',
+        'source/talk/p2p/base/stunport.cc',
+        'source/talk/p2p/base/stunport.h',
+        'source/talk/p2p/base/stunrequest.cc',
+        'source/talk/p2p/base/stunrequest.h',
+        'source/talk/p2p/base/tcpport.cc',
+        'source/talk/p2p/base/tcpport.h',
+        'source/talk/p2p/base/transport.cc',
+        'source/talk/p2p/base/transport.h',
+        'source/talk/p2p/base/transportchannel.cc',
+        'source/talk/p2p/base/transportchannel.h',
+        'source/talk/p2p/base/transportchannelimpl.h',
+        'source/talk/p2p/base/transportchannelproxy.cc',
+        'source/talk/p2p/base/transportchannelproxy.h',
+        'source/talk/p2p/base/udpport.cc',
+        'source/talk/p2p/base/udpport.h',
+        'source/talk/p2p/client/basicportallocator.cc',
+        'source/talk/p2p/client/basicportallocator.h',
+        'source/talk/p2p/client/httpportallocator.cc',
+        'source/talk/p2p/client/httpportallocator.h',
+        'source/talk/p2p/client/sessionmanagertask.h',
+        'source/talk/p2p/client/sessionsendtask.h',
+        'source/talk/p2p/client/socketmonitor.cc',
+        'source/talk/p2p/client/socketmonitor.h',
+        'source/talk/session/phone/audiomonitor.cc',
+        'source/talk/session/phone/audiomonitor.h',
+        'source/talk/session/phone/call.cc',
+        'source/talk/session/phone/call.h',
+        'source/talk/session/phone/channel.cc',
+        'source/talk/session/phone/channel.h',
+        'source/talk/session/phone/channelmanager.cc',
+        'source/talk/session/phone/channelmanager.h',
+        'source/talk/session/phone/codec.cc',
+        'source/talk/session/phone/codec.h',
+        'source/talk/session/phone/cryptoparams.h',
+        'source/talk/session/phone/currentspeakermonitor.cc',
+        'source/talk/session/phone/currentspeakermonitor.h',
+        'source/talk/session/phone/devicemanager.cc',
+        'source/talk/session/phone/devicemanager.h',
+        'source/talk/session/phone/filemediaengine.cc',
+        'source/talk/session/phone/filemediaengine.h',   
+        'source/talk/session/phone/mediachannel.h',
+        'source/talk/session/phone/mediaengine.cc',
+        'source/talk/session/phone/mediaengine.h',
+        'source/talk/session/phone/mediamessages.cc',
+        'source/talk/session/phone/mediamessages.h',
+        'source/talk/session/phone/mediamonitor.cc',
+        'source/talk/session/phone/mediamonitor.h',
+        'source/talk/session/phone/mediasession.cc',
+        'source/talk/session/phone/mediasession.h',
+        'source/talk/session/phone/mediasessionclient.cc',
+        'source/talk/session/phone/mediasessionclient.h',
+        'source/talk/session/phone/mediasink.h',
+        'source/talk/session/phone/rtcpmuxfilter.cc',
+        'source/talk/session/phone/rtcpmuxfilter.h',        
+        'source/talk/session/phone/rtpdump.cc',
+        'source/talk/session/phone/rtpdump.h',
+        'source/talk/session/phone/rtputils.cc',
+        'source/talk/session/phone/rtputils.h',
+        'source/talk/session/phone/soundclip.cc',
+        'source/talk/session/phone/soundclip.h',
+        'source/talk/session/phone/srtpfilter.cc',
+        'source/talk/session/phone/srtpfilter.h',
+        'source/talk/session/phone/ssrcmuxfilter.cc',
+        'source/talk/session/phone/ssrcmuxfilter.h',
+        'source/talk/session/phone/streamparams.cc',
+        'source/talk/session/phone/videocapturer.cc',
+        'source/talk/session/phone/videocapturer.h',
+        'source/talk/session/phone/videocommon.cc',
+        'source/talk/session/phone/videocommon.h',
+        'source/talk/session/phone/videoframe.cc',
+        'source/talk/session/phone/videoframe.h',
+        'source/talk/session/phone/voicechannel.h',
+        'source/talk/session/phone/webrtcpassthroughrender.cc',
+        'source/talk/session/phone/webrtccommon.h',
+        'source/talk/session/phone/webrtcvideocapturer.cc',
+        'source/talk/session/phone/webrtcvideocapturer.h',
+        'source/talk/session/phone/webrtcvideoengine.cc',
+        'source/talk/session/phone/webrtcvideoengine.h',
+        'source/talk/session/phone/webrtcvideoframe.cc',
+        'source/talk/session/phone/webrtcvideoframe.h',
+        'source/talk/session/phone/webrtcvie.h',
+        'source/talk/session/phone/webrtcvoe.h',
+        'source/talk/session/phone/webrtcvoiceengine.cc',
+        'source/talk/session/phone/webrtcvoiceengine.h',
+        'source/talk/session/tunnel/pseudotcpchannel.cc',
+        'source/talk/session/tunnel/pseudotcpchannel.h',
+        'source/talk/session/tunnel/tunnelsessionclient.cc',
+        'source/talk/session/tunnel/tunnelsessionclient.h',
+      ],
+      'conditions': [
+        ['OS=="win"', {
+          'sources': [
+            'source/talk/session/phone/gdivideorenderer.cc',
+            'source/talk/session/phone/gdivideorenderer.h',
+            'source/talk/session/phone/win32devicemanager.cc',
+            'source/talk/session/phone/win32devicemanager.h',
+          ],
+        }],   
+        ['OS=="linux"', {
+          'sources': [
+            'source/talk/session/phone/gtkvideorenderer.cc',
+            'source/talk/session/phone/gtkvideorenderer.cc',
+            'source/talk/session/phone/gtkvideorenderer.h', 
+            'source/talk/session/phone/libudevsymboltable.cc',
+            'source/talk/session/phone/libudevsymboltable.h',
+            'source/talk/session/phone/linuxdevicemanager.cc',
+            'source/talk/session/phone/linuxdevicemanager.h',
+            'source/talk/session/phone/v4llookup.cc',
+            'source/talk/session/phone/v4llookup.h',
+          ],
+          'include_dirs': [
+            'source/talk/third_party/libudev',
+          ],
+          'cflags': [
+             '<!@(pkg-config --cflags gtk+-2.0)',
+          ],
+        }],        
+        ['inside_chromium_build==1', {
+          'dependencies': [
+            '../../third_party/webrtc/modules/modules.gyp:audio_device',
+            '../../third_party/webrtc/modules/modules.gyp:video_capture_module',
+            '../../third_party/webrtc/modules/modules.gyp:video_render_module',
+            '../../third_party/webrtc/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+            '../../third_party/webrtc/video_engine/video_engine.gyp:video_engine_core',
+            '../../third_party/webrtc/voice_engine/voice_engine.gyp:voice_engine_core',
+            '<(DEPTH)/third_party/libsrtp/libsrtp.gyp:libsrtp',
+            'libjingle',
+          ],
+        }, {
+          'dependencies': [
+            '../../src/modules/modules.gyp:audio_device',
+            '../../src/modules/modules.gyp:video_capture_module',
+            '../../src/modules/modules.gyp:video_render_module',
+            '../../src/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+            '../../src/video_engine/video_engine.gyp:video_engine_core',
+            '../../src/voice_engine/voice_engine.gyp:voice_engine_core',
+            '<(DEPTH)/third_party/libsrtp/libsrtp.gyp:libsrtp',
+            'libjingle',
+          ],
+        } ],  # inside_chromium_build
+      ],  # conditions
+    },
+    # seperate project for app
+    {
+      'target_name': 'libjingle_app',
+      'type': '<(library)',
+      'sources': [        
+        'source/talk/app/webrtc/audiotrackimpl.cc',
+        'source/talk/app/webrtc/audiotrackimpl.h',
+        'source/talk/app/webrtc/mediastream.h',
+        'source/talk/app/webrtc/mediastreamhandler.cc',
+        'source/talk/app/webrtc/mediastreamhandler.h',
+        'source/talk/app/webrtc/mediastreamimpl.cc',
+        'source/talk/app/webrtc/mediastreamimpl.h',
+        'source/talk/app/webrtc/mediastreamprovider.h',
+        'source/talk/app/webrtc/mediastreamproxy.cc',
+        'source/talk/app/webrtc/mediastreamproxy.h',
+        'source/talk/app/webrtc/mediastreamtrackproxy.cc',
+        'source/talk/app/webrtc/mediastreamtrackproxy.h',
+        'source/talk/app/webrtc/mediatrackimpl.h',
+        'source/talk/app/webrtc/notifierimpl.h',
+        'source/talk/app/webrtc/peerconnection.h',
+        'source/talk/app/webrtc/peerconnectionfactoryimpl.cc',
+        'source/talk/app/webrtc/peerconnectionfactoryimpl.h',
+        'source/talk/app/webrtc/peerconnectionimpl.cc',
+        'source/talk/app/webrtc/peerconnectionimpl.h',
+        'source/talk/app/webrtc/peerconnectionsignaling.cc',
+        'source/talk/app/webrtc/peerconnectionsignaling.h',
+        'source/talk/app/webrtc/portallocatorfactory.cc',
+        'source/talk/app/webrtc/portallocatorfactory.h',
+        'source/talk/app/webrtc/roaperrorcodes.h',
+        'source/talk/app/webrtc/roapmessages.cc',
+        'source/talk/app/webrtc/roapmessages.h',
+        'source/talk/app/webrtc/roapsession.cc',
+        'source/talk/app/webrtc/roapsession.h',
+        'source/talk/app/webrtc/sessiondescriptionprovider.h',
+        'source/talk/app/webrtc/streamcollectionimpl.h',
+        'source/talk/app/webrtc/videorendererimpl.cc',
+        'source/talk/app/webrtc/videotrackimpl.cc',
+        'source/talk/app/webrtc/videotrackimpl.h',
+        'source/talk/app/webrtc/webrtcjson.cc',
+        'source/talk/app/webrtc/webrtcjson.h',
+        'source/talk/app/webrtc/webrtcsdp.cc',
+        'source/talk/app/webrtc/webrtcsdp.h',
+        'source/talk/app/webrtc/webrtcsession.cc',
+        'source/talk/app/webrtc/webrtcsession.h',
+        'source/talk/app/webrtc/webrtcsessionobserver.h',
+      ],
+      'dependencies': [
+        '<(DEPTH)/third_party/jsoncpp/jsoncpp.gyp:jsoncpp',
+        '<(DEPTH)/third_party/libsrtp/libsrtp.gyp:libsrtp',
+      ],
+      'conditions': [
+        ['inside_chromium_build==1', {        
+          'dependencies': [
+            '../../third_party/webrtc/modules/modules.gyp:video_capture_module',
+            '../../third_party/webrtc/modules/modules.gyp:video_render_module',
+            '../../third_party/webrtc/video_engine/video_engine.gyp:video_engine_core',
+            '../../third_party/webrtc/voice_engine/voice_engine.gyp:voice_engine_core',
+            '../../third_party/webrtc/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+            'libjingle_p2p',
+          ],          
+        }, {
+          'dependencies': [
+            '../../src/modules/modules.gyp:video_capture_module',
+            '../../src/modules/modules.gyp:video_render_module',
+            '../../src/video_engine/video_engine.gyp:video_engine_core',
+            '../../src/voice_engine/voice_engine.gyp:voice_engine_core',
+            '../../src/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+            'libjingle_p2p',
+          ],          
+        } ],  # inside_chromium_build
+      ],  # conditions
+    },    
+  ],
+}
diff --git a/trunk/src/LICENSE b/trunk/src/LICENSE
new file mode 100644
index 0000000..4c41b7b
--- /dev/null
+++ b/trunk/src/LICENSE
@@ -0,0 +1,29 @@
+Copyright (c) 2011, The WebRTC project authors. All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+  * Redistributions of source code must retain the above copyright
+    notice, this list of conditions and the following disclaimer.
+
+  * Redistributions in binary form must reproduce the above copyright
+    notice, this list of conditions and the following disclaimer in
+    the documentation and/or other materials provided with the
+    distribution.
+
+  * Neither the name of Google nor the names of its contributors may
+    be used to endorse or promote products derived from this software
+    without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+HOLDER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
diff --git a/trunk/src/LICENSE_THIRD_PARTY b/trunk/src/LICENSE_THIRD_PARTY
new file mode 100644
index 0000000..e19708a
--- /dev/null
+++ b/trunk/src/LICENSE_THIRD_PARTY
@@ -0,0 +1,27 @@
+This source tree contains third party source code which is governed by third 
+party licenses. This file contains references to files which are under other 
+licenses than the one provided in the LICENSE file in the root of the source
+tree.
+
+Files governed by third party licenses:
+common_audio/signal_processing/spl_sqrt_floor.c
+common_audio/signal_processing/spl_sqrt_floor.s
+modules/audio_coding/codecs/G711/main/source/g711.h
+modules/audio_coding/codecs/G711/main/source/g711.c
+modules/audio_coding/codecs/G722/main/source/g722_decode.h
+modules/audio_coding/codecs/G722/main/source/g722_decode.c
+modules/audio_coding/codecs/G722/main/source/g722_encode.h
+modules/audio_coding/codecs/G722/main/source/g722_encode.c
+modules/audio_coding/codecs/iSAC/main/source/fft.c
+modules/audio_device/main/source/Mac/portaudio/pa_memorybarrier.h
+modules/audio_device/main/source/Mac/portaudio/pa_ringbuffer.h
+modules/audio_device/main/source/Mac/portaudio/pa_ringbuffer.c
+modules/audio_processing/utility/fft4g.c
+modules/audio_processing/aec/aec_rdft.c
+system_wrappers/interface/fix_interlocked_exchange_pointer_windows.h
+system_wrappers/interface/scoped_ptr.h
+system_wrappers/interface/scoped_refptr.h
+system_wrappers/source/condition_variable_windows.cc
+system_wrappers/source/spreadsortlib/constants.hpp
+system_wrappers/source/spreadsortlib/spreadsort.hpp
+system_wrappers/source/thread_windows_set_name.h
diff --git a/trunk/src/PATENTS b/trunk/src/PATENTS
new file mode 100644
index 0000000..190607a
--- /dev/null
+++ b/trunk/src/PATENTS
@@ -0,0 +1,24 @@
+Additional IP Rights Grant (Patents)
+
+"This implementation" means the copyrightable works distributed by
+Google as part of the WebRTC code package.
+
+Google hereby grants to you a perpetual, worldwide, non-exclusive,
+no-charge, irrevocable (except as stated in this section) patent
+license to make, have made, use, offer to sell, sell, import,
+transfer, and otherwise run, modify and propagate the contents of this
+implementation of the WebRTC code package, where such license applies
+only to those patent claims, both currently owned by Google and
+acquired in the future, licensable by Google that are necessarily
+infringed by this implementation of the WebRTC code package. This
+grant does not include claims that would be infringed only as a
+consequence of further modification of this implementation. If you or
+your agent or exclusive licensee institute or order or agree to the
+institution of patent litigation against any entity (including a
+cross-claim or counterclaim in a lawsuit) alleging that this
+implementation of the WebRTC code package or any code incorporated
+within this implementation of the WebRTC code package constitutes
+direct or contributory patent infringement, or inducement of patent
+infringement, then any patent rights granted to you under this License
+for this implementation of the WebRTC code package shall terminate as
+of the date such litigation is filed.
diff --git a/trunk/src/README.chromium b/trunk/src/README.chromium
new file mode 100644
index 0000000..246c13d
--- /dev/null
+++ b/trunk/src/README.chromium
@@ -0,0 +1,13 @@
+Name: WebRTC

+URL: http://www.webrtc.org

+Version: 90

+License: BSD

+License File: LICENSE

+

+Description:

+WebRTC provides real time voice and video processing

+functionality to enable the implementation of 

+PeerConnection/MediaStream.

+

+Third party code used in this project is described 

+in the file LICENSE_THIRD_PARTY.

diff --git a/trunk/src/build/common.gypi b/trunk/src/build/common.gypi
new file mode 100644
index 0000000..74b6003
--- /dev/null
+++ b/trunk/src/build/common.gypi
@@ -0,0 +1,180 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+# This file contains common settings for building WebRTC components.
+
+{
+  'variables': {
+    # These variables need to be nested in order to use them in a conditions
+    # block to set other variables.
+    'variables': {
+      # This will be set to zero in the supplement.gypi triggered by a gclient
+      # hook in the standalone build.
+      'build_with_chromium%': 1,
+    },
+
+    'build_with_chromium%': '<(build_with_chromium)',
+
+    # The Chromium common.gypi we use treats all gyp files without
+    # chromium_code==1 as third party code. This disables many of the
+    # preferred warning settings.
+    #
+    # We can set this here to have WebRTC code treated as Chromium code. Our
+    # third party code will still have the reduced warning settings.
+    'chromium_code': 1,
+
+    # Adds video support to dependencies shared by voice and video engine.
+    # This should normally be enabled; the intended use is to disable only
+    # when building voice engine exclusively.
+    'enable_video%': 1,
+
+    # Selects fixed-point code where possible.
+    # TODO(andrew): we'd like to set this based on the target OS/architecture.
+    'prefer_fixed_point%': 0,
+
+    # Enable data logging. Produces text files with data logged within engines
+    # which can be easily parsed for offline processing.
+    'enable_data_logging%': 0,
+
+    # Disable these to not build components which can be externally provided.
+    'build_libjpeg%': 1,
+    'build_libyuv%': 1,
+
+    'conditions': [
+      ['OS=="win"', {
+        # TODO(andrew, perkj): does this need to be here?
+        # Path needed to build Direct Show base classes on Windows.
+        # The code is included in the Windows SDK.
+        'direct_show_base_classes':
+          'C:/Program Files/Microsoft SDKs/Windows/v7.1/Samples/multimedia/directshow/baseclasses/',
+      }],
+      ['build_with_chromium==1', {
+        # Exclude pulse audio on Chromium since its prerequisites don't require
+        # pulse audio.
+        'include_pulse_audio%': 0,
+
+        # Exclude internal ADM since Chromium uses its own IO handling.
+        'include_internal_audio_device%': 0,
+
+        # Exclude internal VCM in Chromium build.
+        'include_internal_video_capture%': 0,
+
+        # Exclude internal video render module in Chromium build.
+        'include_internal_video_render%': 0,
+
+        # Disable the use of protocol buffers in production code.
+        'enable_protobuf%': 0,
+
+        'webrtc_root%': '<(DEPTH)/third_party/webrtc',
+      }, {
+        # Settings for the standalone (not-in-Chromium) build.
+
+        'include_pulse_audio%': 1,
+
+        'include_internal_audio_device%': 1,
+
+        'include_internal_video_capture%': 1,
+
+        'include_internal_video_render%': 1,
+
+        'enable_protobuf%': 1,
+
+        'webrtc_root%': '<(DEPTH)/src',
+
+        'conditions': [
+          ['OS=="mac"', {
+            # TODO(andrew): clang is the default on Mac. For now, disable the
+            # Chrome plugins, which causes a flood of chromium-style warnings.
+            # Investigate enabling the plugins:
+            # http://code.google.com/p/webrtc/issues/detail?id=163
+            'clang_use_chrome_plugins%': 0,
+          }],
+        ],
+      }],
+    ], # conditions
+  },
+  'target_defaults': {
+    'include_dirs': [
+      '..','../..', # common_types.h, typedefs.h
+    ],
+    'conditions': [
+      ['build_with_chromium==1', {
+        'defines': [
+          # Changes settings for Chromium build.
+          'WEBRTC_CHROMIUM_BUILD',
+        ],
+      }, {
+        'conditions': [
+          ['os_posix==1', {
+            'cflags': [
+              '-Wextra',
+              # We need to repeat some flags from Chromium's common.gypi here
+              # that get overridden by -Wextra.
+              '-Wno-unused-parameter',
+              '-Wno-missing-field-initializers',
+            ],
+            'cflags_cc': [
+              # This is enabled for clang; enable for gcc as well.
+              '-Woverloaded-virtual',
+            ],
+          }],
+        ],
+      }],
+      ['OS=="linux"', {
+        'defines': [
+          'WEBRTC_TARGET_PC',
+          'WEBRTC_LINUX',
+          'WEBRTC_THREAD_RR',
+          # TODO(andrew): can we select this automatically?
+          # Define this if the Linux system does not support CLOCK_MONOTONIC.
+          #'WEBRTC_CLOCK_TYPE_REALTIME',
+        ],
+      }],
+      ['OS=="mac"', {
+        # TODO(andrew): what about PowerPC?
+        # Setup for Intel
+        'defines': [
+          'WEBRTC_TARGET_MAC_INTEL',
+          'WEBRTC_MAC_INTEL',
+          'WEBRTC_MAC',
+          'WEBRTC_THREAD_RR',
+          'WEBRTC_CLOCK_TYPE_REALTIME',
+        ],
+      }],
+      ['OS=="win"', {
+        'defines': [
+          'WEBRTC_TARGET_PC',
+        ],
+        # TODO(andrew): remove this block when possible.
+        # 4389: Signed/unsigned mismatch.
+        # 4373: MSVC legacy warning for ignoring const / volatile in
+        # signatures. TODO(phoglund): get rid of 4373 supression when
+        # http://code.google.com/p/webrtc/issues/detail?id=261 is solved.
+        'msvs_disabled_warnings': [4389, 4373],
+      }],
+      ['OS=="android"', {
+        'defines': [
+          'WEBRTC_TARGET_PC',
+          'WEBRTC_LINUX',
+          # TODO(leozwang): Investigate CLOCK_REALTIME and CLOCK_MONOTONIC
+          # support on Android. Keep WEBRTC_CLOCK_TYPE_REALTIME for now,
+          # remove it after I verify that CLOCK_MONOTONIC is fully functional
+          # with condition and event functions in system_wrappers.
+          'WEBRTC_CLOCK_TYPE_REALTIME',
+          'WEBRTC_THREAD_RR',
+          'WEBRTC_ANDROID',
+          'WEBRTC_ARM_INLINE_CALLS',
+          # TODO(leozwang): move WEBRTC_ARCH_ARM to typedefs.h.
+          'WEBRTC_ARCH_ARM',
+          'WEBRTC_ANDROID_OPENSLES',
+         ],
+      }],
+    ], # conditions
+  }, # target_defaults
+}
+
diff --git a/trunk/src/build/merge_libs.gyp b/trunk/src/build/merge_libs.gyp
new file mode 100644
index 0000000..d23bf11
--- /dev/null
+++ b/trunk/src/build/merge_libs.gyp
@@ -0,0 +1,43 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+  'includes': [ 'common.gypi', ],
+  'targets': [
+    {
+      'target_name': 'no_op',
+      'type': 'executable',
+      'dependencies': [
+        '../video_engine/video_engine.gyp:video_engine_core',
+      ],
+      'sources': [ 'no_op.cc', ],
+    },
+    {
+      'target_name': 'merged_lib',
+      'type': 'none',
+      'dependencies': [
+        'no_op',
+      ],
+      'actions': [
+        {
+          'variables': {
+            'output_lib_name': 'webrtc',
+            'output_lib': '<(PRODUCT_DIR)/<(STATIC_LIB_PREFIX)<(output_lib_name)_<(OS)_<(target_arch)_<(CONFIGURATION_NAME)<(STATIC_LIB_SUFFIX)',
+          },
+          'action_name': 'merge_libs',
+          'inputs': ['<(PRODUCT_DIR)/<(EXECUTABLE_PREFIX)no_op<(EXECUTABLE_SUFFIX)'],
+          'outputs': ['<(output_lib)'],
+          'action': ['python',
+                     'merge_libs.py',
+                     '<(PRODUCT_DIR)',
+                     '<(output_lib)',],
+        },
+      ],
+    },
+  ],
+}
diff --git a/trunk/src/build/merge_libs.py b/trunk/src/build/merge_libs.py
new file mode 100644
index 0000000..31c5efb
--- /dev/null
+++ b/trunk/src/build/merge_libs.py
@@ -0,0 +1,58 @@
+#!/usr/bin/env python
+
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+# Searches for libraries and/or object files on the specified path and
+# merges them into a single library.
+
+import subprocess
+import sys
+
+if __name__ == '__main__':
+  if len(sys.argv) != 3:
+    sys.stderr.write('Usage: ' + sys.argv[0] + ' <search_path> <output_lib>\n')
+    sys.exit(2)
+
+  search_path = sys.argv[1]
+  output_lib = sys.argv[2]
+
+  from subprocess import call, PIPE
+  if sys.platform.startswith('linux'):
+    call(["rm -f " + output_lib], shell=True)
+    call(["rm -rf " + search_path + "/obj.target/*do_not_use*"], shell=True)
+    call(["ar crs " + output_lib + " $(find " + search_path +
+          "/obj.target -name *\.o)"], shell=True)
+    call(["ar crs " + output_lib + " $(find " + search_path +
+          "/obj/gen -name *\.o)"], shell=True)
+
+  elif sys.platform == 'darwin':
+    call(["rm -f " + output_lib], shell=True)
+    call(["rm -f " + search_path + "/*do_not_use*"], shell=True)
+    call(["libtool -static -v -o " + output_lib + " " + search_path + "/*.a"],
+         shell=True)
+
+  elif sys.platform == 'win32':
+    # We need to execute a batch file to set some environment variables for the
+    # lib command. VS 8 uses vsvars.bat and VS 9 uses vsvars32.bat. It's
+    # required that at least one of them is in the system PATH. We try both and
+    # suppress stderr and stdout to fail silently.
+    call(["vsvars.bat"], stderr=PIPE, stdout=PIPE, shell=True)
+    call(["vsvars32.bat"], stderr=PIPE, stdout=PIPE, shell=True)
+    call(["del " + output_lib], shell=True)
+    call(["del /F /S /Q " + search_path + "/lib/*do_not_use*"],
+          shell=True)
+    call(["lib /OUT:" + output_lib + " " + search_path + "/lib/*.lib"],
+         shell=True)
+
+  else:
+    sys.stderr.write('Platform not supported: %r\n\n' % sys.platform)
+    sys.exit(1)
+
+  sys.exit(0)
+
diff --git a/trunk/src/build/no_op.cc b/trunk/src/build/no_op.cc
new file mode 100644
index 0000000..4cce7f6
--- /dev/null
+++ b/trunk/src/build/no_op.cc
@@ -0,0 +1,3 @@
+int main() {
+  return 0;
+}
diff --git a/trunk/src/build/protoc.gypi b/trunk/src/build/protoc.gypi
new file mode 100644
index 0000000..70bf71e
--- /dev/null
+++ b/trunk/src/build/protoc.gypi
@@ -0,0 +1,97 @@
+# Copyright (c) 2011 The Chromium Authors. All rights reserved.
+# Use of this source code is governed by a BSD-style license that can be
+# found in the LICENSE file.
+
+# It was necessary to copy this file to WebRTC, because the path to
+# build/common.gypi is different for the standalone and Chromium builds. Gyp
+# doesn't permit conditional inclusion or variable expansion in include paths.
+# http://code.google.com/p/gyp/wiki/InputFormatReference#Including_Other_Files
+
+# This file is meant to be included into a target to provide a rule
+# to invoke protoc in a consistent manner.
+#
+# To use this, create a gyp target with the following form:
+# {
+#   'target_name': 'my_proto_lib',
+#   'type': 'static_library',
+#   'sources': [
+#     'foo.proto',
+#     'bar.proto',
+#   ],
+#   'variables': {
+#     # Optional, see below: 'proto_in_dir': '.'
+#     'proto_out_dir': 'dir/for/my_proto_lib'
+#   },
+#   'includes': ['path/to/this/gypi/file'],
+# }
+# If necessary, you may add normal .cc files to the sources list or other gyp
+# dependencies.  The proto headers are guaranteed to be generated before any
+# source files, even within this target, are compiled.
+#
+# The 'proto_in_dir' variable must be the relative path to the
+# directory containing the .proto files.  If left out, it defaults to '.'.
+#
+# The 'proto_out_dir' variable specifies the path suffix that output
+# files are generated under.  Targets that gyp-depend on my_proto_lib
+# will be able to include the resulting proto headers with an include
+# like:
+#   #include "dir/for/my_proto_lib/foo.pb.h"
+#
+# Implementation notes:
+# A proto_out_dir of foo/bar produces
+#   <(SHARED_INTERMEDIATE_DIR)/protoc_out/foo/bar/{file1,file2}.pb.{cc,h}
+#   <(SHARED_INTERMEDIATE_DIR)/pyproto/foo/bar/{file1,file2}_pb2.py
+
+{
+  'variables': {
+    'protoc': '<(PRODUCT_DIR)/<(EXECUTABLE_PREFIX)protoc<(EXECUTABLE_SUFFIX)',
+    'cc_dir': '<(SHARED_INTERMEDIATE_DIR)/protoc_out/<(proto_out_dir)',
+    'py_dir': '<(PRODUCT_DIR)/pyproto/<(proto_out_dir)',
+    'proto_in_dir%': '.',
+  },
+  'rules': [
+    {
+      'rule_name': 'genproto',
+      'extension': 'proto',
+      'inputs': [
+        '<(protoc)',
+      ],
+      'outputs': [
+        '<(py_dir)/<(RULE_INPUT_ROOT)_pb2.py',
+        '<(cc_dir)/<(RULE_INPUT_ROOT).pb.cc',
+        '<(cc_dir)/<(RULE_INPUT_ROOT).pb.h',
+      ],
+      'action': [
+        '<(protoc)',
+        '--proto_path=<(proto_in_dir)',
+        # Naively you'd use <(RULE_INPUT_PATH) here, but protoc requires
+        # --proto_path is a strict prefix of the path given as an argument.
+        '<(proto_in_dir)/<(RULE_INPUT_ROOT)<(RULE_INPUT_EXT)',
+        '--cpp_out=<(cc_dir)',
+        '--python_out=<(py_dir)',
+        ],
+      'message': 'Generating C++ and Python code from <(RULE_INPUT_PATH)',
+      'process_outputs_as_sources': 1,
+    },
+  ],
+  'dependencies': [
+    '<(DEPTH)/third_party/protobuf/protobuf.gyp:protoc#host',
+    '<(DEPTH)/third_party/protobuf/protobuf.gyp:protobuf_lite',
+  ],
+  'include_dirs': [
+    '<(SHARED_INTERMEDIATE_DIR)/protoc_out',
+  ],
+  'direct_dependent_settings': {
+    'include_dirs': [
+      '<(SHARED_INTERMEDIATE_DIR)/protoc_out',
+    ]
+  },
+  'export_dependent_settings': [
+    # The generated headers reference headers within protobuf_lite,
+    # so dependencies must be able to find those headers too.
+    '<(DEPTH)/third_party/protobuf/protobuf.gyp:protobuf_lite',
+  ],
+  # This target exports a hard dependency because it generates header
+  # files.
+  'hard_dependency': 1,
+}
diff --git a/trunk/src/common_audio/OWNERS b/trunk/src/common_audio/OWNERS
new file mode 100644
index 0000000..84582f2
--- /dev/null
+++ b/trunk/src/common_audio/OWNERS
@@ -0,0 +1,4 @@
+bjornv@webrtc.org
+tina.legrand@webrtc.org
+jan.skoglund@webrtc.org
+andrew@webrtc.org
diff --git a/trunk/src/common_audio/common_audio.gyp b/trunk/src/common_audio/common_audio.gyp
new file mode 100644
index 0000000..3d3da3f
--- /dev/null
+++ b/trunk/src/common_audio/common_audio.gyp
@@ -0,0 +1,16 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+  'includes': [
+    '../build/common.gypi',
+    'signal_processing/signal_processing.gypi',
+    'resampler/resampler.gypi',
+    'vad/vad.gypi',
+  ],
+}
diff --git a/trunk/src/common_audio/resampler/Android.mk b/trunk/src/common_audio/resampler/Android.mk
new file mode 100644
index 0000000..b1d630a
--- /dev/null
+++ b/trunk/src/common_audio/resampler/Android.mk
@@ -0,0 +1,47 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+include $(LOCAL_PATH)/../../../android-webrtc.mk
+
+LOCAL_ARM_MODE := arm
+LOCAL_MODULE_CLASS := STATIC_LIBRARIES
+LOCAL_MODULE := libwebrtc_resampler
+LOCAL_MODULE_TAGS := optional
+LOCAL_CPP_EXTENSION := .cc
+LOCAL_SRC_FILES := resampler.cc
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+    $(MY_WEBRTC_COMMON_DEFS)
+
+LOCAL_C_INCLUDES := \
+    $(LOCAL_PATH)/include \
+    $(LOCAL_PATH)/../.. \
+    $(LOCAL_PATH)/../signal_processing/include 
+
+LOCAL_SHARED_LIBRARIES := \
+    libcutils \
+    libdl \
+    libstlport
+
+ifeq ($(TARGET_OS)-$(TARGET_SIMULATOR),linux-true)
+LOCAL_LDLIBS += -ldl -lpthread
+endif
+
+ifneq ($(TARGET_SIMULATOR),true)
+LOCAL_SHARED_LIBRARIES += libdl
+endif
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/common_audio/resampler/include/resampler.h b/trunk/src/common_audio/resampler/include/resampler.h
new file mode 100644
index 0000000..38e6bd3
--- /dev/null
+++ b/trunk/src/common_audio/resampler/include/resampler.h
@@ -0,0 +1,116 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * A wrapper for resampling a numerous amount of sampling combinations.
+ */
+
+#ifndef WEBRTC_RESAMPLER_RESAMPLER_H_
+#define WEBRTC_RESAMPLER_RESAMPLER_H_
+
+#include "typedefs.h"
+
+namespace webrtc
+{
+
+// TODO(andrew): the implementation depends on the exact values of this enum.
+// It should be rewritten in a less fragile way.
+enum ResamplerType
+{
+    // 4 MSB = Number of channels
+    // 4 LSB = Synchronous or asynchronous
+
+    kResamplerSynchronous = 0x10,
+    kResamplerAsynchronous = 0x11,
+    kResamplerSynchronousStereo = 0x20,
+    kResamplerAsynchronousStereo = 0x21,
+    kResamplerInvalid = 0xff
+};
+
+// TODO(andrew): doesn't need to be part of the interface.
+enum ResamplerMode
+{
+    kResamplerMode1To1,
+    kResamplerMode1To2,
+    kResamplerMode1To3,
+    kResamplerMode1To4,
+    kResamplerMode1To6,
+    kResamplerMode1To12,
+    kResamplerMode2To3,
+    kResamplerMode2To11,
+    kResamplerMode4To11,
+    kResamplerMode8To11,
+    kResamplerMode11To16,
+    kResamplerMode11To32,
+    kResamplerMode2To1,
+    kResamplerMode3To1,
+    kResamplerMode4To1,
+    kResamplerMode6To1,
+    kResamplerMode12To1,
+    kResamplerMode3To2,
+    kResamplerMode11To2,
+    kResamplerMode11To4,
+    kResamplerMode11To8
+};
+
+class Resampler
+{
+
+public:
+    Resampler();
+    // TODO(andrew): use an init function instead.
+    Resampler(int inFreq, int outFreq, ResamplerType type);
+    ~Resampler();
+
+    // Reset all states
+    int Reset(int inFreq, int outFreq, ResamplerType type);
+
+    // Reset all states if any parameter has changed
+    int ResetIfNeeded(int inFreq, int outFreq, ResamplerType type);
+
+    // Synchronous resampling, all output samples are written to samplesOut
+    int Push(const WebRtc_Word16* samplesIn, int lengthIn, WebRtc_Word16* samplesOut,
+             int maxLen, int &outLen);
+
+    // Asynchronous resampling, input
+    int Insert(WebRtc_Word16* samplesIn, int lengthIn);
+
+    // Asynchronous resampling output, remaining samples are buffered
+    int Pull(WebRtc_Word16* samplesOut, int desiredLen, int &outLen);
+
+private:
+    // Generic pointers since we don't know what states we'll need
+    void* state1_;
+    void* state2_;
+    void* state3_;
+
+    // Storage if needed
+    WebRtc_Word16* in_buffer_;
+    WebRtc_Word16* out_buffer_;
+    int in_buffer_size_;
+    int out_buffer_size_;
+    int in_buffer_size_max_;
+    int out_buffer_size_max_;
+
+    // State
+    int my_in_frequency_khz_;
+    int my_out_frequency_khz_;
+    ResamplerMode my_mode_;
+    ResamplerType my_type_;
+
+    // Extra instance for stereo
+    Resampler* slave_left_;
+    Resampler* slave_right_;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_RESAMPLER_RESAMPLER_H_
diff --git a/trunk/src/common_audio/resampler/resampler.cc b/trunk/src/common_audio/resampler/resampler.cc
new file mode 100644
index 0000000..2db27b1
--- /dev/null
+++ b/trunk/src/common_audio/resampler/resampler.cc
@@ -0,0 +1,1084 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * A wrapper for resampling a numerous amount of sampling combinations.
+ */
+
+#include <stdlib.h>
+#include <string.h>
+
+#include "signal_processing_library.h"
+#include "resampler.h"
+
+
+namespace webrtc
+{
+
+Resampler::Resampler()
+{
+    state1_ = NULL;
+    state2_ = NULL;
+    state3_ = NULL;
+    in_buffer_ = NULL;
+    out_buffer_ = NULL;
+    in_buffer_size_ = 0;
+    out_buffer_size_ = 0;
+    in_buffer_size_max_ = 0;
+    out_buffer_size_max_ = 0;
+    // we need a reset before we will work
+    my_in_frequency_khz_ = 0;
+    my_out_frequency_khz_ = 0;
+    my_mode_ = kResamplerMode1To1;
+    my_type_ = kResamplerInvalid;
+    slave_left_ = NULL;
+    slave_right_ = NULL;
+}
+
+Resampler::Resampler(int inFreq, int outFreq, ResamplerType type)
+{
+    state1_ = NULL;
+    state2_ = NULL;
+    state3_ = NULL;
+    in_buffer_ = NULL;
+    out_buffer_ = NULL;
+    in_buffer_size_ = 0;
+    out_buffer_size_ = 0;
+    in_buffer_size_max_ = 0;
+    out_buffer_size_max_ = 0;
+    // we need a reset before we will work
+    my_in_frequency_khz_ = 0;
+    my_out_frequency_khz_ = 0;
+    my_mode_ = kResamplerMode1To1;
+    my_type_ = kResamplerInvalid;
+    slave_left_ = NULL;
+    slave_right_ = NULL;
+
+    Reset(inFreq, outFreq, type);
+}
+
+Resampler::~Resampler()
+{
+    if (state1_)
+    {
+        free(state1_);
+    }
+    if (state2_)
+    {
+        free(state2_);
+    }
+    if (state3_)
+    {
+        free(state3_);
+    }
+    if (in_buffer_)
+    {
+        free(in_buffer_);
+    }
+    if (out_buffer_)
+    {
+        free(out_buffer_);
+    }
+    if (slave_left_)
+    {
+        delete slave_left_;
+    }
+    if (slave_right_)
+    {
+        delete slave_right_;
+    }
+}
+
+int Resampler::ResetIfNeeded(int inFreq, int outFreq, ResamplerType type)
+{
+    int tmpInFreq_kHz = inFreq / 1000;
+    int tmpOutFreq_kHz = outFreq / 1000;
+
+    if ((tmpInFreq_kHz != my_in_frequency_khz_) || (tmpOutFreq_kHz != my_out_frequency_khz_)
+            || (type != my_type_))
+    {
+        return Reset(inFreq, outFreq, type);
+    } else
+    {
+        return 0;
+    }
+}
+
+int Resampler::Reset(int inFreq, int outFreq, ResamplerType type)
+{
+
+    if (state1_)
+    {
+        free(state1_);
+        state1_ = NULL;
+    }
+    if (state2_)
+    {
+        free(state2_);
+        state2_ = NULL;
+    }
+    if (state3_)
+    {
+        free(state3_);
+        state3_ = NULL;
+    }
+    if (in_buffer_)
+    {
+        free(in_buffer_);
+        in_buffer_ = NULL;
+    }
+    if (out_buffer_)
+    {
+        free(out_buffer_);
+        out_buffer_ = NULL;
+    }
+    if (slave_left_)
+    {
+        delete slave_left_;
+        slave_left_ = NULL;
+    }
+    if (slave_right_)
+    {
+        delete slave_right_;
+        slave_right_ = NULL;
+    }
+
+    in_buffer_size_ = 0;
+    out_buffer_size_ = 0;
+    in_buffer_size_max_ = 0;
+    out_buffer_size_max_ = 0;
+
+    // This might be overridden if parameters are not accepted.
+    my_type_ = type;
+
+    // Start with a math exercise, Euclid's algorithm to find the gcd:
+
+    int a = inFreq;
+    int b = outFreq;
+    int c = a % b;
+    while (c != 0)
+    {
+        a = b;
+        b = c;
+        c = a % b;
+    }
+    // b is now the gcd;
+
+    // We need to track what domain we're in.
+    my_in_frequency_khz_ = inFreq / 1000;
+    my_out_frequency_khz_ = outFreq / 1000;
+
+    // Scale with GCD
+    inFreq = inFreq / b;
+    outFreq = outFreq / b;
+
+    // Do we need stereo?
+    if ((my_type_ & 0xf0) == 0x20)
+    {
+        // Change type to mono
+        type = static_cast<ResamplerType>(
+            ((static_cast<int>(type) & 0x0f) + 0x10));
+        slave_left_ = new Resampler(inFreq, outFreq, type);
+        slave_right_ = new Resampler(inFreq, outFreq, type);
+    }
+
+    if (inFreq == outFreq)
+    {
+        my_mode_ = kResamplerMode1To1;
+    } else if (inFreq == 1)
+    {
+        switch (outFreq)
+        {
+            case 2:
+                my_mode_ = kResamplerMode1To2;
+                break;
+            case 3:
+                my_mode_ = kResamplerMode1To3;
+                break;
+            case 4:
+                my_mode_ = kResamplerMode1To4;
+                break;
+            case 6:
+                my_mode_ = kResamplerMode1To6;
+                break;
+            case 12:
+                my_mode_ = kResamplerMode1To12;
+                break;
+            default:
+                my_type_ = kResamplerInvalid;
+                return -1;
+        }
+    } else if (outFreq == 1)
+    {
+        switch (inFreq)
+        {
+            case 2:
+                my_mode_ = kResamplerMode2To1;
+                break;
+            case 3:
+                my_mode_ = kResamplerMode3To1;
+                break;
+            case 4:
+                my_mode_ = kResamplerMode4To1;
+                break;
+            case 6:
+                my_mode_ = kResamplerMode6To1;
+                break;
+            case 12:
+                my_mode_ = kResamplerMode12To1;
+                break;
+            default:
+                my_type_ = kResamplerInvalid;
+                return -1;
+        }
+    } else if ((inFreq == 2) && (outFreq == 3))
+    {
+        my_mode_ = kResamplerMode2To3;
+    } else if ((inFreq == 2) && (outFreq == 11))
+    {
+        my_mode_ = kResamplerMode2To11;
+    } else if ((inFreq == 4) && (outFreq == 11))
+    {
+        my_mode_ = kResamplerMode4To11;
+    } else if ((inFreq == 8) && (outFreq == 11))
+    {
+        my_mode_ = kResamplerMode8To11;
+    } else if ((inFreq == 3) && (outFreq == 2))
+    {
+        my_mode_ = kResamplerMode3To2;
+    } else if ((inFreq == 11) && (outFreq == 2))
+    {
+        my_mode_ = kResamplerMode11To2;
+    } else if ((inFreq == 11) && (outFreq == 4))
+    {
+        my_mode_ = kResamplerMode11To4;
+    } else if ((inFreq == 11) && (outFreq == 16))
+    {
+        my_mode_ = kResamplerMode11To16;
+    } else if ((inFreq == 11) && (outFreq == 32))
+    {
+        my_mode_ = kResamplerMode11To32;
+    } else if ((inFreq == 11) && (outFreq == 8))
+    {
+        my_mode_ = kResamplerMode11To8;
+    } else
+    {
+        my_type_ = kResamplerInvalid;
+        return -1;
+    }
+
+    // Now create the states we need
+    switch (my_mode_)
+    {
+        case kResamplerMode1To1:
+            // No state needed;
+            break;
+        case kResamplerMode1To2:
+            state1_ = malloc(8 * sizeof(WebRtc_Word32));
+            memset(state1_, 0, 8 * sizeof(WebRtc_Word32));
+            break;
+        case kResamplerMode1To3:
+            state1_ = malloc(sizeof(WebRtcSpl_State16khzTo48khz));
+            WebRtcSpl_ResetResample16khzTo48khz((WebRtcSpl_State16khzTo48khz *)state1_);
+            break;
+        case kResamplerMode1To4:
+            // 1:2
+            state1_ = malloc(8 * sizeof(WebRtc_Word32));
+            memset(state1_, 0, 8 * sizeof(WebRtc_Word32));
+            // 2:4
+            state2_ = malloc(8 * sizeof(WebRtc_Word32));
+            memset(state2_, 0, 8 * sizeof(WebRtc_Word32));
+            break;
+        case kResamplerMode1To6:
+            // 1:2
+            state1_ = malloc(8 * sizeof(WebRtc_Word32));
+            memset(state1_, 0, 8 * sizeof(WebRtc_Word32));
+            // 2:6
+            state2_ = malloc(sizeof(WebRtcSpl_State16khzTo48khz));
+            WebRtcSpl_ResetResample16khzTo48khz((WebRtcSpl_State16khzTo48khz *)state2_);
+            break;
+        case kResamplerMode1To12:
+            // 1:2
+            state1_ = malloc(8 * sizeof(WebRtc_Word32));
+            memset(state1_, 0, 8 * sizeof(WebRtc_Word32));
+            // 2:4
+            state2_ = malloc(8 * sizeof(WebRtc_Word32));
+            memset(state2_, 0, 8 * sizeof(WebRtc_Word32));
+            // 4:12
+            state3_ = malloc(sizeof(WebRtcSpl_State16khzTo48khz));
+            WebRtcSpl_ResetResample16khzTo48khz(
+                (WebRtcSpl_State16khzTo48khz*) state3_);
+            break;
+        case kResamplerMode2To3:
+            // 2:6
+            state1_ = malloc(sizeof(WebRtcSpl_State16khzTo48khz));
+            WebRtcSpl_ResetResample16khzTo48khz((WebRtcSpl_State16khzTo48khz *)state1_);
+            // 6:3
+            state2_ = malloc(8 * sizeof(WebRtc_Word32));
+            memset(state2_, 0, 8 * sizeof(WebRtc_Word32));
+            break;
+        case kResamplerMode2To11:
+            state1_ = malloc(8 * sizeof(WebRtc_Word32));
+            memset(state1_, 0, 8 * sizeof(WebRtc_Word32));
+
+            state2_ = malloc(sizeof(WebRtcSpl_State8khzTo22khz));
+            WebRtcSpl_ResetResample8khzTo22khz((WebRtcSpl_State8khzTo22khz *)state2_);
+            break;
+        case kResamplerMode4To11:
+            state1_ = malloc(sizeof(WebRtcSpl_State8khzTo22khz));
+            WebRtcSpl_ResetResample8khzTo22khz((WebRtcSpl_State8khzTo22khz *)state1_);
+            break;
+        case kResamplerMode8To11:
+            state1_ = malloc(sizeof(WebRtcSpl_State16khzTo22khz));
+            WebRtcSpl_ResetResample16khzTo22khz((WebRtcSpl_State16khzTo22khz *)state1_);
+            break;
+        case kResamplerMode11To16:
+            state1_ = malloc(8 * sizeof(WebRtc_Word32));
+            memset(state1_, 0, 8 * sizeof(WebRtc_Word32));
+
+            state2_ = malloc(sizeof(WebRtcSpl_State22khzTo16khz));
+            WebRtcSpl_ResetResample22khzTo16khz((WebRtcSpl_State22khzTo16khz *)state2_);
+            break;
+        case kResamplerMode11To32:
+            // 11 -> 22
+            state1_ = malloc(8 * sizeof(WebRtc_Word32));
+            memset(state1_, 0, 8 * sizeof(WebRtc_Word32));
+
+            // 22 -> 16
+            state2_ = malloc(sizeof(WebRtcSpl_State22khzTo16khz));
+            WebRtcSpl_ResetResample22khzTo16khz((WebRtcSpl_State22khzTo16khz *)state2_);
+
+            // 16 -> 32
+            state3_ = malloc(8 * sizeof(WebRtc_Word32));
+            memset(state3_, 0, 8 * sizeof(WebRtc_Word32));
+
+            break;
+        case kResamplerMode2To1:
+            state1_ = malloc(8 * sizeof(WebRtc_Word32));
+            memset(state1_, 0, 8 * sizeof(WebRtc_Word32));
+            break;
+        case kResamplerMode3To1:
+            state1_ = malloc(sizeof(WebRtcSpl_State48khzTo16khz));
+            WebRtcSpl_ResetResample48khzTo16khz((WebRtcSpl_State48khzTo16khz *)state1_);
+            break;
+        case kResamplerMode4To1:
+            // 4:2
+            state1_ = malloc(8 * sizeof(WebRtc_Word32));
+            memset(state1_, 0, 8 * sizeof(WebRtc_Word32));
+            // 2:1
+            state2_ = malloc(8 * sizeof(WebRtc_Word32));
+            memset(state2_, 0, 8 * sizeof(WebRtc_Word32));
+            break;
+        case kResamplerMode6To1:
+            // 6:2
+            state1_ = malloc(sizeof(WebRtcSpl_State48khzTo16khz));
+            WebRtcSpl_ResetResample48khzTo16khz((WebRtcSpl_State48khzTo16khz *)state1_);
+            // 2:1
+            state2_ = malloc(8 * sizeof(WebRtc_Word32));
+            memset(state2_, 0, 8 * sizeof(WebRtc_Word32));
+            break;
+        case kResamplerMode12To1:
+            // 12:4
+            state1_ = malloc(sizeof(WebRtcSpl_State48khzTo16khz));
+            WebRtcSpl_ResetResample48khzTo16khz(
+                (WebRtcSpl_State48khzTo16khz*) state1_);
+            // 4:2
+            state2_ = malloc(8 * sizeof(WebRtc_Word32));
+            memset(state2_, 0, 8 * sizeof(WebRtc_Word32));
+            // 2:1
+            state3_ = malloc(8 * sizeof(WebRtc_Word32));
+            memset(state3_, 0, 8 * sizeof(WebRtc_Word32));
+            break;
+        case kResamplerMode3To2:
+            // 3:6
+            state1_ = malloc(8 * sizeof(WebRtc_Word32));
+            memset(state1_, 0, 8 * sizeof(WebRtc_Word32));
+            // 6:2
+            state2_ = malloc(sizeof(WebRtcSpl_State48khzTo16khz));
+            WebRtcSpl_ResetResample48khzTo16khz((WebRtcSpl_State48khzTo16khz *)state2_);
+            break;
+        case kResamplerMode11To2:
+            state1_ = malloc(sizeof(WebRtcSpl_State22khzTo8khz));
+            WebRtcSpl_ResetResample22khzTo8khz((WebRtcSpl_State22khzTo8khz *)state1_);
+
+            state2_ = malloc(8 * sizeof(WebRtc_Word32));
+            memset(state2_, 0, 8 * sizeof(WebRtc_Word32));
+
+            break;
+        case kResamplerMode11To4:
+            state1_ = malloc(sizeof(WebRtcSpl_State22khzTo8khz));
+            WebRtcSpl_ResetResample22khzTo8khz((WebRtcSpl_State22khzTo8khz *)state1_);
+            break;
+        case kResamplerMode11To8:
+            state1_ = malloc(sizeof(WebRtcSpl_State22khzTo16khz));
+            WebRtcSpl_ResetResample22khzTo16khz((WebRtcSpl_State22khzTo16khz *)state1_);
+            break;
+
+    }
+
+    return 0;
+}
+
+// Synchronous resampling, all output samples are written to samplesOut
+int Resampler::Push(const WebRtc_Word16 * samplesIn, int lengthIn, WebRtc_Word16* samplesOut,
+                    int maxLen, int &outLen)
+{
+    // Check that the resampler is not in asynchronous mode
+    if (my_type_ & 0x0f)
+    {
+        return -1;
+    }
+
+    // Do we have a stereo signal?
+    if ((my_type_ & 0xf0) == 0x20)
+    {
+
+        // Split up the signal and call the slave object for each channel
+
+        WebRtc_Word16* left = (WebRtc_Word16*)malloc(lengthIn * sizeof(WebRtc_Word16) / 2);
+        WebRtc_Word16* right = (WebRtc_Word16*)malloc(lengthIn * sizeof(WebRtc_Word16) / 2);
+        WebRtc_Word16* out_left = (WebRtc_Word16*)malloc(maxLen / 2 * sizeof(WebRtc_Word16));
+        WebRtc_Word16* out_right =
+                (WebRtc_Word16*)malloc(maxLen / 2 * sizeof(WebRtc_Word16));
+        int res = 0;
+        for (int i = 0; i < lengthIn; i += 2)
+        {
+            left[i >> 1] = samplesIn[i];
+            right[i >> 1] = samplesIn[i + 1];
+        }
+
+        // It's OK to overwrite the local parameter, since it's just a copy
+        lengthIn = lengthIn / 2;
+
+        int actualOutLen_left = 0;
+        int actualOutLen_right = 0;
+        // Do resampling for right channel
+        res |= slave_left_->Push(left, lengthIn, out_left, maxLen / 2, actualOutLen_left);
+        res |= slave_right_->Push(right, lengthIn, out_right, maxLen / 2, actualOutLen_right);
+        if (res || (actualOutLen_left != actualOutLen_right))
+        {
+            free(left);
+            free(right);
+            free(out_left);
+            free(out_right);
+            return -1;
+        }
+
+        // Reassemble the signal
+        for (int i = 0; i < actualOutLen_left; i++)
+        {
+            samplesOut[i * 2] = out_left[i];
+            samplesOut[i * 2 + 1] = out_right[i];
+        }
+        outLen = 2 * actualOutLen_left;
+
+        free(left);
+        free(right);
+        free(out_left);
+        free(out_right);
+
+        return 0;
+    }
+
+    // Containers for temp samples
+    WebRtc_Word16* tmp;
+    WebRtc_Word16* tmp_2;
+    // tmp data for resampling routines
+    WebRtc_Word32* tmp_mem;
+
+    switch (my_mode_)
+    {
+        case kResamplerMode1To1:
+            memcpy(samplesOut, samplesIn, lengthIn * sizeof(WebRtc_Word16));
+            outLen = lengthIn;
+            break;
+        case kResamplerMode1To2:
+            if (maxLen < (lengthIn * 2))
+            {
+                return -1;
+            }
+            WebRtcSpl_UpsampleBy2(samplesIn, lengthIn, samplesOut, (WebRtc_Word32*)state1_);
+            outLen = lengthIn * 2;
+            return 0;
+        case kResamplerMode1To3:
+
+            // We can only handle blocks of 160 samples
+            // Can be fixed, but I don't think it's needed
+            if ((lengthIn % 160) != 0)
+            {
+                return -1;
+            }
+            if (maxLen < (lengthIn * 3))
+            {
+                return -1;
+            }
+            tmp_mem = (WebRtc_Word32*)malloc(336 * sizeof(WebRtc_Word32));
+
+            for (int i = 0; i < lengthIn; i += 160)
+            {
+                WebRtcSpl_Resample16khzTo48khz(samplesIn + i, samplesOut + i * 3,
+                                               (WebRtcSpl_State16khzTo48khz *)state1_,
+                                               tmp_mem);
+            }
+            outLen = lengthIn * 3;
+            free(tmp_mem);
+            return 0;
+        case kResamplerMode1To4:
+            if (maxLen < (lengthIn * 4))
+            {
+                return -1;
+            }
+
+            tmp = (WebRtc_Word16*)malloc(sizeof(WebRtc_Word16) * 2 * lengthIn);
+            // 1:2
+            WebRtcSpl_UpsampleBy2(samplesIn, lengthIn, tmp, (WebRtc_Word32*)state1_);
+            // 2:4
+            WebRtcSpl_UpsampleBy2(tmp, lengthIn * 2, samplesOut, (WebRtc_Word32*)state2_);
+            outLen = lengthIn * 4;
+            free(tmp);
+            return 0;
+        case kResamplerMode1To6:
+            // We can only handle blocks of 80 samples
+            // Can be fixed, but I don't think it's needed
+            if ((lengthIn % 80) != 0)
+            {
+                return -1;
+            }
+            if (maxLen < (lengthIn * 6))
+            {
+                return -1;
+            }
+
+            //1:2
+
+            tmp_mem = (WebRtc_Word32*)malloc(336 * sizeof(WebRtc_Word32));
+            tmp = (WebRtc_Word16*)malloc(sizeof(WebRtc_Word16) * 2 * lengthIn);
+
+            WebRtcSpl_UpsampleBy2(samplesIn, lengthIn, tmp, (WebRtc_Word32*)state1_);
+            outLen = lengthIn * 2;
+
+            for (int i = 0; i < outLen; i += 160)
+            {
+                WebRtcSpl_Resample16khzTo48khz(tmp + i, samplesOut + i * 3,
+                                               (WebRtcSpl_State16khzTo48khz *)state2_,
+                                               tmp_mem);
+            }
+            outLen = outLen * 3;
+            free(tmp_mem);
+            free(tmp);
+
+            return 0;
+        case kResamplerMode1To12:
+            // We can only handle blocks of 40 samples
+            // Can be fixed, but I don't think it's needed
+            if ((lengthIn % 40) != 0) {
+              return -1;
+            }
+            if (maxLen < (lengthIn * 12)) {
+              return -1;
+            }
+
+            tmp_mem = (WebRtc_Word32*) malloc(336 * sizeof(WebRtc_Word32));
+            tmp = (WebRtc_Word16*) malloc(sizeof(WebRtc_Word16) * 4 * lengthIn);
+            //1:2
+            WebRtcSpl_UpsampleBy2(samplesIn, lengthIn, samplesOut,
+                                  (WebRtc_Word32*) state1_);
+            outLen = lengthIn * 2;
+            //2:4
+            WebRtcSpl_UpsampleBy2(samplesOut, outLen, tmp, (WebRtc_Word32*) state2_);
+            outLen = outLen * 2;
+            // 4:12
+            for (int i = 0; i < outLen; i += 160) {
+              // WebRtcSpl_Resample16khzTo48khz() takes a block of 160 samples
+              // as input and outputs a resampled block of 480 samples. The
+              // data is now actually in 32 kHz sampling rate, despite the
+              // function name, and with a resampling factor of three becomes
+              // 96 kHz.
+              WebRtcSpl_Resample16khzTo48khz(tmp + i, samplesOut + i * 3,
+                                             (WebRtcSpl_State16khzTo48khz*) state3_,
+                                             tmp_mem);
+            }
+            outLen = outLen * 3;
+            free(tmp_mem);
+            free(tmp);
+
+            return 0;
+        case kResamplerMode2To3:
+            if (maxLen < (lengthIn * 3 / 2))
+            {
+                return -1;
+            }
+            // 2:6
+            // We can only handle blocks of 160 samples
+            // Can be fixed, but I don't think it's needed
+            if ((lengthIn % 160) != 0)
+            {
+                return -1;
+            }
+            tmp = static_cast<WebRtc_Word16*> (malloc(sizeof(WebRtc_Word16) * lengthIn * 3));
+            tmp_mem = (WebRtc_Word32*)malloc(336 * sizeof(WebRtc_Word32));
+            for (int i = 0; i < lengthIn; i += 160)
+            {
+                WebRtcSpl_Resample16khzTo48khz(samplesIn + i, tmp + i * 3,
+                                               (WebRtcSpl_State16khzTo48khz *)state1_,
+                                               tmp_mem);
+            }
+            lengthIn = lengthIn * 3;
+            // 6:3
+            WebRtcSpl_DownsampleBy2(tmp, lengthIn, samplesOut, (WebRtc_Word32*)state2_);
+            outLen = lengthIn / 2;
+            free(tmp);
+            free(tmp_mem);
+            return 0;
+        case kResamplerMode2To11:
+
+            // We can only handle blocks of 80 samples
+            // Can be fixed, but I don't think it's needed
+            if ((lengthIn % 80) != 0)
+            {
+                return -1;
+            }
+            if (maxLen < ((lengthIn * 11) / 2))
+            {
+                return -1;
+            }
+            tmp = (WebRtc_Word16*)malloc(sizeof(WebRtc_Word16) * 2 * lengthIn);
+            // 1:2
+            WebRtcSpl_UpsampleBy2(samplesIn, lengthIn, tmp, (WebRtc_Word32*)state1_);
+            lengthIn *= 2;
+
+            tmp_mem = (WebRtc_Word32*)malloc(98 * sizeof(WebRtc_Word32));
+
+            for (int i = 0; i < lengthIn; i += 80)
+            {
+                WebRtcSpl_Resample8khzTo22khz(tmp + i, samplesOut + (i * 11) / 4,
+                                              (WebRtcSpl_State8khzTo22khz *)state2_,
+                                              tmp_mem);
+            }
+            outLen = (lengthIn * 11) / 4;
+            free(tmp_mem);
+            free(tmp);
+            return 0;
+        case kResamplerMode4To11:
+
+            // We can only handle blocks of 80 samples
+            // Can be fixed, but I don't think it's needed
+            if ((lengthIn % 80) != 0)
+            {
+                return -1;
+            }
+            if (maxLen < ((lengthIn * 11) / 4))
+            {
+                return -1;
+            }
+            tmp_mem = (WebRtc_Word32*)malloc(98 * sizeof(WebRtc_Word32));
+
+            for (int i = 0; i < lengthIn; i += 80)
+            {
+                WebRtcSpl_Resample8khzTo22khz(samplesIn + i, samplesOut + (i * 11) / 4,
+                                              (WebRtcSpl_State8khzTo22khz *)state1_,
+                                              tmp_mem);
+            }
+            outLen = (lengthIn * 11) / 4;
+            free(tmp_mem);
+            return 0;
+        case kResamplerMode8To11:
+            // We can only handle blocks of 160 samples
+            // Can be fixed, but I don't think it's needed
+            if ((lengthIn % 160) != 0)
+            {
+                return -1;
+            }
+            if (maxLen < ((lengthIn * 11) / 8))
+            {
+                return -1;
+            }
+            tmp_mem = (WebRtc_Word32*)malloc(88 * sizeof(WebRtc_Word32));
+
+            for (int i = 0; i < lengthIn; i += 160)
+            {
+                WebRtcSpl_Resample16khzTo22khz(samplesIn + i, samplesOut + (i * 11) / 8,
+                                               (WebRtcSpl_State16khzTo22khz *)state1_,
+                                               tmp_mem);
+            }
+            outLen = (lengthIn * 11) / 8;
+            free(tmp_mem);
+            return 0;
+
+        case kResamplerMode11To16:
+            // We can only handle blocks of 110 samples
+            if ((lengthIn % 110) != 0)
+            {
+                return -1;
+            }
+            if (maxLen < ((lengthIn * 16) / 11))
+            {
+                return -1;
+            }
+
+            tmp_mem = (WebRtc_Word32*)malloc(104 * sizeof(WebRtc_Word32));
+            tmp = (WebRtc_Word16*)malloc((sizeof(WebRtc_Word16) * lengthIn * 2));
+
+            WebRtcSpl_UpsampleBy2(samplesIn, lengthIn, tmp, (WebRtc_Word32*)state1_);
+
+            for (int i = 0; i < (lengthIn * 2); i += 220)
+            {
+                WebRtcSpl_Resample22khzTo16khz(tmp + i, samplesOut + (i / 220) * 160,
+                                               (WebRtcSpl_State22khzTo16khz *)state2_,
+                                               tmp_mem);
+            }
+
+            outLen = (lengthIn * 16) / 11;
+
+            free(tmp_mem);
+            free(tmp);
+            return 0;
+
+        case kResamplerMode11To32:
+
+            // We can only handle blocks of 110 samples
+            if ((lengthIn % 110) != 0)
+            {
+                return -1;
+            }
+            if (maxLen < ((lengthIn * 32) / 11))
+            {
+                return -1;
+            }
+
+            tmp_mem = (WebRtc_Word32*)malloc(104 * sizeof(WebRtc_Word32));
+            tmp = (WebRtc_Word16*)malloc((sizeof(WebRtc_Word16) * lengthIn * 2));
+
+            // 11 -> 22 kHz in samplesOut
+            WebRtcSpl_UpsampleBy2(samplesIn, lengthIn, samplesOut, (WebRtc_Word32*)state1_);
+
+            // 22 -> 16 in tmp
+            for (int i = 0; i < (lengthIn * 2); i += 220)
+            {
+                WebRtcSpl_Resample22khzTo16khz(samplesOut + i, tmp + (i / 220) * 160,
+                                               (WebRtcSpl_State22khzTo16khz *)state2_,
+                                               tmp_mem);
+            }
+
+            // 16 -> 32 in samplesOut
+            WebRtcSpl_UpsampleBy2(tmp, (lengthIn * 16) / 11, samplesOut,
+                                  (WebRtc_Word32*)state3_);
+
+            outLen = (lengthIn * 32) / 11;
+
+            free(tmp_mem);
+            free(tmp);
+            return 0;
+
+        case kResamplerMode2To1:
+            if (maxLen < (lengthIn / 2))
+            {
+                return -1;
+            }
+            WebRtcSpl_DownsampleBy2(samplesIn, lengthIn, samplesOut, (WebRtc_Word32*)state1_);
+            outLen = lengthIn / 2;
+            return 0;
+        case kResamplerMode3To1:
+            // We can only handle blocks of 480 samples
+            // Can be fixed, but I don't think it's needed
+            if ((lengthIn % 480) != 0)
+            {
+                return -1;
+            }
+            if (maxLen < (lengthIn / 3))
+            {
+                return -1;
+            }
+            tmp_mem = (WebRtc_Word32*)malloc(496 * sizeof(WebRtc_Word32));
+
+            for (int i = 0; i < lengthIn; i += 480)
+            {
+                WebRtcSpl_Resample48khzTo16khz(samplesIn + i, samplesOut + i / 3,
+                                               (WebRtcSpl_State48khzTo16khz *)state1_,
+                                               tmp_mem);
+            }
+            outLen = lengthIn / 3;
+            free(tmp_mem);
+            return 0;
+        case kResamplerMode4To1:
+            if (maxLen < (lengthIn / 4))
+            {
+                return -1;
+            }
+            tmp = (WebRtc_Word16*)malloc(sizeof(WebRtc_Word16) * lengthIn / 2);
+            // 4:2
+            WebRtcSpl_DownsampleBy2(samplesIn, lengthIn, tmp, (WebRtc_Word32*)state1_);
+            // 2:1
+            WebRtcSpl_DownsampleBy2(tmp, lengthIn / 2, samplesOut, (WebRtc_Word32*)state2_);
+            outLen = lengthIn / 4;
+            free(tmp);
+            return 0;
+
+        case kResamplerMode6To1:
+            // We can only handle blocks of 480 samples
+            // Can be fixed, but I don't think it's needed
+            if ((lengthIn % 480) != 0)
+            {
+                return -1;
+            }
+            if (maxLen < (lengthIn / 6))
+            {
+                return -1;
+            }
+
+            tmp_mem = (WebRtc_Word32*)malloc(496 * sizeof(WebRtc_Word32));
+            tmp = (WebRtc_Word16*)malloc((sizeof(WebRtc_Word16) * lengthIn) / 3);
+
+            for (int i = 0; i < lengthIn; i += 480)
+            {
+                WebRtcSpl_Resample48khzTo16khz(samplesIn + i, tmp + i / 3,
+                                               (WebRtcSpl_State48khzTo16khz *)state1_,
+                                               tmp_mem);
+            }
+            outLen = lengthIn / 3;
+            free(tmp_mem);
+            WebRtcSpl_DownsampleBy2(tmp, outLen, samplesOut, (WebRtc_Word32*)state2_);
+            free(tmp);
+            outLen = outLen / 2;
+            return 0;
+        case kResamplerMode12To1:
+            // We can only handle blocks of 480 samples
+            // Can be fixed, but I don't think it's needed
+            if ((lengthIn % 480) != 0) {
+              return -1;
+            }
+            if (maxLen < (lengthIn / 12)) {
+              return -1;
+            }
+
+            tmp_mem = (WebRtc_Word32*) malloc(496 * sizeof(WebRtc_Word32));
+            tmp = (WebRtc_Word16*) malloc((sizeof(WebRtc_Word16) * lengthIn) / 3);
+            tmp_2 = (WebRtc_Word16*) malloc((sizeof(WebRtc_Word16) * lengthIn) / 6);
+            // 12:4
+            for (int i = 0; i < lengthIn; i += 480) {
+              // WebRtcSpl_Resample48khzTo16khz() takes a block of 480 samples
+              // as input and outputs a resampled block of 160 samples. The
+              // data is now actually in 96 kHz sampling rate, despite the
+              // function name, and with a resampling factor of 1/3 becomes
+              // 32 kHz.
+              WebRtcSpl_Resample48khzTo16khz(samplesIn + i, tmp + i / 3,
+                                             (WebRtcSpl_State48khzTo16khz*) state1_,
+                                             tmp_mem);
+            }
+            outLen = lengthIn / 3;
+            free(tmp_mem);
+            // 4:2
+            WebRtcSpl_DownsampleBy2(tmp, outLen, tmp_2,
+                                    (WebRtc_Word32*) state2_);
+            outLen = outLen / 2;
+            free(tmp);
+            // 2:1
+            WebRtcSpl_DownsampleBy2(tmp_2, outLen, samplesOut,
+                                    (WebRtc_Word32*) state3_);
+            free(tmp_2);
+            outLen = outLen / 2;
+            return 0;
+        case kResamplerMode3To2:
+            if (maxLen < (lengthIn * 2 / 3))
+            {
+                return -1;
+            }
+            // 3:6
+            tmp = static_cast<WebRtc_Word16*> (malloc(sizeof(WebRtc_Word16) * lengthIn * 2));
+            WebRtcSpl_UpsampleBy2(samplesIn, lengthIn, tmp, (WebRtc_Word32*)state1_);
+            lengthIn *= 2;
+            // 6:2
+            // We can only handle blocks of 480 samples
+            // Can be fixed, but I don't think it's needed
+            if ((lengthIn % 480) != 0)
+            {
+                free(tmp);
+                return -1;
+            }
+            tmp_mem = (WebRtc_Word32*)malloc(496 * sizeof(WebRtc_Word32));
+            for (int i = 0; i < lengthIn; i += 480)
+            {
+                WebRtcSpl_Resample48khzTo16khz(tmp + i, samplesOut + i / 3,
+                                               (WebRtcSpl_State48khzTo16khz *)state2_,
+                                               tmp_mem);
+            }
+            outLen = lengthIn / 3;
+            free(tmp);
+            free(tmp_mem);
+            return 0;
+        case kResamplerMode11To2:
+            // We can only handle blocks of 220 samples
+            // Can be fixed, but I don't think it's needed
+            if ((lengthIn % 220) != 0)
+            {
+                return -1;
+            }
+            if (maxLen < ((lengthIn * 2) / 11))
+            {
+                return -1;
+            }
+            tmp_mem = (WebRtc_Word32*)malloc(126 * sizeof(WebRtc_Word32));
+            tmp = (WebRtc_Word16*)malloc((lengthIn * 4) / 11 * sizeof(WebRtc_Word16));
+
+            for (int i = 0; i < lengthIn; i += 220)
+            {
+                WebRtcSpl_Resample22khzTo8khz(samplesIn + i, tmp + (i * 4) / 11,
+                                              (WebRtcSpl_State22khzTo8khz *)state1_,
+                                              tmp_mem);
+            }
+            lengthIn = (lengthIn * 4) / 11;
+
+            WebRtcSpl_DownsampleBy2(tmp, lengthIn, samplesOut, (WebRtc_Word32*)state2_);
+            outLen = lengthIn / 2;
+
+            free(tmp_mem);
+            free(tmp);
+            return 0;
+        case kResamplerMode11To4:
+            // We can only handle blocks of 220 samples
+            // Can be fixed, but I don't think it's needed
+            if ((lengthIn % 220) != 0)
+            {
+                return -1;
+            }
+            if (maxLen < ((lengthIn * 4) / 11))
+            {
+                return -1;
+            }
+            tmp_mem = (WebRtc_Word32*)malloc(126 * sizeof(WebRtc_Word32));
+
+            for (int i = 0; i < lengthIn; i += 220)
+            {
+                WebRtcSpl_Resample22khzTo8khz(samplesIn + i, samplesOut + (i * 4) / 11,
+                                              (WebRtcSpl_State22khzTo8khz *)state1_,
+                                              tmp_mem);
+            }
+            outLen = (lengthIn * 4) / 11;
+            free(tmp_mem);
+            return 0;
+        case kResamplerMode11To8:
+            // We can only handle blocks of 160 samples
+            // Can be fixed, but I don't think it's needed
+            if ((lengthIn % 220) != 0)
+            {
+                return -1;
+            }
+            if (maxLen < ((lengthIn * 8) / 11))
+            {
+                return -1;
+            }
+            tmp_mem = (WebRtc_Word32*)malloc(104 * sizeof(WebRtc_Word32));
+
+            for (int i = 0; i < lengthIn; i += 220)
+            {
+                WebRtcSpl_Resample22khzTo16khz(samplesIn + i, samplesOut + (i * 8) / 11,
+                                               (WebRtcSpl_State22khzTo16khz *)state1_,
+                                               tmp_mem);
+            }
+            outLen = (lengthIn * 8) / 11;
+            free(tmp_mem);
+            return 0;
+            break;
+
+    }
+    return 0;
+}
+
+// Asynchronous resampling, input
+int Resampler::Insert(WebRtc_Word16 * samplesIn, int lengthIn)
+{
+    if (my_type_ != kResamplerAsynchronous)
+    {
+        return -1;
+    }
+    int sizeNeeded, tenMsblock;
+
+    // Determine need for size of outBuffer
+    sizeNeeded = out_buffer_size_ + ((lengthIn + in_buffer_size_) * my_out_frequency_khz_)
+            / my_in_frequency_khz_;
+    if (sizeNeeded > out_buffer_size_max_)
+    {
+        // Round the value upwards to complete 10 ms blocks
+        tenMsblock = my_out_frequency_khz_ * 10;
+        sizeNeeded = (sizeNeeded / tenMsblock + 1) * tenMsblock;
+        out_buffer_ = (WebRtc_Word16*)realloc(out_buffer_, sizeNeeded * sizeof(WebRtc_Word16));
+        out_buffer_size_max_ = sizeNeeded;
+    }
+
+    // If we need to use inBuffer, make sure all input data fits there.
+
+    tenMsblock = my_in_frequency_khz_ * 10;
+    if (in_buffer_size_ || (lengthIn % tenMsblock))
+    {
+        // Check if input buffer size is enough
+        if ((in_buffer_size_ + lengthIn) > in_buffer_size_max_)
+        {
+            // Round the value upwards to complete 10 ms blocks
+            sizeNeeded = ((in_buffer_size_ + lengthIn) / tenMsblock + 1) * tenMsblock;
+            in_buffer_ = (WebRtc_Word16*)realloc(in_buffer_,
+                                                 sizeNeeded * sizeof(WebRtc_Word16));
+            in_buffer_size_max_ = sizeNeeded;
+        }
+        // Copy in data to input buffer
+        memcpy(in_buffer_ + in_buffer_size_, samplesIn, lengthIn * sizeof(WebRtc_Word16));
+
+        // Resample all available 10 ms blocks
+        int lenOut;
+        int dataLenToResample = (in_buffer_size_ / tenMsblock) * tenMsblock;
+        Push(in_buffer_, dataLenToResample, out_buffer_ + out_buffer_size_,
+             out_buffer_size_max_ - out_buffer_size_, lenOut);
+        out_buffer_size_ += lenOut;
+
+        // Save the rest
+        memmove(in_buffer_, in_buffer_ + dataLenToResample,
+                (in_buffer_size_ - dataLenToResample) * sizeof(WebRtc_Word16));
+        in_buffer_size_ -= dataLenToResample;
+    } else
+    {
+        // Just resample
+        int lenOut;
+        Push(in_buffer_, lengthIn, out_buffer_ + out_buffer_size_,
+             out_buffer_size_max_ - out_buffer_size_, lenOut);
+        out_buffer_size_ += lenOut;
+    }
+
+    return 0;
+}
+
+// Asynchronous resampling output, remaining samples are buffered
+int Resampler::Pull(WebRtc_Word16* samplesOut, int desiredLen, int &outLen)
+{
+    if (my_type_ != kResamplerAsynchronous)
+    {
+        return -1;
+    }
+
+    // Check that we have enough data
+    if (desiredLen <= out_buffer_size_)
+    {
+        // Give out the date
+        memcpy(samplesOut, out_buffer_, desiredLen * sizeof(WebRtc_Word32));
+
+        // Shuffle down remaining
+        memmove(out_buffer_, out_buffer_ + desiredLen,
+                (out_buffer_size_ - desiredLen) * sizeof(WebRtc_Word16));
+
+        // Update remaining size
+        out_buffer_size_ -= desiredLen;
+
+        return 0;
+    } else
+    {
+        return -1;
+    }
+}
+
+} // namespace webrtc
diff --git a/trunk/src/common_audio/resampler/resampler.gypi b/trunk/src/common_audio/resampler/resampler.gypi
new file mode 100644
index 0000000..69f9b0e
--- /dev/null
+++ b/trunk/src/common_audio/resampler/resampler.gypi
@@ -0,0 +1,55 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+  'targets': [
+    {
+      'target_name': 'resampler',
+      'type': '<(library)',
+      'dependencies': [
+        'signal_processing',
+      ],
+      'include_dirs': [
+        'include',
+      ],
+      'direct_dependent_settings': {
+        'include_dirs': [
+          'include',
+        ],
+      },
+      'sources': [
+        'include/resampler.h',
+        'resampler.cc',
+      ],
+    },
+  ], # targets
+  'conditions': [
+    ['build_with_chromium==0', {
+      'targets' : [
+        {
+          'target_name': 'resampler_unittests',
+          'type': 'executable',
+          'dependencies': [
+            'resampler',
+            '<(webrtc_root)/../test/test.gyp:test_support_main',
+            '<(webrtc_root)/../testing/gtest.gyp:gtest',
+          ],
+          'sources': [
+            'resampler_unittest.cc',
+          ],
+        }, # resampler_unittests
+      ], # targets
+    }], # build_with_chromium
+  ], # conditions
+}
+
+# Local Variables:
+# tab-width:2
+# indent-tabs-mode:nil
+# End:
+# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/common_audio/resampler/resampler_unittest.cc b/trunk/src/common_audio/resampler/resampler_unittest.cc
new file mode 100644
index 0000000..9b1061a
--- /dev/null
+++ b/trunk/src/common_audio/resampler/resampler_unittest.cc
@@ -0,0 +1,143 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "gtest/gtest.h"
+
+#include "common_audio/resampler/include/resampler.h"
+
+// TODO(andrew): this is a work-in-progress. Many more tests are needed.
+
+namespace webrtc {
+namespace {
+const ResamplerType kTypes[] = {
+  kResamplerSynchronous,
+  kResamplerAsynchronous,
+  kResamplerSynchronousStereo,
+  kResamplerAsynchronousStereo
+  // kResamplerInvalid excluded
+};
+const size_t kTypesSize = sizeof(kTypes) / sizeof(*kTypes);
+
+// Rates we must support.
+const int kMaxRate = 96000;
+const int kRates[] = {
+  8000,
+  16000,
+  32000,
+  44000,
+  48000,
+  kMaxRate
+};
+const size_t kRatesSize = sizeof(kRates) / sizeof(*kRates);
+const int kMaxChannels = 2;
+const size_t kDataSize = static_cast<size_t> (kMaxChannels * kMaxRate / 100);
+
+// TODO(andrew): should we be supporting these combinations?
+bool ValidRates(int in_rate, int out_rate) {
+  // Not the most compact notation, for clarity.
+  if ((in_rate == 44000 && (out_rate == 48000 || out_rate == 96000)) ||
+      (out_rate == 44000 && (in_rate == 48000 || in_rate == 96000))) {
+    return false;
+  }
+
+  return true;
+}
+
+class ResamplerTest : public testing::Test {
+ protected:
+  ResamplerTest();
+  virtual void SetUp();
+  virtual void TearDown();
+
+  Resampler rs_;
+  int16_t data_in_[kDataSize];
+  int16_t data_out_[kDataSize];
+};
+
+ResamplerTest::ResamplerTest() {}
+
+void ResamplerTest::SetUp() {
+  // Initialize input data with anything. The tests are content independent.
+  memset(data_in_, 1, sizeof(data_in_));
+}
+
+void ResamplerTest::TearDown() {}
+
+TEST_F(ResamplerTest, Reset) {
+  // The only failure mode for the constructor is if Reset() fails. For the
+  // time being then (until an Init function is added), we rely on Reset()
+  // to test the constructor.
+
+  // Check that all required combinations are supported.
+  for (size_t i = 0; i < kRatesSize; ++i) {
+    for (size_t j = 0; j < kRatesSize; ++j) {
+      for (size_t k = 0; k < kTypesSize; ++k) {
+        std::ostringstream ss;
+        ss << "Input rate: " << kRates[i] << ", output rate: " << kRates[j]
+            << ", type: " << kTypes[k];
+        SCOPED_TRACE(ss.str());
+        if (ValidRates(kRates[i], kRates[j]))
+          EXPECT_EQ(0, rs_.Reset(kRates[i], kRates[j], kTypes[k]));
+        else
+          EXPECT_EQ(-1, rs_.Reset(kRates[i], kRates[j], kTypes[k]));
+      }
+    }
+  }
+}
+
+// TODO(tlegrand): Replace code inside the two tests below with a function
+// with number of channels and ResamplerType as input.
+TEST_F(ResamplerTest, Synchronous) {
+  for (size_t i = 0; i < kRatesSize; ++i) {
+    for (size_t j = 0; j < kRatesSize; ++j) {
+      std::ostringstream ss;
+      ss << "Input rate: " << kRates[i] << ", output rate: " << kRates[j];
+      SCOPED_TRACE(ss.str());
+
+      if (ValidRates(kRates[i], kRates[j])) {
+        int in_length = kRates[i] / 100;
+        int out_length = 0;
+        EXPECT_EQ(0, rs_.Reset(kRates[i], kRates[j], kResamplerSynchronous));
+        EXPECT_EQ(0, rs_.Push(data_in_, in_length, data_out_, kDataSize,
+                              out_length));
+        EXPECT_EQ(kRates[j] / 100, out_length);
+      } else {
+        EXPECT_EQ(-1, rs_.Reset(kRates[i], kRates[j], kResamplerSynchronous));
+      }
+    }
+  }
+}
+
+TEST_F(ResamplerTest, SynchronousStereo) {
+  // Number of channels is 2, stereo mode.
+  const int kChannels = 2;
+  for (size_t i = 0; i < kRatesSize; ++i) {
+    for (size_t j = 0; j < kRatesSize; ++j) {
+      std::ostringstream ss;
+      ss << "Input rate: " << kRates[i] << ", output rate: " << kRates[j];
+      SCOPED_TRACE(ss.str());
+
+      if (ValidRates(kRates[i], kRates[j])) {
+        int in_length = kChannels * kRates[i] / 100;
+        int out_length = 0;
+        EXPECT_EQ(0, rs_.Reset(kRates[i], kRates[j],
+                               kResamplerSynchronousStereo));
+        EXPECT_EQ(0, rs_.Push(data_in_, in_length, data_out_, kDataSize,
+                              out_length));
+        EXPECT_EQ(kChannels * kRates[j] / 100, out_length);
+      } else {
+        EXPECT_EQ(-1, rs_.Reset(kRates[i], kRates[j],
+                                kResamplerSynchronousStereo));
+      }
+    }
+  }
+}
+}  // namespace
+}  // namespace webrtc
diff --git a/trunk/src/common_audio/signal_processing/Android.mk b/trunk/src/common_audio/signal_processing/Android.mk
new file mode 100644
index 0000000..3ff066c
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/Android.mk
@@ -0,0 +1,100 @@
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+include $(LOCAL_PATH)/../../../android-webrtc.mk
+
+LOCAL_ARM_MODE := arm
+LOCAL_MODULE_CLASS := STATIC_LIBRARIES
+LOCAL_MODULE := libwebrtc_spl
+LOCAL_MODULE_TAGS := optional
+LOCAL_SRC_FILES := \
+    auto_corr_to_refl_coef.c \
+    auto_correlation.c \
+    complex_fft.c \
+    copy_set_operations.c \
+    division_operations.c \
+    dot_product_with_scale.c \
+    energy.c \
+    filter_ar.c \
+    filter_ma_fast_q12.c \
+    get_hanning_window.c \
+    get_scaling_square.c \
+    ilbc_specific_functions.c \
+    levinson_durbin.c \
+    lpc_to_refl_coef.c \
+    min_max_operations.c \
+    randomization_functions.c \
+    refl_coef_to_lpc.c \
+    resample.c \
+    resample_48khz.c \
+    resample_by_2.c \
+    resample_by_2_internal.c \
+    resample_fractional.c \
+    spl_sqrt.c \
+    spl_version.c \
+    splitting_filter.c \
+    sqrt_of_one_minus_x_squared.c \
+    vector_scaling_operations.c
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+    $(MY_WEBRTC_COMMON_DEFS)
+
+LOCAL_C_INCLUDES := \
+    $(LOCAL_PATH)/include \
+    $(LOCAL_PATH)/../.. 
+
+ifeq ($(ARCH_ARM_HAVE_NEON),true)
+LOCAL_SRC_FILES += \
+    min_max_operations_neon.c \
+    cross_correlation_neon.s \
+    downsample_fast_neon.s
+LOCAL_CFLAGS += \
+    $(MY_ARM_CFLAGS_NEON)
+else
+LOCAL_SRC_FILES += \
+    cross_correlation.c \
+    downsample_fast.c
+endif
+
+ifeq ($(ARCH_ARM_HAVE_ARMV7A),true)
+LOCAL_SRC_FILES += \
+    filter_ar_fast_q12_armv7.s
+else
+LOCAL_SRC_FILES += \
+    filter_ar_fast_q12.c
+endif
+
+ifeq ($(TARGET_ARCH),arm)
+LOCAL_SRC_FILES += \
+    complex_bit_reverse_arm.s \
+    spl_sqrt_floor.s
+else
+LOCAL_SRC_FILES += \
+    complex_bit_reverse.c \
+    spl_sqrt_floor.c
+endif
+
+LOCAL_SHARED_LIBRARIES := libstlport
+
+ifeq ($(TARGET_OS)-$(TARGET_SIMULATOR),linux-true)
+LOCAL_LDLIBS += -ldl -lpthread
+endif
+
+ifneq ($(TARGET_SIMULATOR),true)
+LOCAL_SHARED_LIBRARIES += libdl
+endif
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/common_audio/signal_processing/auto_corr_to_refl_coef.c b/trunk/src/common_audio/signal_processing/auto_corr_to_refl_coef.c
new file mode 100644
index 0000000..b7e8858
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/auto_corr_to_refl_coef.c
@@ -0,0 +1,103 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains the function WebRtcSpl_AutoCorrToReflCoef().
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "signal_processing_library.h"
+
+void WebRtcSpl_AutoCorrToReflCoef(G_CONST WebRtc_Word32 *R, int use_order, WebRtc_Word16 *K)
+{
+    int i, n;
+    WebRtc_Word16 tmp;
+    G_CONST WebRtc_Word32 *rptr;
+    WebRtc_Word32 L_num, L_den;
+    WebRtc_Word16 *acfptr, *pptr, *wptr, *p1ptr, *w1ptr, ACF[WEBRTC_SPL_MAX_LPC_ORDER],
+            P[WEBRTC_SPL_MAX_LPC_ORDER], W[WEBRTC_SPL_MAX_LPC_ORDER];
+
+    // Initialize loop and pointers.
+    acfptr = ACF;
+    rptr = R;
+    pptr = P;
+    p1ptr = &P[1];
+    w1ptr = &W[1];
+    wptr = w1ptr;
+
+    // First loop; n=0. Determine shifting.
+    tmp = WebRtcSpl_NormW32(*R);
+    *acfptr = (WebRtc_Word16)((*rptr++ << tmp) >> 16);
+    *pptr++ = *acfptr++;
+
+    // Initialize ACF, P and W.
+    for (i = 1; i <= use_order; i++)
+    {
+        *acfptr = (WebRtc_Word16)((*rptr++ << tmp) >> 16);
+        *wptr++ = *acfptr;
+        *pptr++ = *acfptr++;
+    }
+
+    // Compute reflection coefficients.
+    for (n = 1; n <= use_order; n++, K++)
+    {
+        tmp = WEBRTC_SPL_ABS_W16(*p1ptr);
+        if (*P < tmp)
+        {
+            for (i = n; i <= use_order; i++)
+                *K++ = 0;
+
+            return;
+        }
+
+        // Division: WebRtcSpl_div(tmp, *P)
+        *K = 0;
+        if (tmp != 0)
+        {
+            L_num = tmp;
+            L_den = *P;
+            i = 15;
+            while (i--)
+            {
+                (*K) <<= 1;
+                L_num <<= 1;
+                if (L_num >= L_den)
+                {
+                    L_num -= L_den;
+                    (*K)++;
+                }
+            }
+            if (*p1ptr > 0)
+                *K = -*K;
+        }
+
+        // Last iteration; don't do Schur recursion.
+        if (n == use_order)
+            return;
+
+        // Schur recursion.
+        pptr = P;
+        wptr = w1ptr;
+        tmp = (WebRtc_Word16)(((WebRtc_Word32)*p1ptr * (WebRtc_Word32)*K + 16384) >> 15);
+        *pptr = WEBRTC_SPL_ADD_SAT_W16( *pptr, tmp );
+        pptr++;
+        for (i = 1; i <= use_order - n; i++)
+        {
+            tmp = (WebRtc_Word16)(((WebRtc_Word32)*wptr * (WebRtc_Word32)*K + 16384) >> 15);
+            *pptr = WEBRTC_SPL_ADD_SAT_W16( *(pptr+1), tmp );
+            pptr++;
+            tmp = (WebRtc_Word16)(((WebRtc_Word32)*pptr * (WebRtc_Word32)*K + 16384) >> 15);
+            *wptr = WEBRTC_SPL_ADD_SAT_W16( *wptr, tmp );
+            wptr++;
+        }
+    }
+}
diff --git a/trunk/src/common_audio/signal_processing/auto_correlation.c b/trunk/src/common_audio/signal_processing/auto_correlation.c
new file mode 100644
index 0000000..a00fde4
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/auto_correlation.c
@@ -0,0 +1,141 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains the function WebRtcSpl_AutoCorrelation().
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "signal_processing_library.h"
+
+int WebRtcSpl_AutoCorrelation(G_CONST WebRtc_Word16* in_vector,
+                              int in_vector_length,
+                              int order,
+                              WebRtc_Word32* result,
+                              int* scale)
+{
+    WebRtc_Word32 sum;
+    int i, j;
+    WebRtc_Word16 smax; // Sample max
+    G_CONST WebRtc_Word16* xptr1;
+    G_CONST WebRtc_Word16* xptr2;
+    WebRtc_Word32* resultptr;
+    int scaling = 0;
+
+#ifdef _ARM_OPT_
+#pragma message("NOTE: _ARM_OPT_ optimizations are used")
+    WebRtc_Word16 loops4;
+#endif
+
+    if (order < 0)
+        order = in_vector_length;
+
+    // Find the max. sample
+    smax = WebRtcSpl_MaxAbsValueW16(in_vector, in_vector_length);
+
+    // In order to avoid overflow when computing the sum we should scale the samples so that
+    // (in_vector_length * smax * smax) will not overflow.
+
+    if (smax == 0)
+    {
+        scaling = 0;
+    } else
+    {
+        int nbits = WebRtcSpl_GetSizeInBits(in_vector_length); // # of bits in the sum loop
+        int t = WebRtcSpl_NormW32(WEBRTC_SPL_MUL(smax, smax)); // # of bits to normalize smax
+
+        if (t > nbits)
+        {
+            scaling = 0;
+        } else
+        {
+            scaling = nbits - t;
+        }
+
+    }
+
+    resultptr = result;
+
+    // Perform the actual correlation calculation
+    for (i = 0; i < order + 1; i++)
+    {
+        int loops = (in_vector_length - i);
+        sum = 0;
+        xptr1 = in_vector;
+        xptr2 = &in_vector[i];
+#ifndef _ARM_OPT_
+        for (j = loops; j > 0; j--)
+        {
+            sum += WEBRTC_SPL_MUL_16_16_RSFT(*xptr1++, *xptr2++, scaling);
+        }
+#else
+        loops4 = (loops >> 2) << 2;
+
+        if (scaling == 0)
+        {
+            for (j = 0; j < loops4; j = j + 4)
+            {
+                sum += WEBRTC_SPL_MUL_16_16(*xptr1, *xptr2);
+                xptr1++;
+                xptr2++;
+                sum += WEBRTC_SPL_MUL_16_16(*xptr1, *xptr2);
+                xptr1++;
+                xptr2++;
+                sum += WEBRTC_SPL_MUL_16_16(*xptr1, *xptr2);
+                xptr1++;
+                xptr2++;
+                sum += WEBRTC_SPL_MUL_16_16(*xptr1, *xptr2);
+                xptr1++;
+                xptr2++;
+            }
+
+            for (j = loops4; j < loops; j++)
+            {
+                sum += WEBRTC_SPL_MUL_16_16(*xptr1, *xptr2);
+                xptr1++;
+                xptr2++;
+            }
+        }
+        else
+        {
+            for (j = 0; j < loops4; j = j + 4)
+            {
+                sum += WEBRTC_SPL_MUL_16_16_RSFT(*xptr1, *xptr2, scaling);
+                xptr1++;
+                xptr2++;
+                sum += WEBRTC_SPL_MUL_16_16_RSFT(*xptr1, *xptr2, scaling);
+                xptr1++;
+                xptr2++;
+                sum += WEBRTC_SPL_MUL_16_16_RSFT(*xptr1, *xptr2, scaling);
+                xptr1++;
+                xptr2++;
+                sum += WEBRTC_SPL_MUL_16_16_RSFT(*xptr1, *xptr2, scaling);
+                xptr1++;
+                xptr2++;
+            }
+
+            for (j = loops4; j < loops; j++)
+            {
+                sum += WEBRTC_SPL_MUL_16_16_RSFT(*xptr1, *xptr2, scaling);
+                xptr1++;
+                xptr2++;
+            }
+        }
+
+#endif
+        *resultptr++ = sum;
+    }
+
+    *scale = scaling;
+
+    return order + 1;
+}
diff --git a/trunk/src/common_audio/signal_processing/complex_bit_reverse.c b/trunk/src/common_audio/signal_processing/complex_bit_reverse.c
new file mode 100644
index 0000000..02fde1e
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/complex_bit_reverse.c
@@ -0,0 +1,109 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "signal_processing_library.h"
+
+/* Tables for data buffer indexes that are bit reversed and thus need to be
+ * swapped. Note that, index_7[{0, 2, 4, ...}] are for the left side of the swap
+ * operations, while index_7[{1, 3, 5, ...}] are for the right side of the
+ * operation. Same for index_8.
+ */
+
+/* Indexes for the case of stages == 7. */
+static const int16_t index_7[112] = {
+  1, 64, 2, 32, 3, 96, 4, 16, 5, 80, 6, 48, 7, 112, 9, 72, 10, 40, 11, 104,
+  12, 24, 13, 88, 14, 56, 15, 120, 17, 68, 18, 36, 19, 100, 21, 84, 22, 52,
+  23, 116, 25, 76, 26, 44, 27, 108, 29, 92, 30, 60, 31, 124, 33, 66, 35, 98,
+  37, 82, 38, 50, 39, 114, 41, 74, 43, 106, 45, 90, 46, 58, 47, 122, 49, 70,
+  51, 102, 53, 86, 55, 118, 57, 78, 59, 110, 61, 94, 63, 126, 67, 97, 69,
+  81, 71, 113, 75, 105, 77, 89, 79, 121, 83, 101, 87, 117, 91, 109, 95, 125,
+  103, 115, 111, 123
+};
+
+/* Indexes for the case of stages == 8. */
+static const int16_t index_8[240] = {
+  1, 128, 2, 64, 3, 192, 4, 32, 5, 160, 6, 96, 7, 224, 8, 16, 9, 144, 10, 80,
+  11, 208, 12, 48, 13, 176, 14, 112, 15, 240, 17, 136, 18, 72, 19, 200, 20,
+  40, 21, 168, 22, 104, 23, 232, 25, 152, 26, 88, 27, 216, 28, 56, 29, 184,
+  30, 120, 31, 248, 33, 132, 34, 68, 35, 196, 37, 164, 38, 100, 39, 228, 41,
+  148, 42, 84, 43, 212, 44, 52, 45, 180, 46, 116, 47, 244, 49, 140, 50, 76,
+  51, 204, 53, 172, 54, 108, 55, 236, 57, 156, 58, 92, 59, 220, 61, 188, 62,
+  124, 63, 252, 65, 130, 67, 194, 69, 162, 70, 98, 71, 226, 73, 146, 74, 82,
+  75, 210, 77, 178, 78, 114, 79, 242, 81, 138, 83, 202, 85, 170, 86, 106, 87,
+  234, 89, 154, 91, 218, 93, 186, 94, 122, 95, 250, 97, 134, 99, 198, 101,
+  166, 103, 230, 105, 150, 107, 214, 109, 182, 110, 118, 111, 246, 113, 142,
+  115, 206, 117, 174, 119, 238, 121, 158, 123, 222, 125, 190, 127, 254, 131,
+  193, 133, 161, 135, 225, 137, 145, 139, 209, 141, 177, 143, 241, 147, 201,
+  149, 169, 151, 233, 155, 217, 157, 185, 159, 249, 163, 197, 167, 229, 171,
+  213, 173, 181, 175, 245, 179, 205, 183, 237, 187, 221, 191, 253, 199, 227,
+  203, 211, 207, 243, 215, 235, 223, 251, 239, 247
+};
+
+void WebRtcSpl_ComplexBitReverse(int16_t* __restrict complex_data, int stages) {
+  /* For any specific value of stages, we know exactly the indexes that are
+   * bit reversed. Currently (Feb. 2012) in WebRTC the only possible values of
+   * stages are 7 and 8, so we use tables to save unnecessary iterations and
+   * calculations for these two cases.
+   */
+  if (stages == 7 || stages == 8) {
+    int m = 0;
+    int length = 112;
+    const int16_t* index = index_7;
+
+    if (stages == 8) {
+      length = 240;
+      index = index_8;
+    }
+
+    /* Decimation in time. Swap the elements with bit-reversed indexes. */
+    for (m = 0; m < length; m += 2) {
+      /* We declare a int32_t* type pointer, to load both the 16-bit real
+       * and imaginary elements from complex_data in one instruction, reducing
+       * complexity.
+       */
+      int32_t* complex_data_ptr = (int32_t*)complex_data;
+      int32_t temp = 0;
+
+      temp = complex_data_ptr[index[m]];  /* Real and imaginary */
+      complex_data_ptr[index[m]] = complex_data_ptr[index[m + 1]];
+      complex_data_ptr[index[m + 1]] = temp;
+    }
+  }
+  else {
+    int m = 0, mr = 0, l = 0;
+    int n = 1 << stages;
+    int nn = n - 1;
+
+    /* Decimation in time - re-order data */
+    for (m = 1; m <= nn; ++m) {
+      int32_t* complex_data_ptr = (int32_t*)complex_data;
+      int32_t temp = 0;
+
+      /* Find out indexes that are bit-reversed. */
+      l = n;
+      do {
+        l >>= 1;
+      } while (l > nn - mr);
+      mr = (mr & (l - 1)) + l;
+
+      if (mr <= m) {
+        continue;
+      }
+
+      /* Swap the elements with bit-reversed indexes.
+       * This is similar to the loop in the stages == 7 or 8 cases.
+       */
+      temp = complex_data_ptr[m];  /* Real and imaginary */
+      complex_data_ptr[m] = complex_data_ptr[mr];
+      complex_data_ptr[mr] = temp;
+    }
+  }
+}
+
diff --git a/trunk/src/common_audio/signal_processing/complex_bit_reverse_arm.s b/trunk/src/common_audio/signal_processing/complex_bit_reverse_arm.s
new file mode 100644
index 0000000..4828077
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/complex_bit_reverse_arm.s
@@ -0,0 +1,126 @@
+@
+@ Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+@
+@ Use of this source code is governed by a BSD-style license
+@ that can be found in the LICENSE file in the root of the source
+@ tree. An additional intellectual property rights grant can be found
+@ in the file PATENTS.  All contributing project authors may
+@ be found in the AUTHORS file in the root of the source tree.
+@
+
+@ This file contains the function WebRtcSpl_ComplexBitReverse(), optimized
+@ for ARMv5 platforms.
+@ Reference C code is in file complex_bit_reverse.c. Bit-exact.
+
+.arch armv5
+
+.global WebRtcSpl_ComplexBitReverse
+
+.align  2
+
+WebRtcSpl_ComplexBitReverse:
+.fnstart
+
+  push {r4-r7}
+
+  cmp r1, #7
+  adr r3, index_7                 @ Table pointer.
+  mov r4, #112                    @ Number of interations.
+  beq PRE_LOOP_STAGES_7_OR_8
+
+  cmp r1, #8
+  adr r3, index_8                 @ Table pointer.
+  mov r4, #240                    @ Number of interations.
+  beq PRE_LOOP_STAGES_7_OR_8
+
+  mov r3, #1                      @ Initialize m.
+  mov r1, r3, asl r1              @ n = 1 << stages;
+  subs r6, r1, #1                 @ nn = n - 1;
+  ble END
+
+  mov r5, r0                      @ &complex_data
+  mov r4, #0                      @ ml
+
+LOOP_GENERIC:
+  rsb r12, r4, r6                 @ l > nn - mr
+  mov r2, r1                      @ n
+
+LOOP_SHIFT:
+  asr r2, #1                      @ l >>= 1;
+  cmp r2, r12
+  bgt LOOP_SHIFT
+
+  sub r12, r2, #1
+  and r4, r12, r4
+  add r4, r2                      @ mr = (mr & (l - 1)) + l;
+  cmp r4, r3                      @ mr <= m ?
+  ble UPDATE_REGISTERS
+
+  mov r12, r4, asl #2
+  ldr r7, [r5, #4]                @ complex_data[2 * m, 2 * m + 1].
+                                  @   Offset 4 due to m incrementing from 1.
+  ldr r2, [r0, r12]               @ complex_data[2 * mr, 2 * mr + 1].
+  str r7, [r0, r12]
+  str r2, [r5, #4]
+
+UPDATE_REGISTERS:
+  add r3, r3, #1
+  add r5, #4
+  cmp r3, r1
+  bne LOOP_GENERIC
+
+  b END
+
+PRE_LOOP_STAGES_7_OR_8:
+  add r4, r3, r4, asl #1
+
+LOOP_STAGES_7_OR_8:
+  ldrsh r2, [r3], #2              @ index[m]
+  ldrsh r5, [r3], #2              @ index[m + 1]
+  ldr r1, [r0, r2]                @ complex_data[index[m], index[m] + 1]
+  ldr r12, [r0, r5]               @ complex_data[index[m + 1], index[m + 1] + 1]
+  cmp r3, r4
+  str r1, [r0, r5]
+  str r12, [r0, r2]
+  bne LOOP_STAGES_7_OR_8
+
+END:
+  pop {r4-r7}
+  bx lr
+
+.fnend
+
+
+@ The index tables. Note the values are doubles of the actual indexes for 16-bit
+@ elements, different from the generic C code. It actually provides byte offsets
+@ for the indexes.
+
+.align  2
+index_7:  @ Indexes for stages == 7.
+  .hword 4, 256, 8, 128, 12, 384, 16, 64, 20, 320, 24, 192, 28, 448, 36, 288
+  .hword 40, 160, 44, 416, 48, 96, 52, 352, 56, 224, 60, 480, 68, 272, 72, 144
+  .hword 76, 400, 84, 336, 88, 208, 92, 464, 100, 304, 104, 176, 108, 432, 116
+  .hword 368, 120, 240, 124, 496, 132, 264, 140, 392, 148, 328, 152, 200, 156
+  .hword 456, 164, 296, 172, 424, 180, 360, 184, 232, 188, 488, 196, 280, 204
+  .hword 408, 212, 344, 220, 472, 228, 312, 236, 440, 244, 376, 252, 504, 268
+  .hword 388, 276, 324, 284, 452, 300, 420, 308, 356, 316, 484, 332, 404, 348
+  .hword 468, 364, 436, 380, 500, 412, 460, 444, 492
+
+index_8:  @ Indexes for stages == 8.
+  .hword 4, 512, 8, 256, 12, 768, 16, 128, 20, 640, 24, 384, 28, 896, 32, 64
+  .hword 36, 576, 40, 320, 44, 832, 48, 192, 52, 704, 56, 448, 60, 960, 68, 544
+  .hword 72, 288, 76, 800, 80, 160, 84, 672, 88, 416, 92, 928, 100, 608, 104
+  .hword 352, 108, 864, 112, 224, 116, 736, 120, 480, 124, 992, 132, 528, 136
+  .hword 272, 140, 784, 148, 656, 152, 400, 156, 912, 164, 592, 168, 336, 172
+  .hword 848, 176, 208, 180, 720, 184, 464, 188, 976, 196, 560, 200, 304, 204
+  .hword 816, 212, 688, 216, 432, 220, 944, 228, 624, 232, 368, 236, 880, 244
+  .hword 752, 248, 496, 252, 1008, 260, 520, 268, 776, 276, 648, 280, 392, 284
+  .hword 904, 292, 584, 296, 328, 300, 840, 308, 712, 312, 456, 316, 968, 324
+  .hword 552, 332, 808, 340, 680, 344, 424, 348, 936, 356, 616, 364, 872, 372
+  .hword 744, 376, 488, 380, 1000, 388, 536, 396, 792, 404, 664, 412, 920, 420
+  .hword 600, 428, 856, 436, 728, 440, 472, 444, 984, 452, 568, 460, 824, 468
+  .hword 696, 476, 952, 484, 632, 492, 888, 500, 760, 508, 1016, 524, 772, 532
+  .hword 644, 540, 900, 548, 580, 556, 836, 564, 708, 572, 964, 588, 804, 596
+  .hword 676, 604, 932, 620, 868, 628, 740, 636, 996, 652, 788, 668, 916, 684
+  .hword 852, 692, 724, 700, 980, 716, 820, 732, 948, 748, 884, 764, 1012, 796
+  .hword 908, 812, 844, 828, 972, 860, 940, 892, 1004, 956, 988
diff --git a/trunk/src/common_audio/signal_processing/complex_fft.c b/trunk/src/common_audio/signal_processing/complex_fft.c
new file mode 100644
index 0000000..1e8503c
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/complex_fft.c
@@ -0,0 +1,425 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains the function WebRtcSpl_ComplexFFT().
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "signal_processing_library.h"
+
+#define CFFTSFT 14
+#define CFFTRND 1
+#define CFFTRND2 16384
+
+#define CIFFTSFT 14
+#define CIFFTRND 1
+
+static const WebRtc_Word16 kSinTable1024[] = {
+      0,    201,    402,    603,    804,   1005,   1206,   1406,
+   1607,   1808,   2009,   2209,   2410,   2610,   2811,   3011,
+   3211,   3411,   3611,   3811,   4011,   4210,   4409,   4608,
+   4807,   5006,   5205,   5403,   5601,   5799,   5997,   6195,
+   6392,   6589,   6786,   6982,   7179,   7375,   7571,   7766,
+   7961,   8156,   8351,   8545,   8739,   8932,   9126,   9319,
+   9511,   9703,   9895,  10087,  10278,  10469,  10659,  10849,
+  11038,  11227,  11416,  11604,  11792,  11980,  12166,  12353,
+  12539,  12724,  12909,  13094,  13278,  13462,  13645,  13827,
+  14009,  14191,  14372,  14552,  14732,  14911,  15090,  15268,
+  15446,  15623,  15799,  15975,  16150,  16325,  16499,  16672,
+  16845,  17017,  17189,  17360,  17530,  17699,  17868,  18036,
+  18204,  18371,  18537,  18702,  18867,  19031,  19194,  19357,
+  19519,  19680,  19840,  20000,  20159,  20317,  20474,  20631,
+  20787,  20942,  21096,  21249,  21402,  21554,  21705,  21855,
+  22004,  22153,  22301,  22448,  22594,  22739,  22883,  23027,
+  23169,  23311,  23452,  23592,  23731,  23869,  24006,  24143,
+  24278,  24413,  24546,  24679,  24811,  24942,  25072,  25201,
+  25329,  25456,  25582,  25707,  25831,  25954,  26077,  26198,
+  26318,  26437,  26556,  26673,  26789,  26905,  27019,  27132,
+  27244,  27355,  27466,  27575,  27683,  27790,  27896,  28001,
+  28105,  28208,  28309,  28410,  28510,  28608,  28706,  28802,
+  28897,  28992,  29085,  29177,  29268,  29358,  29446,  29534,
+  29621,  29706,  29790,  29873,  29955,  30036,  30116,  30195,
+  30272,  30349,  30424,  30498,  30571,  30643,  30713,  30783,
+  30851,  30918,  30984,  31049,
+  31113,  31175,  31236,  31297,
+  31356,  31413,  31470,  31525,  31580,  31633,  31684,  31735,
+  31785,  31833,  31880,  31926,  31970,  32014,  32056,  32097,
+  32137,  32176,  32213,  32249,  32284,  32318,  32350,  32382,
+  32412,  32441,  32468,  32495,  32520,  32544,  32567,  32588,
+  32609,  32628,  32646,  32662,  32678,  32692,  32705,  32717,
+  32727,  32736,  32744,  32751,  32757,  32761,  32764,  32766,
+  32767,  32766,  32764,  32761,  32757,  32751,  32744,  32736,
+  32727,  32717,  32705,  32692,  32678,  32662,  32646,  32628,
+  32609,  32588,  32567,  32544,  32520,  32495,  32468,  32441,
+  32412,  32382,  32350,  32318,  32284,  32249,  32213,  32176,
+  32137,  32097,  32056,  32014,  31970,  31926,  31880,  31833,
+  31785,  31735,  31684,  31633,  31580,  31525,  31470,  31413,
+  31356,  31297,  31236,  31175,  31113,  31049,  30984,  30918,
+  30851,  30783,  30713,  30643,  30571,  30498,  30424,  30349,
+  30272,  30195,  30116,  30036,  29955,  29873,  29790,  29706,
+  29621,  29534,  29446,  29358,  29268,  29177,  29085,  28992,
+  28897,  28802,  28706,  28608,  28510,  28410,  28309,  28208,
+  28105,  28001,  27896,  27790,  27683,  27575,  27466,  27355,
+  27244,  27132,  27019,  26905,  26789,  26673,  26556,  26437,
+  26318,  26198,  26077,  25954,  25831,  25707,  25582,  25456,
+  25329,  25201,  25072,  24942,  24811,  24679,  24546,  24413,
+  24278,  24143,  24006,  23869,  23731,  23592,  23452,  23311,
+  23169,  23027,  22883,  22739,  22594,  22448,  22301,  22153,
+  22004,  21855,  21705,  21554,  21402,  21249,  21096,  20942,
+  20787,  20631,  20474,  20317,  20159,  20000,  19840,  19680,
+  19519,  19357,  19194,  19031,  18867,  18702,  18537,  18371,
+  18204,  18036,  17868,  17699,  17530,  17360,  17189,  17017,
+  16845,  16672,  16499,  16325,  16150,  15975,  15799,  15623,
+  15446,  15268,  15090,  14911,  14732,  14552,  14372,  14191,
+  14009,  13827,  13645,  13462,  13278,  13094,  12909,  12724,
+  12539,  12353,  12166,  11980,  11792,  11604,  11416,  11227,
+  11038,  10849,  10659,  10469,  10278,  10087,   9895,   9703,
+   9511,   9319,   9126,   8932,   8739,   8545,   8351,   8156,
+   7961,   7766,   7571,   7375,   7179,   6982,   6786,   6589,
+   6392,   6195,   5997,   5799,   5601,   5403,   5205,   5006,
+   4807,   4608,   4409,   4210,   4011,   3811,   3611,   3411,
+   3211,   3011,   2811,   2610,   2410,   2209,   2009,   1808,
+   1607,   1406,   1206,   1005,    804,    603,    402,    201,
+      0,   -201,   -402,   -603,   -804,  -1005,  -1206,  -1406,
+  -1607,  -1808,  -2009,  -2209,  -2410,  -2610,  -2811,  -3011,
+  -3211,  -3411,  -3611,  -3811,  -4011,  -4210,  -4409,  -4608,
+  -4807,  -5006,  -5205,  -5403,  -5601,  -5799,  -5997,  -6195,
+  -6392,  -6589,  -6786,  -6982,  -7179,  -7375,  -7571,  -7766,
+  -7961,  -8156,  -8351,  -8545,  -8739,  -8932,  -9126,  -9319,
+  -9511,  -9703,  -9895, -10087, -10278, -10469, -10659, -10849,
+ -11038, -11227, -11416, -11604, -11792, -11980, -12166, -12353,
+ -12539, -12724, -12909, -13094, -13278, -13462, -13645, -13827,
+ -14009, -14191, -14372, -14552, -14732, -14911, -15090, -15268,
+ -15446, -15623, -15799, -15975, -16150, -16325, -16499, -16672,
+ -16845, -17017, -17189, -17360, -17530, -17699, -17868, -18036,
+ -18204, -18371, -18537, -18702, -18867, -19031, -19194, -19357,
+ -19519, -19680, -19840, -20000, -20159, -20317, -20474, -20631,
+ -20787, -20942, -21096, -21249, -21402, -21554, -21705, -21855,
+ -22004, -22153, -22301, -22448, -22594, -22739, -22883, -23027,
+ -23169, -23311, -23452, -23592, -23731, -23869, -24006, -24143,
+ -24278, -24413, -24546, -24679, -24811, -24942, -25072, -25201,
+ -25329, -25456, -25582, -25707, -25831, -25954, -26077, -26198,
+ -26318, -26437, -26556, -26673, -26789, -26905, -27019, -27132,
+ -27244, -27355, -27466, -27575, -27683, -27790, -27896, -28001,
+ -28105, -28208, -28309, -28410, -28510, -28608, -28706, -28802,
+ -28897, -28992, -29085, -29177, -29268, -29358, -29446, -29534,
+ -29621, -29706, -29790, -29873, -29955, -30036, -30116, -30195,
+ -30272, -30349, -30424, -30498, -30571, -30643, -30713, -30783,
+ -30851, -30918, -30984, -31049, -31113, -31175, -31236, -31297,
+ -31356, -31413, -31470, -31525, -31580, -31633, -31684, -31735,
+ -31785, -31833, -31880, -31926, -31970, -32014, -32056, -32097,
+ -32137, -32176, -32213, -32249, -32284, -32318, -32350, -32382,
+ -32412, -32441, -32468, -32495, -32520, -32544, -32567, -32588,
+ -32609, -32628, -32646, -32662, -32678, -32692, -32705, -32717,
+ -32727, -32736, -32744, -32751, -32757, -32761, -32764, -32766,
+ -32767, -32766, -32764, -32761, -32757, -32751, -32744, -32736,
+ -32727, -32717, -32705, -32692, -32678, -32662, -32646, -32628,
+ -32609, -32588, -32567, -32544, -32520, -32495, -32468, -32441,
+ -32412, -32382, -32350, -32318, -32284, -32249, -32213, -32176,
+ -32137, -32097, -32056, -32014, -31970, -31926, -31880, -31833,
+ -31785, -31735, -31684, -31633, -31580, -31525, -31470, -31413,
+ -31356, -31297, -31236, -31175, -31113, -31049, -30984, -30918,
+ -30851, -30783, -30713, -30643, -30571, -30498, -30424, -30349,
+ -30272, -30195, -30116, -30036, -29955, -29873, -29790, -29706,
+ -29621, -29534, -29446, -29358, -29268, -29177, -29085, -28992,
+ -28897, -28802, -28706, -28608, -28510, -28410, -28309, -28208,
+ -28105, -28001, -27896, -27790, -27683, -27575, -27466, -27355,
+ -27244, -27132, -27019, -26905, -26789, -26673, -26556, -26437,
+ -26318, -26198, -26077, -25954, -25831, -25707, -25582, -25456,
+ -25329, -25201, -25072, -24942, -24811, -24679, -24546, -24413,
+ -24278, -24143, -24006, -23869, -23731, -23592, -23452, -23311,
+ -23169, -23027, -22883, -22739, -22594, -22448, -22301, -22153,
+ -22004, -21855, -21705, -21554, -21402, -21249, -21096, -20942,
+ -20787, -20631, -20474, -20317, -20159, -20000, -19840, -19680,
+ -19519, -19357, -19194, -19031, -18867, -18702, -18537, -18371,
+ -18204, -18036, -17868, -17699, -17530, -17360, -17189, -17017,
+ -16845, -16672, -16499, -16325, -16150, -15975, -15799, -15623,
+ -15446, -15268, -15090, -14911, -14732, -14552, -14372, -14191,
+ -14009, -13827, -13645, -13462, -13278, -13094, -12909, -12724,
+ -12539, -12353, -12166, -11980, -11792, -11604, -11416, -11227,
+ -11038, -10849, -10659, -10469, -10278, -10087,  -9895,  -9703,
+  -9511,  -9319,  -9126,  -8932,  -8739,  -8545,  -8351,  -8156,
+  -7961,  -7766,  -7571,  -7375,  -7179,  -6982,  -6786,  -6589,
+  -6392,  -6195,  -5997,  -5799,  -5601,  -5403,  -5205,  -5006,
+  -4807,  -4608,  -4409,  -4210,  -4011,  -3811,  -3611,  -3411,
+  -3211,  -3011,  -2811,  -2610,  -2410,  -2209,  -2009,  -1808,
+  -1607,  -1406,  -1206,  -1005,   -804,   -603,   -402,   -201
+};
+
+int WebRtcSpl_ComplexFFT(WebRtc_Word16 frfi[], int stages, int mode)
+{
+    int i, j, l, k, istep, n, m;
+    WebRtc_Word16 wr, wi;
+    WebRtc_Word32 tr32, ti32, qr32, qi32;
+
+    /* The 1024-value is a constant given from the size of kSinTable1024[],
+     * and should not be changed depending on the input parameter 'stages'
+     */
+    n = 1 << stages;
+    if (n > 1024)
+        return -1;
+
+    l = 1;
+    k = 10 - 1; /* Constant for given kSinTable1024[]. Do not change
+         depending on the input parameter 'stages' */
+
+    if (mode == 0)
+    {
+        // mode==0: Low-complexity and Low-accuracy mode
+        while (l < n)
+        {
+            istep = l << 1;
+
+            for (m = 0; m < l; ++m)
+            {
+                j = m << k;
+
+                /* The 256-value is a constant given as 1/4 of the size of
+                 * kSinTable1024[], and should not be changed depending on the input
+                 * parameter 'stages'. It will result in 0 <= j < N_SINE_WAVE/2
+                 */
+                wr = kSinTable1024[j + 256];
+                wi = -kSinTable1024[j];
+
+                for (i = m; i < n; i += istep)
+                {
+                    j = i + l;
+
+                    tr32 = WEBRTC_SPL_RSHIFT_W32((WEBRTC_SPL_MUL_16_16(wr, frfi[2 * j])
+                            - WEBRTC_SPL_MUL_16_16(wi, frfi[2 * j + 1])), 15);
+
+                    ti32 = WEBRTC_SPL_RSHIFT_W32((WEBRTC_SPL_MUL_16_16(wr, frfi[2 * j + 1])
+                            + WEBRTC_SPL_MUL_16_16(wi, frfi[2 * j])), 15);
+
+                    qr32 = (WebRtc_Word32)frfi[2 * i];
+                    qi32 = (WebRtc_Word32)frfi[2 * i + 1];
+                    frfi[2 * j] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qr32 - tr32, 1);
+                    frfi[2 * j + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qi32 - ti32, 1);
+                    frfi[2 * i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qr32 + tr32, 1);
+                    frfi[2 * i + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qi32 + ti32, 1);
+                }
+            }
+
+            --k;
+            l = istep;
+
+        }
+
+    } else
+    {
+        // mode==1: High-complexity and High-accuracy mode
+        while (l < n)
+        {
+            istep = l << 1;
+
+            for (m = 0; m < l; ++m)
+            {
+                j = m << k;
+
+                /* The 256-value is a constant given as 1/4 of the size of
+                 * kSinTable1024[], and should not be changed depending on the input
+                 * parameter 'stages'. It will result in 0 <= j < N_SINE_WAVE/2
+                 */
+                wr = kSinTable1024[j + 256];
+                wi = -kSinTable1024[j];
+
+#ifdef WEBRTC_ARCH_ARM_V7A
+                WebRtc_Word32 wri;
+                WebRtc_Word32 frfi_r;
+                __asm__("pkhbt %0, %1, %2, lsl #16" : "=r"(wri) :
+                    "r"((WebRtc_Word32)wr), "r"((WebRtc_Word32)wi));
+#endif
+
+                for (i = m; i < n; i += istep)
+                {
+                    j = i + l;
+
+#ifdef WEBRTC_ARCH_ARM_V7A
+                    __asm__("pkhbt %0, %1, %2, lsl #16" : "=r"(frfi_r) :
+                        "r"((WebRtc_Word32)frfi[2*j]), "r"((WebRtc_Word32)frfi[2*j +1]));
+                    __asm__("smlsd %0, %1, %2, %3" : "=r"(tr32) :
+                        "r"(wri), "r"(frfi_r), "r"(CFFTRND));
+                    __asm__("smladx %0, %1, %2, %3" : "=r"(ti32) :
+                        "r"(wri), "r"(frfi_r), "r"(CFFTRND));
+    
+#else
+                    tr32 = WEBRTC_SPL_MUL_16_16(wr, frfi[2 * j])
+                            - WEBRTC_SPL_MUL_16_16(wi, frfi[2 * j + 1]) + CFFTRND;
+
+                    ti32 = WEBRTC_SPL_MUL_16_16(wr, frfi[2 * j + 1])
+                            + WEBRTC_SPL_MUL_16_16(wi, frfi[2 * j]) + CFFTRND;
+#endif
+
+                    tr32 = WEBRTC_SPL_RSHIFT_W32(tr32, 15 - CFFTSFT);
+                    ti32 = WEBRTC_SPL_RSHIFT_W32(ti32, 15 - CFFTSFT);
+
+                    qr32 = ((WebRtc_Word32)frfi[2 * i]) << CFFTSFT;
+                    qi32 = ((WebRtc_Word32)frfi[2 * i + 1]) << CFFTSFT;
+
+                    frfi[2 * j] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(
+                            (qr32 - tr32 + CFFTRND2), 1 + CFFTSFT);
+                    frfi[2 * j + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(
+                            (qi32 - ti32 + CFFTRND2), 1 + CFFTSFT);
+                    frfi[2 * i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(
+                            (qr32 + tr32 + CFFTRND2), 1 + CFFTSFT);
+                    frfi[2 * i + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(
+                            (qi32 + ti32 + CFFTRND2), 1 + CFFTSFT);
+                }
+            }
+
+            --k;
+            l = istep;
+        }
+    }
+    return 0;
+}
+
+int WebRtcSpl_ComplexIFFT(WebRtc_Word16 frfi[], int stages, int mode)
+{
+    int i, j, l, k, istep, n, m, scale, shift;
+    WebRtc_Word16 wr, wi;
+    WebRtc_Word32 tr32, ti32, qr32, qi32;
+    WebRtc_Word32 tmp32, round2;
+
+    /* The 1024-value is a constant given from the size of kSinTable1024[],
+     * and should not be changed depending on the input parameter 'stages'
+     */
+    n = 1 << stages;
+    if (n > 1024)
+        return -1;
+
+    scale = 0;
+
+    l = 1;
+    k = 10 - 1; /* Constant for given kSinTable1024[]. Do not change
+         depending on the input parameter 'stages' */
+
+    while (l < n)
+    {
+        // variable scaling, depending upon data
+        shift = 0;
+        round2 = 8192;
+
+        tmp32 = (WebRtc_Word32)WebRtcSpl_MaxAbsValueW16(frfi, 2 * n);
+        if (tmp32 > 13573)
+        {
+            shift++;
+            scale++;
+            round2 <<= 1;
+        }
+        if (tmp32 > 27146)
+        {
+            shift++;
+            scale++;
+            round2 <<= 1;
+        }
+
+        istep = l << 1;
+
+        if (mode == 0)
+        {
+            // mode==0: Low-complexity and Low-accuracy mode
+            for (m = 0; m < l; ++m)
+            {
+                j = m << k;
+
+                /* The 256-value is a constant given as 1/4 of the size of
+                 * kSinTable1024[], and should not be changed depending on the input
+                 * parameter 'stages'. It will result in 0 <= j < N_SINE_WAVE/2
+                 */
+                wr = kSinTable1024[j + 256];
+                wi = kSinTable1024[j];
+
+                for (i = m; i < n; i += istep)
+                {
+                    j = i + l;
+
+                    tr32 = WEBRTC_SPL_RSHIFT_W32((WEBRTC_SPL_MUL_16_16_RSFT(wr, frfi[2 * j], 0)
+                            - WEBRTC_SPL_MUL_16_16_RSFT(wi, frfi[2 * j + 1], 0)), 15);
+
+                    ti32 = WEBRTC_SPL_RSHIFT_W32(
+                            (WEBRTC_SPL_MUL_16_16_RSFT(wr, frfi[2 * j + 1], 0)
+                                    + WEBRTC_SPL_MUL_16_16_RSFT(wi,frfi[2*j],0)), 15);
+
+                    qr32 = (WebRtc_Word32)frfi[2 * i];
+                    qi32 = (WebRtc_Word32)frfi[2 * i + 1];
+                    frfi[2 * j] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qr32 - tr32, shift);
+                    frfi[2 * j + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qi32 - ti32, shift);
+                    frfi[2 * i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qr32 + tr32, shift);
+                    frfi[2 * i + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qi32 + ti32, shift);
+                }
+            }
+        } else
+        {
+            // mode==1: High-complexity and High-accuracy mode
+
+            for (m = 0; m < l; ++m)
+            {
+                j = m << k;
+
+                /* The 256-value is a constant given as 1/4 of the size of
+                 * kSinTable1024[], and should not be changed depending on the input
+                 * parameter 'stages'. It will result in 0 <= j < N_SINE_WAVE/2
+                 */
+                wr = kSinTable1024[j + 256];
+                wi = kSinTable1024[j];
+
+#ifdef WEBRTC_ARCH_ARM_V7A
+                WebRtc_Word32 wri;
+                WebRtc_Word32 frfi_r;
+                __asm__("pkhbt %0, %1, %2, lsl #16" : "=r"(wri) :
+                    "r"((WebRtc_Word32)wr), "r"((WebRtc_Word32)wi));
+#endif
+
+                for (i = m; i < n; i += istep)
+                {
+                    j = i + l;
+
+#ifdef WEBRTC_ARCH_ARM_V7A
+                    __asm__("pkhbt %0, %1, %2, lsl #16" : "=r"(frfi_r) :
+                        "r"((WebRtc_Word32)frfi[2*j]), "r"((WebRtc_Word32)frfi[2*j +1]));
+                    __asm__("smlsd %0, %1, %2, %3" : "=r"(tr32) :
+                        "r"(wri), "r"(frfi_r), "r"(CIFFTRND));
+                    __asm__("smladx %0, %1, %2, %3" : "=r"(ti32) :
+                        "r"(wri), "r"(frfi_r), "r"(CIFFTRND));
+#else
+
+                    tr32 = WEBRTC_SPL_MUL_16_16(wr, frfi[2 * j])
+                            - WEBRTC_SPL_MUL_16_16(wi, frfi[2 * j + 1]) + CIFFTRND;
+
+                    ti32 = WEBRTC_SPL_MUL_16_16(wr, frfi[2 * j + 1])
+                            + WEBRTC_SPL_MUL_16_16(wi, frfi[2 * j]) + CIFFTRND;
+#endif
+                    tr32 = WEBRTC_SPL_RSHIFT_W32(tr32, 15 - CIFFTSFT);
+                    ti32 = WEBRTC_SPL_RSHIFT_W32(ti32, 15 - CIFFTSFT);
+
+                    qr32 = ((WebRtc_Word32)frfi[2 * i]) << CIFFTSFT;
+                    qi32 = ((WebRtc_Word32)frfi[2 * i + 1]) << CIFFTSFT;
+
+                    frfi[2 * j] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((qr32 - tr32+round2),
+                                                                       shift+CIFFTSFT);
+                    frfi[2 * j + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(
+                            (qi32 - ti32 + round2), shift + CIFFTSFT);
+                    frfi[2 * i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((qr32 + tr32 + round2),
+                                                                       shift + CIFFTSFT);
+                    frfi[2 * i + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(
+                            (qi32 + ti32 + round2), shift + CIFFTSFT);
+                }
+            }
+
+        }
+        --k;
+        l = istep;
+    }
+    return scale;
+}
diff --git a/trunk/src/common_audio/signal_processing/copy_set_operations.c b/trunk/src/common_audio/signal_processing/copy_set_operations.c
new file mode 100644
index 0000000..8247337
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/copy_set_operations.c
@@ -0,0 +1,108 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains the implementation of functions
+ * WebRtcSpl_MemSetW16()
+ * WebRtcSpl_MemSetW32()
+ * WebRtcSpl_MemCpyReversedOrder()
+ * WebRtcSpl_CopyFromEndW16()
+ * WebRtcSpl_ZerosArrayW16()
+ * WebRtcSpl_ZerosArrayW32()
+ * WebRtcSpl_OnesArrayW16()
+ * WebRtcSpl_OnesArrayW32()
+ *
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include <string.h>
+#include "signal_processing_library.h"
+
+
+void WebRtcSpl_MemSetW16(WebRtc_Word16 *ptr, WebRtc_Word16 set_value, int length)
+{
+    int j;
+    WebRtc_Word16 *arrptr = ptr;
+
+    for (j = length; j > 0; j--)
+    {
+        *arrptr++ = set_value;
+    }
+}
+
+void WebRtcSpl_MemSetW32(WebRtc_Word32 *ptr, WebRtc_Word32 set_value, int length)
+{
+    int j;
+    WebRtc_Word32 *arrptr = ptr;
+
+    for (j = length; j > 0; j--)
+    {
+        *arrptr++ = set_value;
+    }
+}
+
+void WebRtcSpl_MemCpyReversedOrder(WebRtc_Word16* dest, WebRtc_Word16* source, int length)
+{
+    int j;
+    WebRtc_Word16* destPtr = dest;
+    WebRtc_Word16* sourcePtr = source;
+
+    for (j = 0; j < length; j++)
+    {
+        *destPtr-- = *sourcePtr++;
+    }
+}
+
+WebRtc_Word16 WebRtcSpl_CopyFromEndW16(G_CONST WebRtc_Word16 *vector_in,
+                                       WebRtc_Word16 length,
+                                       WebRtc_Word16 samples,
+                                       WebRtc_Word16 *vector_out)
+{
+    // Copy the last <samples> of the input vector to vector_out
+    WEBRTC_SPL_MEMCPY_W16(vector_out, &vector_in[length - samples], samples);
+
+    return samples;
+}
+
+WebRtc_Word16 WebRtcSpl_ZerosArrayW16(WebRtc_Word16 *vector, WebRtc_Word16 length)
+{
+    WebRtcSpl_MemSetW16(vector, 0, length);
+    return length;
+}
+
+WebRtc_Word16 WebRtcSpl_ZerosArrayW32(WebRtc_Word32 *vector, WebRtc_Word16 length)
+{
+    WebRtcSpl_MemSetW32(vector, 0, length);
+    return length;
+}
+
+WebRtc_Word16 WebRtcSpl_OnesArrayW16(WebRtc_Word16 *vector, WebRtc_Word16 length)
+{
+    WebRtc_Word16 i;
+    WebRtc_Word16 *tmpvec = vector;
+    for (i = 0; i < length; i++)
+    {
+        *tmpvec++ = 1;
+    }
+    return length;
+}
+
+WebRtc_Word16 WebRtcSpl_OnesArrayW32(WebRtc_Word32 *vector, WebRtc_Word16 length)
+{
+    WebRtc_Word16 i;
+    WebRtc_Word32 *tmpvec = vector;
+    for (i = 0; i < length; i++)
+    {
+        *tmpvec++ = 1;
+    }
+    return length;
+}
diff --git a/trunk/src/common_audio/signal_processing/cross_correlation.c b/trunk/src/common_audio/signal_processing/cross_correlation.c
new file mode 100644
index 0000000..726a749
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/cross_correlation.c
@@ -0,0 +1,271 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains the function WebRtcSpl_CrossCorrelation().
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+/* TODO(kma): Clean up the code in this file, and break it up for
+ * various platforms (Xscale, ARM/Neon etc.).
+ */
+
+#include "signal_processing_library.h"
+
+void WebRtcSpl_CrossCorrelation(WebRtc_Word32* cross_correlation, WebRtc_Word16* seq1,
+                                WebRtc_Word16* seq2, WebRtc_Word16 dim_seq,
+                                WebRtc_Word16 dim_cross_correlation,
+                                WebRtc_Word16 right_shifts,
+                                WebRtc_Word16 step_seq2)
+{
+    int i, j;
+    WebRtc_Word16* seq1Ptr;
+    WebRtc_Word16* seq2Ptr;
+    WebRtc_Word32* CrossCorrPtr;
+
+#ifdef _XSCALE_OPT_
+
+#ifdef _WIN32
+#pragma message("NOTE: _XSCALE_OPT_ optimizations are used (overrides _ARM_OPT_ and requires /QRxscale compiler flag)")
+#endif
+
+    __int64 macc40;
+
+    int iseq1[250];
+    int iseq2[250];
+    int iseq3[250];
+    int * iseq1Ptr;
+    int * iseq2Ptr;
+    int * iseq3Ptr;
+    int len, i_len;
+
+    seq1Ptr = seq1;
+    iseq1Ptr = iseq1;
+    for(i = 0; i < ((dim_seq + 1) >> 1); i++)
+    {
+        *iseq1Ptr = (unsigned short)*seq1Ptr++;
+        *iseq1Ptr++ |= (WebRtc_Word32)*seq1Ptr++ << 16;
+
+    }
+
+    if(dim_seq%2)
+    {
+        *(iseq1Ptr-1) &= 0x0000ffff;
+    }
+    *iseq1Ptr = 0;
+    iseq1Ptr++;
+    *iseq1Ptr = 0;
+    iseq1Ptr++;
+    *iseq1Ptr = 0;
+
+    if(step_seq2 < 0)
+    {
+        seq2Ptr = seq2 - dim_cross_correlation + 1;
+        CrossCorrPtr = &cross_correlation[dim_cross_correlation - 1];
+    }
+    else
+    {
+        seq2Ptr = seq2;
+        CrossCorrPtr = cross_correlation;
+    }
+
+    len = dim_seq + dim_cross_correlation - 1;
+    i_len = (len + 1) >> 1;
+    iseq2Ptr = iseq2;
+
+    iseq3Ptr = iseq3;
+    for(i = 0; i < i_len; i++)
+    {
+        *iseq2Ptr = (unsigned short)*seq2Ptr++;
+        *iseq3Ptr = (unsigned short)*seq2Ptr;
+        *iseq2Ptr++ |= (WebRtc_Word32)*seq2Ptr++ << 16;
+        *iseq3Ptr++ |= (WebRtc_Word32)*seq2Ptr << 16;
+    }
+
+    if(len % 2)
+    {
+        iseq2[i_len - 1] &= 0x0000ffff;
+        iseq3[i_len - 1] = 0;
+    }
+    else
+    iseq3[i_len - 1] &= 0x0000ffff;
+
+    iseq2[i_len] = 0;
+    iseq3[i_len] = 0;
+    iseq2[i_len + 1] = 0;
+    iseq3[i_len + 1] = 0;
+    iseq2[i_len + 2] = 0;
+    iseq3[i_len + 2] = 0;
+
+    // Set pointer to start value
+    iseq2Ptr = iseq2;
+    iseq3Ptr = iseq3;
+
+    i_len = (dim_seq + 7) >> 3;
+    for (i = 0; i < dim_cross_correlation; i++)
+    {
+
+        iseq1Ptr = iseq1;
+
+        macc40 = 0;
+
+        _WriteCoProcessor(macc40, 0);
+
+        if((i & 1))
+        {
+            iseq3Ptr = iseq3 + (i >> 1);
+            for (j = i_len; j > 0; j--)
+            {
+                _SmulAddPack_2SW_ACC(*iseq1Ptr++, *iseq3Ptr++);
+                _SmulAddPack_2SW_ACC(*iseq1Ptr++, *iseq3Ptr++);
+                _SmulAddPack_2SW_ACC(*iseq1Ptr++, *iseq3Ptr++);
+                _SmulAddPack_2SW_ACC(*iseq1Ptr++, *iseq3Ptr++);
+            }
+        }
+        else
+        {
+            iseq2Ptr = iseq2 + (i >> 1);
+            for (j = i_len; j > 0; j--)
+            {
+                _SmulAddPack_2SW_ACC(*iseq1Ptr++, *iseq2Ptr++);
+                _SmulAddPack_2SW_ACC(*iseq1Ptr++, *iseq2Ptr++);
+                _SmulAddPack_2SW_ACC(*iseq1Ptr++, *iseq2Ptr++);
+                _SmulAddPack_2SW_ACC(*iseq1Ptr++, *iseq2Ptr++);
+            }
+
+        }
+
+        macc40 = _ReadCoProcessor(0);
+        *CrossCorrPtr = (WebRtc_Word32)(macc40 >> right_shifts);
+        CrossCorrPtr += step_seq2;
+    }
+#else // #ifdef _XSCALE_OPT_
+#ifdef _ARM_OPT_
+    WebRtc_Word16 dim_seq8 = (dim_seq >> 3) << 3;
+#endif
+
+    CrossCorrPtr = cross_correlation;
+
+    for (i = 0; i < dim_cross_correlation; i++)
+    {
+        // Set the pointer to the static vector, set the pointer to the sliding vector
+        // and initialize cross_correlation
+        seq1Ptr = seq1;
+        seq2Ptr = seq2 + (step_seq2 * i);
+        (*CrossCorrPtr) = 0;
+
+#ifndef _ARM_OPT_ 
+#ifdef _WIN32
+#pragma message("NOTE: default implementation is used")
+#endif
+        // Perform the cross correlation
+        for (j = 0; j < dim_seq; j++)
+        {
+            (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16_RSFT((*seq1Ptr), (*seq2Ptr), right_shifts);
+            seq1Ptr++;
+            seq2Ptr++;
+        }
+#else
+#ifdef _WIN32
+#pragma message("NOTE: _ARM_OPT_ optimizations are used")
+#endif
+        if (right_shifts == 0)
+        {
+            // Perform the optimized cross correlation
+            for (j = 0; j < dim_seq8; j = j + 8)
+            {
+                (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16((*seq1Ptr), (*seq2Ptr));
+                seq1Ptr++;
+                seq2Ptr++;
+                (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16((*seq1Ptr), (*seq2Ptr));
+                seq1Ptr++;
+                seq2Ptr++;
+                (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16((*seq1Ptr), (*seq2Ptr));
+                seq1Ptr++;
+                seq2Ptr++;
+                (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16((*seq1Ptr), (*seq2Ptr));
+                seq1Ptr++;
+                seq2Ptr++;
+                (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16((*seq1Ptr), (*seq2Ptr));
+                seq1Ptr++;
+                seq2Ptr++;
+                (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16((*seq1Ptr), (*seq2Ptr));
+                seq1Ptr++;
+                seq2Ptr++;
+                (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16((*seq1Ptr), (*seq2Ptr));
+                seq1Ptr++;
+                seq2Ptr++;
+                (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16((*seq1Ptr), (*seq2Ptr));
+                seq1Ptr++;
+                seq2Ptr++;
+            }
+
+            for (j = dim_seq8; j < dim_seq; j++)
+            {
+                (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16((*seq1Ptr), (*seq2Ptr));
+                seq1Ptr++;
+                seq2Ptr++;
+            }
+        }
+        else // right_shifts != 0
+
+        {
+            // Perform the optimized cross correlation
+            for (j = 0; j < dim_seq8; j = j + 8)
+            {
+                (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16_RSFT((*seq1Ptr), (*seq2Ptr),
+                                                             right_shifts);
+                seq1Ptr++;
+                seq2Ptr++;
+                (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16_RSFT((*seq1Ptr), (*seq2Ptr),
+                                                             right_shifts);
+                seq1Ptr++;
+                seq2Ptr++;
+                (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16_RSFT((*seq1Ptr), (*seq2Ptr),
+                                                             right_shifts);
+                seq1Ptr++;
+                seq2Ptr++;
+                (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16_RSFT((*seq1Ptr), (*seq2Ptr),
+                                                             right_shifts);
+                seq1Ptr++;
+                seq2Ptr++;
+                (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16_RSFT((*seq1Ptr), (*seq2Ptr),
+                                                             right_shifts);
+                seq1Ptr++;
+                seq2Ptr++;
+                (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16_RSFT((*seq1Ptr), (*seq2Ptr),
+                                                             right_shifts);
+                seq1Ptr++;
+                seq2Ptr++;
+                (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16_RSFT((*seq1Ptr), (*seq2Ptr),
+                                                             right_shifts);
+                seq1Ptr++;
+                seq2Ptr++;
+                (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16_RSFT((*seq1Ptr), (*seq2Ptr),
+                                                             right_shifts);
+                seq1Ptr++;
+                seq2Ptr++;
+            }
+
+            for (j = dim_seq8; j < dim_seq; j++)
+            {
+                (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16_RSFT((*seq1Ptr), (*seq2Ptr),
+                                                             right_shifts);
+                seq1Ptr++;
+                seq2Ptr++;
+            }
+        }
+#endif
+        CrossCorrPtr++;
+    }
+#endif
+}
diff --git a/trunk/src/common_audio/signal_processing/cross_correlation_neon.s b/trunk/src/common_audio/signal_processing/cross_correlation_neon.s
new file mode 100644
index 0000000..e9b1c69
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/cross_correlation_neon.s
@@ -0,0 +1,168 @@
+@
+@ Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+@
+@ Use of this source code is governed by a BSD-style license
+@ that can be found in the LICENSE file in the root of the source
+@ tree. An additional intellectual property rights grant can be found
+@ in the file PATENTS.  All contributing project authors may
+@ be found in the AUTHORS file in the root of the source tree.
+@
+
+@ cross_correlation_neon.s
+@ This file contains the function WebRtcSpl_CrossCorrelation(),
+@ optimized for ARM Neon platform.
+@
+@ Reference Ccode at end of this file.
+@ Output is bit-exact with the reference C code, but not with the generic
+@ C code in file cross_correlation.c, due to reduction of shift operations
+@ from using Neon registers.
+
+@ Register usage:
+@
+@ r0: *cross_correlation (function argument)
+@ r1: *seq1 (function argument)
+@ r2: *seq2 (function argument)
+@ r3: dim_seq (function argument); then, total iteration of LOOP_DIM_SEQ
+@ r4: counter for LOOP_DIM_CROSS_CORRELATION
+@ r5: seq2_ptr
+@ r6: seq1_ptr
+@ r7: Total iteration of LOOP_DIM_SEQ_RESIDUAL
+@ r8, r9, r10, r11, r12: scratch
+
+.arch armv7-a
+.fpu neon
+
+.align  2
+.global WebRtcSpl_CrossCorrelation
+
+WebRtcSpl_CrossCorrelation:
+
+.fnstart
+
+.save {r4-r11}
+  push {r4-r11}
+
+  @ Put the shift value (-right_shifts) into a Neon register.
+  ldrsh r10, [sp, #36]
+  rsb r10, r10, #0
+  mov r8, r10, asr #31
+  vmov.32 d16, r10, r8
+
+  @ Initialize loop counters.
+  and r7, r3, #7              @ inner_loop_len2 = dim_seq % 8;
+  asr r3, r3, #3              @ inner_loop_len1 = dim_seq / 8;
+  ldrsh r4, [sp, #32]         @ dim_cross_correlation
+
+LOOP_DIM_CROSS_CORRELATION:
+  vmov.i32 q9, #0
+  vmov.i32 q14, #0
+  movs r8, r3                 @ inner_loop_len1
+  mov r6, r1                  @ seq1_ptr
+  mov r5, r2                  @ seq2_ptr
+  ble POST_LOOP_DIM_SEQ
+
+LOOP_DIM_SEQ:
+  vld1.16 {d20, d21}, [r6]!   @ seq1_ptr
+  vld1.16 {d22, d23}, [r5]!   @ seq2_ptr 
+  subs r8, r8, #1
+  vmull.s16 q12, d20, d22
+  vmull.s16 q13, d21, d23
+  vpadal.s32 q9, q12
+  vpadal.s32 q14, q13
+  bgt LOOP_DIM_SEQ
+
+POST_LOOP_DIM_SEQ:
+  movs r10, r7                @ Loop counter
+  mov r12, #0
+  mov r8, #0
+  ble POST_LOOP_DIM_SEQ_RESIDUAL
+
+LOOP_DIM_SEQ_RESIDUAL:
+  ldrh r11, [r6], #2
+  ldrh r9, [r5], #2
+  smulbb r11, r11, r9
+  adds r8, r8, r11
+  adc r12, r12, r11, asr #31
+  subs r10, #1
+  bgt LOOP_DIM_SEQ_RESIDUAL
+
+POST_LOOP_DIM_SEQ_RESIDUAL:   @ Sum the results up and do the shift.
+  vadd.i64 d18, d19
+  vadd.i64 d28, d29
+  vadd.i64 d18, d28
+  vmov.32 d17[0], r8
+  vmov.32 d17[1], r12
+  vadd.i64 d17, d18
+  vshl.s64 d17, d16
+  vst1.32 d17[0], [r0]!       @ Store the output
+
+  ldr r8, [sp, #40]           @ step_seq2
+  add r2, r8, lsl #1          @ prepare for seq2_ptr(r5) in the next loop.
+
+  subs r4, #1
+  bgt LOOP_DIM_CROSS_CORRELATION
+
+  pop {r4-r11}
+  bx  lr
+
+.fnend
+
+
+@ TODO(kma): Place this piece of reference code into a C code file.
+@ void WebRtcSpl_CrossCorrelation(WebRtc_Word32* cross_correlation,
+@                                 WebRtc_Word16* seq1,
+@                                 WebRtc_Word16* seq2,
+@                                 WebRtc_Word16 dim_seq,
+@                                 WebRtc_Word16 dim_cross_correlation,
+@                                 WebRtc_Word16 right_shifts,
+@                                 WebRtc_Word16 step_seq2) {
+@   int i = 0;
+@   int j = 0;
+@   int inner_loop_len1 = dim_seq >> 3;
+@   int inner_loop_len2 = dim_seq - (inner_loop_len1 << 3);
+@ 
+@   assert(dim_cross_correlation > 0);
+@   assert(dim_seq > 0);
+@ 
+@   for (i = 0; i < dim_cross_correlation; i++) {
+@     int16_t *seq1_ptr = seq1;
+@     int16_t *seq2_ptr = seq2 + (step_seq2 * i);
+@     int64_t sum = 0;
+@ 
+@     for (j = inner_loop_len1; j > 0; j -= 1) {
+@       sum += WEBRTC_SPL_MUL_16_16(*seq1_ptr, *seq2_ptr);
+@       seq1_ptr++;
+@       seq2_ptr++;
+@       sum += WEBRTC_SPL_MUL_16_16(*seq1_ptr, *seq2_ptr);
+@       seq1_ptr++;
+@       seq2_ptr++;
+@       sum += WEBRTC_SPL_MUL_16_16(*seq1_ptr, *seq2_ptr);
+@       seq1_ptr++;
+@       seq2_ptr++;
+@       sum += WEBRTC_SPL_MUL_16_16(*seq1_ptr, *seq2_ptr);
+@       seq1_ptr++;
+@       seq2_ptr++;
+@       sum += WEBRTC_SPL_MUL_16_16(*seq1_ptr, *seq2_ptr);
+@       seq1_ptr++;
+@       seq2_ptr++;
+@       sum += WEBRTC_SPL_MUL_16_16(*seq1_ptr, *seq2_ptr);
+@       seq1_ptr++;
+@       seq2_ptr++;
+@       sum += WEBRTC_SPL_MUL_16_16(*seq1_ptr, *seq2_ptr);
+@       seq1_ptr++;
+@       seq2_ptr++;
+@       sum += WEBRTC_SPL_MUL_16_16(*seq1_ptr, *seq2_ptr);
+@       seq1_ptr++;
+@       seq2_ptr++;
+@     }
+@ 
+@     // Calculate the rest of the samples.
+@     for (j = inner_loop_len2; j > 0; j -= 1) {
+@       sum += WEBRTC_SPL_MUL_16_16(*seq1_ptr, *seq2_ptr);
+@       seq1_ptr++;
+@       seq2_ptr++;
+@     }
+@ 
+@     *cross_correlation++ = (int32_t)(sum >> right_shifts);
+@   }
+@ }
diff --git a/trunk/src/common_audio/signal_processing/division_operations.c b/trunk/src/common_audio/signal_processing/division_operations.c
new file mode 100644
index 0000000..b143373
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/division_operations.c
@@ -0,0 +1,144 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains implementations of the divisions
+ * WebRtcSpl_DivU32U16()
+ * WebRtcSpl_DivW32W16()
+ * WebRtcSpl_DivW32W16ResW16()
+ * WebRtcSpl_DivResultInQ31()
+ * WebRtcSpl_DivW32HiLow()
+ *
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "signal_processing_library.h"
+
+WebRtc_UWord32 WebRtcSpl_DivU32U16(WebRtc_UWord32 num, WebRtc_UWord16 den)
+{
+    // Guard against division with 0
+    if (den != 0)
+    {
+        return (WebRtc_UWord32)(num / den);
+    } else
+    {
+        return (WebRtc_UWord32)0xFFFFFFFF;
+    }
+}
+
+WebRtc_Word32 WebRtcSpl_DivW32W16(WebRtc_Word32 num, WebRtc_Word16 den)
+{
+    // Guard against division with 0
+    if (den != 0)
+    {
+        return (WebRtc_Word32)(num / den);
+    } else
+    {
+        return (WebRtc_Word32)0x7FFFFFFF;
+    }
+}
+
+WebRtc_Word16 WebRtcSpl_DivW32W16ResW16(WebRtc_Word32 num, WebRtc_Word16 den)
+{
+    // Guard against division with 0
+    if (den != 0)
+    {
+        return (WebRtc_Word16)(num / den);
+    } else
+    {
+        return (WebRtc_Word16)0x7FFF;
+    }
+}
+
+WebRtc_Word32 WebRtcSpl_DivResultInQ31(WebRtc_Word32 num, WebRtc_Word32 den)
+{
+    WebRtc_Word32 L_num = num;
+    WebRtc_Word32 L_den = den;
+    WebRtc_Word32 div = 0;
+    int k = 31;
+    int change_sign = 0;
+
+    if (num == 0)
+        return 0;
+
+    if (num < 0)
+    {
+        change_sign++;
+        L_num = -num;
+    }
+    if (den < 0)
+    {
+        change_sign++;
+        L_den = -den;
+    }
+    while (k--)
+    {
+        div <<= 1;
+        L_num <<= 1;
+        if (L_num >= L_den)
+        {
+            L_num -= L_den;
+            div++;
+        }
+    }
+    if (change_sign == 1)
+    {
+        div = -div;
+    }
+    return div;
+}
+
+WebRtc_Word32 WebRtcSpl_DivW32HiLow(WebRtc_Word32 num, WebRtc_Word16 den_hi,
+                                    WebRtc_Word16 den_low)
+{
+    WebRtc_Word16 approx, tmp_hi, tmp_low, num_hi, num_low;
+    WebRtc_Word32 tmpW32;
+
+    approx = (WebRtc_Word16)WebRtcSpl_DivW32W16((WebRtc_Word32)0x1FFFFFFF, den_hi);
+    // result in Q14 (Note: 3FFFFFFF = 0.5 in Q30)
+
+    // tmpW32 = 1/den = approx * (2.0 - den * approx) (in Q30)
+    tmpW32 = (WEBRTC_SPL_MUL_16_16(den_hi, approx) << 1)
+            + ((WEBRTC_SPL_MUL_16_16(den_low, approx) >> 15) << 1);
+    // tmpW32 = den * approx
+
+    tmpW32 = (WebRtc_Word32)0x7fffffffL - tmpW32; // result in Q30 (tmpW32 = 2.0-(den*approx))
+
+    // Store tmpW32 in hi and low format
+    tmp_hi = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmpW32, 16);
+    tmp_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((tmpW32
+            - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)tmp_hi, 16)), 1);
+
+    // tmpW32 = 1/den in Q29
+    tmpW32 = ((WEBRTC_SPL_MUL_16_16(tmp_hi, approx) + (WEBRTC_SPL_MUL_16_16(tmp_low, approx)
+            >> 15)) << 1);
+
+    // 1/den in hi and low format
+    tmp_hi = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmpW32, 16);
+    tmp_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((tmpW32
+            - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)tmp_hi, 16)), 1);
+
+    // Store num in hi and low format
+    num_hi = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(num, 16);
+    num_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((num
+            - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)num_hi, 16)), 1);
+
+    // num * (1/den) by 32 bit multiplication (result in Q28)
+
+    tmpW32 = (WEBRTC_SPL_MUL_16_16(num_hi, tmp_hi) + (WEBRTC_SPL_MUL_16_16(num_hi, tmp_low)
+            >> 15) + (WEBRTC_SPL_MUL_16_16(num_low, tmp_hi) >> 15));
+
+    // Put result in Q31 (convert from Q28)
+    tmpW32 = WEBRTC_SPL_LSHIFT_W32(tmpW32, 3);
+
+    return tmpW32;
+}
diff --git a/trunk/src/common_audio/signal_processing/dot_product_with_scale.c b/trunk/src/common_audio/signal_processing/dot_product_with_scale.c
new file mode 100644
index 0000000..6e085fd
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/dot_product_with_scale.c
@@ -0,0 +1,91 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains the function WebRtcSpl_DotProductWithScale().
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "signal_processing_library.h"
+
+WebRtc_Word32 WebRtcSpl_DotProductWithScale(WebRtc_Word16 *vector1, WebRtc_Word16 *vector2,
+                                            int length, int scaling)
+{
+    WebRtc_Word32 sum;
+    int i;
+#ifdef _ARM_OPT_
+#pragma message("NOTE: _ARM_OPT_ optimizations are used")
+    WebRtc_Word16 len4 = (length >> 2) << 2;
+#endif
+
+    sum = 0;
+
+#ifndef _ARM_OPT_
+    for (i = 0; i < length; i++)
+    {
+        sum += WEBRTC_SPL_MUL_16_16_RSFT(*vector1++, *vector2++, scaling);
+    }
+#else
+    if (scaling == 0)
+    {
+        for (i = 0; i < len4; i = i + 4)
+        {
+            sum += WEBRTC_SPL_MUL_16_16(*vector1, *vector2);
+            vector1++;
+            vector2++;
+            sum += WEBRTC_SPL_MUL_16_16(*vector1, *vector2);
+            vector1++;
+            vector2++;
+            sum += WEBRTC_SPL_MUL_16_16(*vector1, *vector2);
+            vector1++;
+            vector2++;
+            sum += WEBRTC_SPL_MUL_16_16(*vector1, *vector2);
+            vector1++;
+            vector2++;
+        }
+
+        for (i = len4; i < length; i++)
+        {
+            sum += WEBRTC_SPL_MUL_16_16(*vector1, *vector2);
+            vector1++;
+            vector2++;
+        }
+    }
+    else
+    {
+        for (i = 0; i < len4; i = i + 4)
+        {
+            sum += WEBRTC_SPL_MUL_16_16_RSFT(*vector1, *vector2, scaling);
+            vector1++;
+            vector2++;
+            sum += WEBRTC_SPL_MUL_16_16_RSFT(*vector1, *vector2, scaling);
+            vector1++;
+            vector2++;
+            sum += WEBRTC_SPL_MUL_16_16_RSFT(*vector1, *vector2, scaling);
+            vector1++;
+            vector2++;
+            sum += WEBRTC_SPL_MUL_16_16_RSFT(*vector1, *vector2, scaling);
+            vector1++;
+            vector2++;
+        }
+
+        for (i = len4; i < length; i++)
+        {
+            sum += WEBRTC_SPL_MUL_16_16_RSFT(*vector1, *vector2, scaling);
+            vector1++;
+            vector2++;
+        }
+    }
+#endif
+
+    return sum;
+}
diff --git a/trunk/src/common_audio/signal_processing/downsample_fast.c b/trunk/src/common_audio/signal_processing/downsample_fast.c
new file mode 100644
index 0000000..526cdca
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/downsample_fast.c
@@ -0,0 +1,47 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "signal_processing_library.h"
+
+// TODO(Bjornv): Change the function parameter order to WebRTC code style.
+int WebRtcSpl_DownsampleFast(const int16_t* data_in,
+                             int data_in_length,
+                             int16_t* data_out,
+                             int data_out_length,
+                             const int16_t* __restrict coefficients,
+                             int coefficients_length,
+                             int factor,
+                             int delay) {
+  int i = 0;
+  int j = 0;
+  int32_t out_s32 = 0;
+  int endpos = delay + factor * (data_out_length - 1) + 1;
+
+  // Return error if any of the running conditions doesn't meet.
+  if (data_out_length <= 0 || coefficients_length <= 0
+                           || data_in_length < endpos) {
+    return -1;
+  }
+
+  for (i = delay; i < endpos; i += factor) {
+    out_s32 = 2048;  // Round value, 0.5 in Q12.
+
+    for (j = 0; j < coefficients_length; j++) {
+      out_s32 += coefficients[j] * data_in[i - j];  // Q12.
+    }
+
+    out_s32 >>= 12;  // Q0.
+
+    // Saturate and store the output.
+    *data_out++ = WebRtcSpl_SatW32ToW16(out_s32);
+  }
+
+  return 0;
+}
diff --git a/trunk/src/common_audio/signal_processing/downsample_fast_neon.s b/trunk/src/common_audio/signal_processing/downsample_fast_neon.s
new file mode 100644
index 0000000..906b0a1
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/downsample_fast_neon.s
@@ -0,0 +1,222 @@
+@
+@ Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+@
+@ Use of this source code is governed by a BSD-style license
+@ that can be found in the LICENSE file in the root of the source
+@ tree. An additional intellectual property rights grant can be found
+@ in the file PATENTS.  All contributing project authors may
+@ be found in the AUTHORS file in the root of the source tree.
+@
+
+@ This file contains the function WebRtcSpl_DownsampleFast(), optimized for
+@ ARM Neon platform. The description header can be found in
+@ signal_processing_library.h
+@
+@ The reference C code is in file downsample_fast.c. Bit-exact.
+
+.arch armv7-a
+.fpu neon
+
+.align  2
+.global WebRtcSpl_DownsampleFast
+
+WebRtcSpl_DownsampleFast:
+
+.fnstart
+
+.save {r4-r11}
+  push {r4-r11}
+
+  cmp r3, #0                                @ data_out_length <= 0?
+  movle r0, #-1
+  ble END
+
+  ldrsh r12, [sp, #44]
+  ldr r5, [sp, #40]                         @ r5: factor
+  add r4, r12, #1                           @ r4: delay + 1
+  sub r3, r3, #1                            @ r3: data_out_length - 1
+  smulbb r3, r5, r3
+  ldr r8, [sp, #32]                         @ &coefficients[0]
+  mov r9, r12                               @ Iteration counter for outer loops.
+  add r3, r4                                @ delay + factor * (out_length-1) +1
+
+  cmp r3, r1                                @ data_in_length < endpos?
+  movgt r0, #-1
+  bgt END
+
+  @ Initializations.
+  sub r3, r5, asl #3
+  add r11, r0, r12, asl #1                  @ &data_in[delay]
+  ldr r0, [sp, #36]                         @ coefficients_length
+  add r3, r5                                @ endpos - factor * 7
+
+  cmp r0, #0                                @ coefficients_length <= 0 ?
+  movle r0, #-1
+  ble END
+
+  add r8, r0, asl #1                        @ &coeffieient[coefficients_length]
+  cmp r9, r3
+  bge POST_LOOP_ENDPOS                      @ branch when Iteration < 8 times.
+
+@
+@ First part, unroll the loop 8 times, with 3 subcases (factor == 2, 4, others)
+@
+  mov r4, #-2
+
+  @ Direct program flow to the right channel.
+
+  @ r10 is an offset to &data_in[] in the loop. After an iteration, we need to
+  @ move the pointer back to original after advancing 16 bytes by a vld1, and
+  @ then move 2 bytes forward to increment one more sample.
+  cmp r5, #2
+  moveq r10, #-14
+  beq LOOP_ENDPOS_FACTOR2                   @ Branch when factor == 2
+
+  @ Similar here, for r10, we need to move the pointer back to original after
+  @ advancing 32 bytes, then move 2 bytes forward to increment one sample.
+  cmp r5, #4
+  moveq r10, #-30
+  beq LOOP_ENDPOS_FACTOR4                   @ Branch when factor == 4
+
+  @ For r10, we need to move the pointer back to original after advancing
+  @ (factor * 7 * 2) bytes, then move 2 bytes forward to increment one sample.
+  mov r10, r5, asl #4
+  rsb r10, #2
+  add r10, r5, asl #1
+  lsl r5, #1                                @ r5 = factor * sizeof(data_in)
+
+@ The general case (factor != 2 && factor != 4)
+LOOP_ENDPOS_GENERAL:
+  @ Initializations.
+  vmov.i32 q2, #2048
+  vmov.i32 q3, #2048
+  sub r7, r8, #2
+  sub r12, r0, #1                           @ coefficients_length - 1
+  sub r1, r11, r12, asl #1                  @ &data_in[i - j]
+
+LOOP_COEFF_LENGTH_GENERAL:
+  vld1.16 {d2[], d3[]}, [r7], r4            @ coefficients[j]
+  vld1.16 d0[0], [r1], r5                   @ data_in[i - j]
+  vld1.16 d0[1], [r1], r5                   @ data_in[i + factor - j]
+  vld1.16 d0[2], [r1], r5                   @ data_in[i + factor * 2 - j]
+  vld1.16 d0[3], [r1], r5                   @ data_in[i + factor * 3 - j]
+  vld1.16 d1[0], [r1], r5                   @ data_in[i + factor * 4 - j]
+  vld1.16 d1[1], [r1], r5                   @ data_in[i + factor * 5 - j]
+  vld1.16 d1[2], [r1], r5                   @ data_in[i + factor * 6 - j]
+  vld1.16 d1[3], [r1], r10                  @ data_in[i + factor * 7 - j]
+  subs r12, #1
+  vmlal.s16 q2, d0, d2
+  vmlal.s16 q3, d1, d3
+  bge LOOP_COEFF_LENGTH_GENERAL
+
+  @ Shift, saturate, and store the result.
+  vqshrn.s32 d0, q2, #12
+  vqshrn.s32 d1, q3, #12
+  vst1.16 {d0, d1}, [r2]!
+
+  add r11, r5, asl #3                       @ r11 -> &data_in[i + factor * 8]
+  add r9, r5, asl #2                        @ Counter i = delay + factor * 8.
+  cmp r9, r3                                @ i < endpos - factor * 7 ?
+  blt LOOP_ENDPOS_GENERAL
+  asr r5, #1                                @ Restore r5 to the value of factor.
+  b POST_LOOP_ENDPOS
+
+@ The case for factor == 2.
+LOOP_ENDPOS_FACTOR2:
+  @ Initializations.
+  vmov.i32 q2, #2048
+  vmov.i32 q3, #2048
+  sub r7, r8, #2
+  sub r12, r0, #1                           @ coefficients_length - 1
+  sub r1, r11, r12, asl #1                  @ &data_in[i - j]
+
+LOOP_COEFF_LENGTH_FACTOR2:
+  vld1.16 {d16[], d17[]}, [r7], r4          @ coefficients[j]
+  vld2.16 {d0, d1}, [r1]!                   @ data_in[]
+  vld2.16 {d2, d3}, [r1], r10               @ data_in[]
+  subs r12, #1
+  vmlal.s16 q2, d0, d16
+  vmlal.s16 q3, d2, d17
+  bge LOOP_COEFF_LENGTH_FACTOR2
+
+  @ Shift, saturate, and store the result.
+  vqshrn.s32 d0, q2, #12
+  vqshrn.s32 d1, q3, #12
+  vst1.16 {d0, d1}, [r2]!
+
+  add r11, r5, asl #4                       @ r11 -> &data_in[i + factor * 8]
+  add r9, r5, asl #3                        @ Counter i = delay + factor * 8.
+  cmp r9, r3                                @ i < endpos - factor * 7 ?
+  blt LOOP_ENDPOS_FACTOR2
+  b POST_LOOP_ENDPOS
+
+@ The case for factor == 4.
+LOOP_ENDPOS_FACTOR4:
+  @ Initializations.
+  vmov.i32 q2, #2048
+  vmov.i32 q3, #2048
+  sub r7, r8, #2
+  sub r12, r0, #1                           @ coefficients_length - 1
+  sub r1, r11, r12, asl #1                  @ &data_in[i - j]
+
+LOOP_COEFF_LENGTH_FACTOR4:
+  vld1.16 {d16[], d17[]}, [r7], r4          @ coefficients[j]
+  vld4.16 {d0, d1, d2, d3}, [r1]!           @ data_in[]
+  vld4.16 {d18, d19, d20, d21}, [r1], r10   @ data_in[]
+  subs r12, #1
+  vmlal.s16 q2, d0, d16
+  vmlal.s16 q3, d18, d17
+  bge LOOP_COEFF_LENGTH_FACTOR4
+
+  @ Shift, saturate, and store the result.
+  vqshrn.s32 d0, q2, #12
+  vqshrn.s32 d1, q3, #12
+  vst1.16 {d0, d1}, [r2]!
+
+  add r11, r5, asl #4                       @ r11 -> &data_in[i + factor * 8]
+  add r9, r5, asl #3                        @ Counter i = delay + factor * 8.
+  cmp r9, r3                                @ i < endpos - factor * 7 ?
+  blt LOOP_ENDPOS_FACTOR4
+
+@
+@ Second part, do the rest iterations (if any).
+@
+
+POST_LOOP_ENDPOS:
+  add r3, r5, asl #3
+  sub r3, r5                                @ Restore r3 to endpos.
+  cmp r9, r3
+  movge r0, #0
+  bge END
+
+LOOP2_ENDPOS:
+  @ Initializations.
+  mov r7, r8
+  sub r12, r0, #1                           @ coefficients_length - 1
+  sub r6, r11, r12, asl #1                  @ &data_in[i - j]
+
+  mov r1, #2048
+
+LOOP2_COEFF_LENGTH:
+  ldrsh r4, [r7, #-2]!                      @ coefficients[j]
+  ldrsh r10, [r6], #2                       @ data_in[i - j]
+  smlabb r1, r4, r10, r1
+  subs r12, #1
+  bge LOOP2_COEFF_LENGTH
+
+  @ Shift, saturate, and store the result.
+  ssat r1, #16, r1, asr #12
+  strh r1, [r2], #2
+
+  add r11, r5, asl #1                       @ r11 -> &data_in[i + factor]
+  add r9, r5                                @ Counter i = delay + factor.
+  cmp r9, r3                                @ i < endpos?
+  blt LOOP2_ENDPOS
+
+  mov r0, #0
+
+END:
+  pop {r4-r11}
+  bx  lr
+
+.fnend
diff --git a/trunk/src/common_audio/signal_processing/energy.c b/trunk/src/common_audio/signal_processing/energy.c
new file mode 100644
index 0000000..e8fdf94
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/energy.c
@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains the function WebRtcSpl_Energy().
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "signal_processing_library.h"
+
+WebRtc_Word32 WebRtcSpl_Energy(WebRtc_Word16* vector, int vector_length, int* scale_factor)
+{
+    WebRtc_Word32 en = 0;
+    int i;
+    int scaling = WebRtcSpl_GetScalingSquare(vector, vector_length, vector_length);
+    int looptimes = vector_length;
+    WebRtc_Word16 *vectorptr = vector;
+
+    for (i = 0; i < looptimes; i++)
+    {
+        en += WEBRTC_SPL_MUL_16_16_RSFT(*vectorptr, *vectorptr, scaling);
+        vectorptr++;
+    }
+    *scale_factor = scaling;
+
+    return en;
+}
diff --git a/trunk/src/common_audio/signal_processing/filter_ar.c b/trunk/src/common_audio/signal_processing/filter_ar.c
new file mode 100644
index 0000000..24e83a6
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/filter_ar.c
@@ -0,0 +1,89 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains the function WebRtcSpl_FilterAR().
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "signal_processing_library.h"
+
+int WebRtcSpl_FilterAR(G_CONST WebRtc_Word16* a,
+                       int a_length,
+                       G_CONST WebRtc_Word16* x,
+                       int x_length,
+                       WebRtc_Word16* state,
+                       int state_length,
+                       WebRtc_Word16* state_low,
+                       int state_low_length,
+                       WebRtc_Word16* filtered,
+                       WebRtc_Word16* filtered_low,
+                       int filtered_low_length)
+{
+    WebRtc_Word32 o;
+    WebRtc_Word32 oLOW;
+    int i, j, stop;
+    G_CONST WebRtc_Word16* x_ptr = &x[0];
+    WebRtc_Word16* filteredFINAL_ptr = filtered;
+    WebRtc_Word16* filteredFINAL_LOW_ptr = filtered_low;
+
+    for (i = 0; i < x_length; i++)
+    {
+        // Calculate filtered[i] and filtered_low[i]
+        G_CONST WebRtc_Word16* a_ptr = &a[1];
+        WebRtc_Word16* filtered_ptr = &filtered[i - 1];
+        WebRtc_Word16* filtered_low_ptr = &filtered_low[i - 1];
+        WebRtc_Word16* state_ptr = &state[state_length - 1];
+        WebRtc_Word16* state_low_ptr = &state_low[state_length - 1];
+
+        o = (WebRtc_Word32)(*x_ptr++) << 12;
+        oLOW = (WebRtc_Word32)0;
+
+        stop = (i < a_length) ? i + 1 : a_length;
+        for (j = 1; j < stop; j++)
+        {
+            o -= WEBRTC_SPL_MUL_16_16(*a_ptr, *filtered_ptr--);
+            oLOW -= WEBRTC_SPL_MUL_16_16(*a_ptr++, *filtered_low_ptr--);
+        }
+        for (j = i + 1; j < a_length; j++)
+        {
+            o -= WEBRTC_SPL_MUL_16_16(*a_ptr, *state_ptr--);
+            oLOW -= WEBRTC_SPL_MUL_16_16(*a_ptr++, *state_low_ptr--);
+        }
+
+        o += (oLOW >> 12);
+        *filteredFINAL_ptr = (WebRtc_Word16)((o + (WebRtc_Word32)2048) >> 12);
+        *filteredFINAL_LOW_ptr++ = (WebRtc_Word16)(o - ((WebRtc_Word32)(*filteredFINAL_ptr++)
+                << 12));
+    }
+
+    // Save the filter state
+    if (x_length >= state_length)
+    {
+        WebRtcSpl_CopyFromEndW16(filtered, x_length, a_length - 1, state);
+        WebRtcSpl_CopyFromEndW16(filtered_low, x_length, a_length - 1, state_low);
+    } else
+    {
+        for (i = 0; i < state_length - x_length; i++)
+        {
+            state[i] = state[i + x_length];
+            state_low[i] = state_low[i + x_length];
+        }
+        for (i = 0; i < x_length; i++)
+        {
+            state[state_length - x_length + i] = filtered[i];
+            state[state_length - x_length + i] = filtered_low[i];
+        }
+    }
+
+    return x_length;
+}
diff --git a/trunk/src/common_audio/signal_processing/filter_ar_fast_q12.c b/trunk/src/common_audio/signal_processing/filter_ar_fast_q12.c
new file mode 100644
index 0000000..0402302
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/filter_ar_fast_q12.c
@@ -0,0 +1,43 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include <assert.h>
+
+#include "signal_processing_library.h"
+
+// TODO(bjornv): Change the return type to report errors.
+
+void WebRtcSpl_FilterARFastQ12(const int16_t* data_in,
+                               int16_t* data_out,
+                               const int16_t* __restrict coefficients,
+                               int coefficients_length,
+                               int data_length) {
+  int i = 0;
+  int j = 0;
+
+  assert(data_length > 0);
+  assert(coefficients_length > 1);
+
+  for (i = 0; i < data_length; i++) {
+    int32_t output = 0;
+    int32_t sum = 0;
+
+    for (j = coefficients_length - 1; j > 0; j--) {
+      sum += coefficients[j] * data_out[i - j];
+    }
+
+    output = coefficients[0] * data_in[i];
+    output -= sum;
+
+    // Saturate and store the output.
+    output = WEBRTC_SPL_SAT(134215679, output, -134217728);
+    data_out[i] = (int16_t)((output + 2048) >> 12);
+  }
+}
+
diff --git a/trunk/src/common_audio/signal_processing/filter_ar_fast_q12_armv7.s b/trunk/src/common_audio/signal_processing/filter_ar_fast_q12_armv7.s
new file mode 100644
index 0000000..5591bb8
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/filter_ar_fast_q12_armv7.s
@@ -0,0 +1,223 @@
+@
+@ Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+@
+@ Use of this source code is governed by a BSD-style license
+@ that can be found in the LICENSE file in the root of the source
+@ tree. An additional intellectual property rights grant can be found
+@ in the file PATENTS.  All contributing project authors may
+@ be found in the AUTHORS file in the root of the source tree.
+@
+
+@ This file contains the function WebRtcSpl_FilterARFastQ12(), optimized for
+@ ARMv7  platform. The description header can be found in
+@ signal_processing_library.h
+@
+@ Output is bit-exact with the generic C code as in filter_ar_fast_q12.c, and
+@ the reference C code at end of this file.
+
+@ Assumptions:
+@ (1) data_length > 0
+@ (2) coefficients_length > 1
+
+@ Register usage:
+@
+@ r0:  &data_in[i]
+@ r1:  &data_out[i], for result ouput
+@ r2:  &coefficients[0]
+@ r3:  coefficients_length
+@ r4:  Iteration counter for the outer loop.
+@ r5:  data_out[j] as multiplication inputs
+@ r6:  Calculated value for output data_out[]; interation counter for inner loop
+@ r7:  Partial sum of a filtering multiplication results
+@ r8:  Partial sum of a filtering multiplication results
+@ r9:  &data_out[], for filtering input; data_in[i]
+@ r10: coefficients[j]
+@ r11: Scratch
+@ r12: &coefficients[j]
+
+.arch armv7-a
+
+.align  2
+.global WebRtcSpl_FilterARFastQ12
+
+WebRtcSpl_FilterARFastQ12:
+
+.fnstart
+
+.save {r4-r11}
+  push {r4-r11}
+
+  ldrsh r12, [sp, #32]         @ data_length
+  subs r4, r12, #1
+  beq ODD_LENGTH               @ jump if data_length == 1
+
+LOOP_LENGTH:
+  add r12, r2, r3, lsl #1
+  sub r12, #4                  @ &coefficients[coefficients_length - 2]
+  sub r9, r1, r3, lsl #1
+  add r9, #2                   @ &data_out[i - coefficients_length + 1]
+  ldr r5, [r9], #4             @ data_out[i - coefficients_length + {1,2}]
+
+  mov r7, #0                   @ sum1
+  mov r8, #0                   @ sum2
+  subs r6, r3, #3              @ Iteration counter for inner loop.
+  beq ODD_A_LENGTH             @ branch if coefficients_length == 3
+  blt POST_LOOP_A_LENGTH       @ branch if coefficients_length == 2
+
+LOOP_A_LENGTH:
+  ldr r10, [r12], #-4          @ coefficients[j - 1], coefficients[j]
+  subs r6, #2
+  smlatt r8, r10, r5, r8       @ sum2 += coefficients[j] * data_out[i - j + 1];
+  smlatb r7, r10, r5, r7       @ sum1 += coefficients[j] * data_out[i - j];
+  smlabt r7, r10, r5, r7       @ coefficients[j - 1] * data_out[i - j + 1];
+  ldr r5, [r9], #4             @ data_out[i - j + 2],  data_out[i - j + 3]
+  smlabb r8, r10, r5, r8       @ coefficients[j - 1] * data_out[i - j + 2];
+  bgt LOOP_A_LENGTH
+  blt POST_LOOP_A_LENGTH
+
+ODD_A_LENGTH:
+  ldrsh r10, [r12, #2]         @ Filter coefficients coefficients[2]
+  sub r12, #2                  @ &coefficients[0]
+  smlabb r7, r10, r5, r7       @ sum1 += coefficients[2] * data_out[i - 2];
+  smlabt r8, r10, r5, r8       @ sum2 += coefficients[2] * data_out[i - 1];
+  ldr r5, [r9, #-2]            @ data_out[i - 1],  data_out[i]
+
+POST_LOOP_A_LENGTH:
+  ldr r10, [r12]               @ coefficients[0], coefficients[1]
+  smlatb r7, r10, r5, r7       @ sum1 += coefficients[1] * data_out[i - 1];
+
+  ldr r9, [r0], #4             @ data_in[i], data_in[i + 1]
+  smulbb r6, r10, r9           @ output1 = coefficients[0] * data_in[i];
+  sub r6, r7                   @ output1 -= sum1;
+
+  sbfx r11, r6, #12, #16
+  ssat r7, #16, r6, asr #12
+  cmp r7, r11
+  addeq r6, r6, #2048
+  ssat r6, #16, r6, asr #12
+  strh r6, [r1], #2            @ Store data_out[i]
+
+  smlatb r8, r10, r6, r8       @ sum2 += coefficients[1] * data_out[i];
+  smulbt r6, r10, r9           @ output2 = coefficients[0] * data_in[i + 1];
+  sub r6, r8                   @ output1 -= sum1;
+
+  sbfx r11, r6, #12, #16
+  ssat r7, #16, r6, asr #12
+  cmp r7, r11
+  addeq r6, r6, #2048
+  ssat r6, #16, r6, asr #12
+  strh r6, [r1], #2            @ Store data_out[i + 1]
+
+  subs r4, #2
+  bgt LOOP_LENGTH
+  blt END                      @ For even data_length, it's done. Jump to END.
+
+@ Process i = data_length -1, for the case of an odd length.
+ODD_LENGTH:
+  add r12, r2, r3, lsl #1
+  sub r12, #4                  @ &coefficients[coefficients_length - 2]
+  sub r9, r1, r3, lsl #1
+  add r9, #2                   @ &data_out[i - coefficients_length + 1]
+  mov r7, #0                   @ sum1
+  mov r8, #0                   @ sum1
+  subs r6, r3, #2              @ inner loop counter
+  beq EVEN_A_LENGTH            @ branch if coefficients_length == 2
+
+LOOP2_A_LENGTH:
+  ldr r10, [r12], #-4          @ coefficients[j - 1], coefficients[j]
+  ldr r5, [r9], #4             @ data_out[i - j],  data_out[i - j + 1]
+  subs r6, #2
+  smlatb r7, r10, r5, r7       @ sum1 += coefficients[j] * data_out[i - j];
+  smlabt r8, r10, r5, r8       @ coefficients[j - 1] * data_out[i - j + 1];
+  bgt LOOP2_A_LENGTH
+  addlt r12, #2
+  blt POST_LOOP2_A_LENGTH
+
+EVEN_A_LENGTH:
+  ldrsh r10, [r12, #2]         @ Filter coefficients coefficients[1]
+  ldrsh r5, [r9]               @ data_out[i - 1]
+  smlabb r7, r10, r5, r7       @ sum1 += coefficients[1] * data_out[i - 1];
+
+POST_LOOP2_A_LENGTH:
+  ldrsh r10, [r12]             @ Filter coefficients coefficients[0]
+  ldrsh r9, [r0]               @ data_in[i]
+  smulbb r6, r10, r9           @ output1 = coefficients[0] * data_in[i];
+  sub r6, r7                   @ output1 -= sum1;
+  sub r6, r8                   @ output1 -= sum1;
+  sbfx r8, r6, #12, #16
+  ssat r7, #16, r6, asr #12
+  cmp r7, r8
+  addeq r6, r6, #2048
+  ssat r6, #16, r6, asr #12
+  strh r6, [r1]                @ Store the data_out[i]
+
+END:
+  pop {r4-r11}
+  bx  lr
+
+.fnend
+
+
+@Reference C code:
+@
+@void WebRtcSpl_FilterARFastQ12(int16_t* data_in,
+@                               int16_t* data_out,
+@                               int16_t* __restrict coefficients,
+@                               int coefficients_length,
+@                               int data_length) {
+@  int i = 0;
+@  int j = 0;
+@
+@  for (i = 0; i < data_length - 1; i += 2) {
+@    int32_t output1 = 0;
+@    int32_t sum1 = 0;
+@    int32_t output2 = 0;
+@    int32_t sum2 = 0;
+@
+@    for (j = coefficients_length - 1; j > 2; j -= 2) {
+@      sum1 += coefficients[j]      * data_out[i - j];
+@      sum1 += coefficients[j - 1]  * data_out[i - j + 1];
+@      sum2 += coefficients[j]     * data_out[i - j + 1];
+@      sum2 += coefficients[j - 1] * data_out[i - j + 2];
+@    }
+@
+@    if (j == 2) {
+@      sum1 += coefficients[2] * data_out[i - 2];
+@      sum2 += coefficients[2] * data_out[i - 1];
+@    }
+@
+@    sum1 += coefficients[1] * data_out[i - 1];
+@    output1 = coefficients[0] * data_in[i];
+@    output1 -= sum1;
+@    // Saturate and store the output.
+@    output1 = WEBRTC_SPL_SAT(134215679, output1, -134217728);
+@    data_out[i] = (int16_t)((output1 + 2048) >> 12);
+@
+@    sum2 += coefficients[1] * data_out[i];
+@    output2 = coefficients[0] * data_in[i + 1];
+@    output2 -= sum2;
+@    // Saturate and store the output.
+@    output2 = WEBRTC_SPL_SAT(134215679, output2, -134217728);
+@    data_out[i + 1] = (int16_t)((output2 + 2048) >> 12);
+@  }
+@
+@  if (i == data_length - 1) {
+@    int32_t output1 = 0;
+@    int32_t sum1 = 0;
+@
+@    for (j = coefficients_length - 1; j > 1; j -= 2) {
+@      sum1 += coefficients[j]      * data_out[i - j];
+@      sum1 += coefficients[j - 1]  * data_out[i - j + 1];
+@    }
+@
+@    if (j == 1) {
+@      sum1 += coefficients[1] * data_out[i - 1];
+@    }
+@
+@    output1 = coefficients[0] * data_in[i];
+@    output1 -= sum1;
+@    // Saturate and store the output.
+@    output1 = WEBRTC_SPL_SAT(134215679, output1, -134217728);
+@    data_out[i] = (int16_t)((output1 + 2048) >> 12);
+@  }
+@}
diff --git a/trunk/src/common_audio/signal_processing/filter_ma_fast_q12.c b/trunk/src/common_audio/signal_processing/filter_ma_fast_q12.c
new file mode 100644
index 0000000..19ad9b1
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/filter_ma_fast_q12.c
@@ -0,0 +1,49 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains the function WebRtcSpl_FilterMAFastQ12().
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "signal_processing_library.h"
+
+void WebRtcSpl_FilterMAFastQ12(WebRtc_Word16* in_ptr,
+                               WebRtc_Word16* out_ptr,
+                               WebRtc_Word16* B,
+                               WebRtc_Word16 B_length,
+                               WebRtc_Word16 length)
+{
+    WebRtc_Word32 o;
+    int i, j;
+    for (i = 0; i < length; i++)
+    {
+        G_CONST WebRtc_Word16* b_ptr = &B[0];
+        G_CONST WebRtc_Word16* x_ptr = &in_ptr[i];
+
+        o = (WebRtc_Word32)0;
+
+        for (j = 0; j < B_length; j++)
+        {
+            o += WEBRTC_SPL_MUL_16_16(*b_ptr++, *x_ptr--);
+        }
+
+        // If output is higher than 32768, saturate it. Same with negative side
+        // 2^27 = 134217728, which corresponds to 32768 in Q12
+
+        // Saturate the output
+        o = WEBRTC_SPL_SAT((WebRtc_Word32)134215679, o, (WebRtc_Word32)-134217728);
+
+        *out_ptr++ = (WebRtc_Word16)((o + (WebRtc_Word32)2048) >> 12);
+    }
+    return;
+}
diff --git a/trunk/src/common_audio/signal_processing/get_hanning_window.c b/trunk/src/common_audio/signal_processing/get_hanning_window.c
new file mode 100644
index 0000000..6d67e60
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/get_hanning_window.c
@@ -0,0 +1,77 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains the function WebRtcSpl_GetHanningWindow().
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "signal_processing_library.h"
+
+// Hanning table with 256 entries
+static const WebRtc_Word16 kHanningTable[] = {
+    1,      2,      6,     10,     15,     22,     30,     39,
+   50,     62,     75,     89,    104,    121,    138,    157,
+  178,    199,    222,    246,    271,    297,    324,    353,
+  383,    413,    446,    479,    513,    549,    586,    624,
+  663,    703,    744,    787,    830,    875,    920,    967,
+ 1015,   1064,   1114,   1165,   1218,   1271,   1325,   1381,
+ 1437,   1494,   1553,   1612,   1673,   1734,   1796,   1859,
+ 1924,   1989,   2055,   2122,   2190,   2259,   2329,   2399,
+ 2471,   2543,   2617,   2691,   2765,   2841,   2918,   2995,
+ 3073,   3152,   3232,   3312,   3393,   3475,   3558,   3641,
+ 3725,   3809,   3895,   3980,   4067,   4154,   4242,   4330,
+ 4419,   4509,   4599,   4689,   4781,   4872,   4964,   5057,
+ 5150,   5244,   5338,   5432,   5527,   5622,   5718,   5814,
+ 5910,   6007,   6104,   6202,   6299,   6397,   6495,   6594,
+ 6693,   6791,   6891,   6990,   7090,   7189,   7289,   7389,
+ 7489,   7589,   7690,   7790,   7890,   7991,   8091,   8192,
+ 8293,   8393,   8494,   8594,   8694,   8795,   8895,   8995,
+ 9095,   9195,   9294,   9394,   9493,   9593,   9691,   9790,
+ 9889,   9987,  10085,  10182,  10280,  10377,  10474,  10570,
+10666,  10762,  10857,  10952,  11046,  11140,  11234,  11327,
+11420,  11512,  11603,  11695,  11785,  11875,  11965,  12054,
+12142,  12230,  12317,  12404,  12489,  12575,  12659,  12743,
+12826,  12909,  12991,  13072,  13152,  13232,  13311,  13389,
+13466,  13543,  13619,  13693,  13767,  13841,  13913,  13985,
+14055,  14125,  14194,  14262,  14329,  14395,  14460,  14525,
+14588,  14650,  14711,  14772,  14831,  14890,  14947,  15003,
+15059,  15113,  15166,  15219,  15270,  15320,  15369,  15417,
+15464,  15509,  15554,  15597,  15640,  15681,  15721,  15760,
+15798,  15835,  15871,  15905,  15938,  15971,  16001,  16031,
+16060,  16087,  16113,  16138,  16162,  16185,  16206,  16227,
+16246,  16263,  16280,  16295,  16309,  16322,  16334,  16345,
+16354,  16362,  16369,  16374,  16378,  16382,  16383,  16384
+};
+
+void WebRtcSpl_GetHanningWindow(WebRtc_Word16 *v, WebRtc_Word16 size)
+{
+    int jj;
+    WebRtc_Word16 *vptr1;
+
+    WebRtc_Word32 index;
+    WebRtc_Word32 factor = ((WebRtc_Word32)0x40000000);
+
+    factor = WebRtcSpl_DivW32W16(factor, size);
+    if (size < 513)
+        index = (WebRtc_Word32)-0x200000;
+    else
+        index = (WebRtc_Word32)-0x100000;
+    vptr1 = v;
+
+    for (jj = 0; jj < size; jj++)
+    {
+        index += factor;
+        (*vptr1++) = kHanningTable[index >> 22];
+    }
+
+}
diff --git a/trunk/src/common_audio/signal_processing/get_scaling_square.c b/trunk/src/common_audio/signal_processing/get_scaling_square.c
new file mode 100644
index 0000000..dccbf33
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/get_scaling_square.c
@@ -0,0 +1,44 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains the function WebRtcSpl_GetScalingSquare().
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "signal_processing_library.h"
+
+int WebRtcSpl_GetScalingSquare(WebRtc_Word16 *in_vector, int in_vector_length, int times)
+{
+    int nbits = WebRtcSpl_GetSizeInBits(times);
+    int i;
+    WebRtc_Word16 smax = -1;
+    WebRtc_Word16 sabs;
+    WebRtc_Word16 *sptr = in_vector;
+    int t;
+    int looptimes = in_vector_length;
+
+    for (i = looptimes; i > 0; i--)
+    {
+        sabs = (*sptr > 0 ? *sptr++ : -*sptr++);
+        smax = (sabs > smax ? sabs : smax);
+    }
+    t = WebRtcSpl_NormW32(WEBRTC_SPL_MUL(smax, smax));
+
+    if (smax == 0)
+    {
+        return 0; // Since norm(0) returns 0
+    } else
+    {
+        return (t > nbits) ? 0 : nbits - t;
+    }
+}
diff --git a/trunk/src/common_audio/signal_processing/ilbc_specific_functions.c b/trunk/src/common_audio/signal_processing/ilbc_specific_functions.c
new file mode 100644
index 0000000..5a9e577
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/ilbc_specific_functions.c
@@ -0,0 +1,120 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains implementations of the iLBC specific functions
+ * WebRtcSpl_ScaleAndAddVectorsWithRound()
+ * WebRtcSpl_ReverseOrderMultArrayElements()
+ * WebRtcSpl_ElementwiseVectorMult()
+ * WebRtcSpl_AddVectorsAndShift()
+ * WebRtcSpl_AddAffineVectorToVector()
+ * WebRtcSpl_AffineTransformVector()
+ *
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "signal_processing_library.h"
+
+void WebRtcSpl_ScaleAndAddVectorsWithRound(WebRtc_Word16 *vector1, WebRtc_Word16 scale1,
+                                           WebRtc_Word16 *vector2, WebRtc_Word16 scale2,
+                                           WebRtc_Word16 right_shifts, WebRtc_Word16 *out,
+                                           WebRtc_Word16 vector_length)
+{
+    int i;
+    WebRtc_Word16 roundVal;
+    roundVal = 1 << right_shifts;
+    roundVal = roundVal >> 1;
+    for (i = 0; i < vector_length; i++)
+    {
+        out[i] = (WebRtc_Word16)((WEBRTC_SPL_MUL_16_16(vector1[i], scale1)
+                + WEBRTC_SPL_MUL_16_16(vector2[i], scale2) + roundVal) >> right_shifts);
+    }
+}
+
+void WebRtcSpl_ReverseOrderMultArrayElements(WebRtc_Word16 *out, G_CONST WebRtc_Word16 *in,
+                                             G_CONST WebRtc_Word16 *win,
+                                             WebRtc_Word16 vector_length,
+                                             WebRtc_Word16 right_shifts)
+{
+    int i;
+    WebRtc_Word16 *outptr = out;
+    G_CONST WebRtc_Word16 *inptr = in;
+    G_CONST WebRtc_Word16 *winptr = win;
+    for (i = 0; i < vector_length; i++)
+    {
+        (*outptr++) = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(*inptr++,
+                                                               *winptr--, right_shifts);
+    }
+}
+
+void WebRtcSpl_ElementwiseVectorMult(WebRtc_Word16 *out, G_CONST WebRtc_Word16 *in,
+                                     G_CONST WebRtc_Word16 *win, WebRtc_Word16 vector_length,
+                                     WebRtc_Word16 right_shifts)
+{
+    int i;
+    WebRtc_Word16 *outptr = out;
+    G_CONST WebRtc_Word16 *inptr = in;
+    G_CONST WebRtc_Word16 *winptr = win;
+    for (i = 0; i < vector_length; i++)
+    {
+        (*outptr++) = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(*inptr++,
+                                                               *winptr++, right_shifts);
+    }
+}
+
+void WebRtcSpl_AddVectorsAndShift(WebRtc_Word16 *out, G_CONST WebRtc_Word16 *in1,
+                                  G_CONST WebRtc_Word16 *in2, WebRtc_Word16 vector_length,
+                                  WebRtc_Word16 right_shifts)
+{
+    int i;
+    WebRtc_Word16 *outptr = out;
+    G_CONST WebRtc_Word16 *in1ptr = in1;
+    G_CONST WebRtc_Word16 *in2ptr = in2;
+    for (i = vector_length; i > 0; i--)
+    {
+        (*outptr++) = (WebRtc_Word16)(((*in1ptr++) + (*in2ptr++)) >> right_shifts);
+    }
+}
+
+void WebRtcSpl_AddAffineVectorToVector(WebRtc_Word16 *out, WebRtc_Word16 *in,
+                                       WebRtc_Word16 gain, WebRtc_Word32 add_constant,
+                                       WebRtc_Word16 right_shifts, int vector_length)
+{
+    WebRtc_Word16 *inPtr;
+    WebRtc_Word16 *outPtr;
+    int i;
+
+    inPtr = in;
+    outPtr = out;
+    for (i = 0; i < vector_length; i++)
+    {
+        (*outPtr++) += (WebRtc_Word16)((WEBRTC_SPL_MUL_16_16((*inPtr++), gain)
+                + (WebRtc_Word32)add_constant) >> right_shifts);
+    }
+}
+
+void WebRtcSpl_AffineTransformVector(WebRtc_Word16 *out, WebRtc_Word16 *in,
+                                     WebRtc_Word16 gain, WebRtc_Word32 add_constant,
+                                     WebRtc_Word16 right_shifts, int vector_length)
+{
+    WebRtc_Word16 *inPtr;
+    WebRtc_Word16 *outPtr;
+    int i;
+
+    inPtr = in;
+    outPtr = out;
+    for (i = 0; i < vector_length; i++)
+    {
+        (*outPtr++) = (WebRtc_Word16)((WEBRTC_SPL_MUL_16_16((*inPtr++), gain)
+                + (WebRtc_Word32)add_constant) >> right_shifts);
+    }
+}
diff --git a/trunk/src/common_audio/signal_processing/include/signal_processing_library.h b/trunk/src/common_audio/signal_processing/include/signal_processing_library.h
new file mode 100644
index 0000000..87b8f29
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/include/signal_processing_library.h
@@ -0,0 +1,1667 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This header file includes all of the fix point signal processing library (SPL) function
+ * descriptions and declarations.
+ * For specific function calls, see bottom of file.
+ */
+
+#ifndef WEBRTC_SPL_SIGNAL_PROCESSING_LIBRARY_H_
+#define WEBRTC_SPL_SIGNAL_PROCESSING_LIBRARY_H_
+
+#include <string.h>
+#include "typedefs.h"
+
+#ifdef ARM_WINM
+#include <Armintr.h> // intrinsic file for windows mobile
+#endif
+
+// Macros specific for the fixed point implementation
+#define WEBRTC_SPL_WORD16_MAX       32767
+#define WEBRTC_SPL_WORD16_MIN       -32768
+#define WEBRTC_SPL_WORD32_MAX       (WebRtc_Word32)0x7fffffff
+#define WEBRTC_SPL_WORD32_MIN       (WebRtc_Word32)0x80000000
+#define WEBRTC_SPL_MAX_LPC_ORDER    14
+#define WEBRTC_SPL_MAX_SEED_USED    0x80000000L
+#define WEBRTC_SPL_MIN(A, B)        (A < B ? A : B) // Get min value
+#define WEBRTC_SPL_MAX(A, B)        (A > B ? A : B) // Get max value
+#define WEBRTC_SPL_ABS_W16(a) \
+    (((WebRtc_Word16)a >= 0) ? ((WebRtc_Word16)a) : -((WebRtc_Word16)a))
+#define WEBRTC_SPL_ABS_W32(a) \
+    (((WebRtc_Word32)a >= 0) ? ((WebRtc_Word32)a) : -((WebRtc_Word32)a))
+
+#if (defined WEBRTC_TARGET_PC)||(defined __TARGET_XSCALE)
+#define WEBRTC_SPL_GET_BYTE(a, nr)  (((WebRtc_Word8 *)a)[nr])
+#define WEBRTC_SPL_SET_BYTE(d_ptr, val, index) \
+    (((WebRtc_Word8 *)d_ptr)[index] = (val))
+#elif defined WEBRTC_BIG_ENDIAN
+#define WEBRTC_SPL_GET_BYTE(a, nr) \
+    ((((WebRtc_Word16 *)a)[nr >> 1]) >> (((nr + 1) & 0x1) * 8) & 0x00ff)
+#define WEBRTC_SPL_SET_BYTE(d_ptr, val, index) \
+    ((WebRtc_Word16 *)d_ptr)[index >> 1] = \
+    ((((WebRtc_Word16 *)d_ptr)[index >> 1]) \
+    & (0x00ff << (8 * ((index) & 0x1)))) | (val << (8 * ((index + 1) & 0x1)))
+#else
+#define WEBRTC_SPL_GET_BYTE(a,nr) \
+    ((((WebRtc_Word16 *)(a))[(nr) >> 1]) >> (((nr) & 0x1) * 8) & 0x00ff)
+#define WEBRTC_SPL_SET_BYTE(d_ptr, val, index) \
+    ((WebRtc_Word16 *)(d_ptr))[(index) >> 1] = \
+    ((((WebRtc_Word16 *)(d_ptr))[(index) >> 1]) \
+    & (0x00ff << (8 * (((index) + 1) & 0x1)))) | \
+    ((val) << (8 * ((index) & 0x1)))
+#endif
+
+#define WEBRTC_SPL_MUL(a, b) \
+    ((WebRtc_Word32) ((WebRtc_Word32)(a) * (WebRtc_Word32)(b)))
+#define WEBRTC_SPL_UMUL(a, b) \
+    ((WebRtc_UWord32) ((WebRtc_UWord32)(a) * (WebRtc_UWord32)(b)))
+#define WEBRTC_SPL_UMUL_RSFT16(a, b) \
+    ((WebRtc_UWord32) ((WebRtc_UWord32)(a) * (WebRtc_UWord32)(b)) >> 16)
+#define WEBRTC_SPL_UMUL_16_16(a, b) \
+    ((WebRtc_UWord32) (WebRtc_UWord16)(a) * (WebRtc_UWord16)(b))
+#define WEBRTC_SPL_UMUL_16_16_RSFT16(a, b) \
+    (((WebRtc_UWord32) (WebRtc_UWord16)(a) * (WebRtc_UWord16)(b)) >> 16)
+#define WEBRTC_SPL_UMUL_32_16(a, b) \
+    ((WebRtc_UWord32) ((WebRtc_UWord32)(a) * (WebRtc_UWord16)(b)))
+#define WEBRTC_SPL_UMUL_32_16_RSFT16(a, b) \
+    ((WebRtc_UWord32) ((WebRtc_UWord32)(a) * (WebRtc_UWord16)(b)) >> 16)
+#define WEBRTC_SPL_MUL_16_U16(a, b) \
+    ((WebRtc_Word32)(WebRtc_Word16)(a) * (WebRtc_UWord16)(b))
+#define WEBRTC_SPL_DIV(a, b) \
+    ((WebRtc_Word32) ((WebRtc_Word32)(a) / (WebRtc_Word32)(b)))
+#define WEBRTC_SPL_UDIV(a, b) \
+    ((WebRtc_UWord32) ((WebRtc_UWord32)(a) / (WebRtc_UWord32)(b)))
+
+#ifndef WEBRTC_ARCH_ARM_V7A
+// For ARMv7 platforms, these are inline functions in spl_inl_armv7.h
+#define WEBRTC_SPL_MUL_16_16(a, b) \
+    ((WebRtc_Word32) (((WebRtc_Word16)(a)) * ((WebRtc_Word16)(b))))
+#define WEBRTC_SPL_MUL_16_32_RSFT16(a, b) \
+    (WEBRTC_SPL_MUL_16_16(a, b >> 16) \
+     + ((WEBRTC_SPL_MUL_16_16(a, (b & 0xffff) >> 1) + 0x4000) >> 15))
+#define WEBRTC_SPL_MUL_32_32_RSFT32(a32a, a32b, b32) \
+    ((WebRtc_Word32)(WEBRTC_SPL_MUL_16_32_RSFT16(a32a, b32) \
+    + (WEBRTC_SPL_MUL_16_32_RSFT16(a32b, b32) >> 16)))
+#define WEBRTC_SPL_MUL_32_32_RSFT32BI(a32, b32) \
+    ((WebRtc_Word32)(WEBRTC_SPL_MUL_16_32_RSFT16(( \
+    (WebRtc_Word16)(a32 >> 16)), b32) + \
+    (WEBRTC_SPL_MUL_16_32_RSFT16(( \
+    (WebRtc_Word16)((a32 & 0x0000FFFF) >> 1)), b32) >> 15)))
+#endif
+
+#define WEBRTC_SPL_MUL_16_32_RSFT11(a, b) \
+    ((WEBRTC_SPL_MUL_16_16(a, (b) >> 16) << 5) \
+    + (((WEBRTC_SPL_MUL_16_U16(a, (WebRtc_UWord16)(b)) >> 1) + 0x0200) >> 10))
+#define WEBRTC_SPL_MUL_16_32_RSFT14(a, b) \
+    ((WEBRTC_SPL_MUL_16_16(a, (b) >> 16) << 2) \
+    + (((WEBRTC_SPL_MUL_16_U16(a, (WebRtc_UWord16)(b)) >> 1) + 0x1000) >> 13))
+#define WEBRTC_SPL_MUL_16_32_RSFT15(a, b) \
+    ((WEBRTC_SPL_MUL_16_16(a, (b) >> 16) << 1) \
+    + (((WEBRTC_SPL_MUL_16_U16(a, (WebRtc_UWord16)(b)) >> 1) + 0x2000) >> 14))
+
+#ifdef ARM_WINM
+#define WEBRTC_SPL_MUL_16_16(a, b) \
+    _SmulLo_SW_SL((WebRtc_Word16)(a), (WebRtc_Word16)(b))
+#endif
+
+#define WEBRTC_SPL_MUL_16_16_RSFT(a, b, c) \
+    (WEBRTC_SPL_MUL_16_16(a, b) >> (c))
+
+#define WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(a, b, c) \
+    ((WEBRTC_SPL_MUL_16_16(a, b) + ((WebRtc_Word32) \
+                                  (((WebRtc_Word32)1) << ((c) - 1)))) >> (c))
+#define WEBRTC_SPL_MUL_16_16_RSFT_WITH_FIXROUND(a, b) \
+    ((WEBRTC_SPL_MUL_16_16(a, b) + ((WebRtc_Word32) (1 << 14))) >> 15)
+
+// C + the 32 most significant bits of A * B
+#define WEBRTC_SPL_SCALEDIFF32(A, B, C) \
+    (C + (B >> 16) * A + (((WebRtc_UWord32)(0x0000FFFF & B) * A) >> 16))
+
+#define WEBRTC_SPL_ADD_SAT_W32(a, b)    WebRtcSpl_AddSatW32(a, b)
+#define WEBRTC_SPL_SAT(a, b, c)         (b > a ? a : b < c ? c : b)
+#define WEBRTC_SPL_MUL_32_16(a, b)      ((a) * (b))
+
+#define WEBRTC_SPL_SUB_SAT_W32(a, b)    WebRtcSpl_SubSatW32(a, b)
+#define WEBRTC_SPL_ADD_SAT_W16(a, b)    WebRtcSpl_AddSatW16(a, b)
+#define WEBRTC_SPL_SUB_SAT_W16(a, b)    WebRtcSpl_SubSatW16(a, b)
+
+// We cannot do casting here due to signed/unsigned problem
+#define WEBRTC_SPL_IS_NEG(a)            ((a) & 0x80000000)
+// Shifting with negative numbers allowed
+// Positive means left shift
+#define WEBRTC_SPL_SHIFT_W16(x, c) \
+    (((c) >= 0) ? ((x) << (c)) : ((x) >> (-(c))))
+#define WEBRTC_SPL_SHIFT_W32(x, c) \
+    (((c) >= 0) ? ((x) << (c)) : ((x) >> (-(c))))
+
+// Shifting with negative numbers not allowed
+// We cannot do casting here due to signed/unsigned problem
+#define WEBRTC_SPL_RSHIFT_W16(x, c)     ((x) >> (c))
+#define WEBRTC_SPL_LSHIFT_W16(x, c)     ((x) << (c))
+#define WEBRTC_SPL_RSHIFT_W32(x, c)     ((x) >> (c))
+#define WEBRTC_SPL_LSHIFT_W32(x, c)     ((x) << (c))
+
+#define WEBRTC_SPL_RSHIFT_U16(x, c)     ((WebRtc_UWord16)(x) >> (c))
+#define WEBRTC_SPL_LSHIFT_U16(x, c)     ((WebRtc_UWord16)(x) << (c))
+#define WEBRTC_SPL_RSHIFT_U32(x, c)     ((WebRtc_UWord32)(x) >> (c))
+#define WEBRTC_SPL_LSHIFT_U32(x, c)     ((WebRtc_UWord32)(x) << (c))
+
+#define WEBRTC_SPL_VNEW(t, n)           (t *) malloc (sizeof (t) * (n))
+#define WEBRTC_SPL_FREE                 free
+
+#define WEBRTC_SPL_RAND(a) \
+    ((WebRtc_Word16)(WEBRTC_SPL_MUL_16_16_RSFT((a), 18816, 7) & 0x00007fff))
+
+#ifdef __cplusplus
+extern "C"
+{
+#endif
+
+#define WEBRTC_SPL_MEMCPY_W8(v1, v2, length) \
+   memcpy(v1, v2, (length) * sizeof(char))
+#define WEBRTC_SPL_MEMCPY_W16(v1, v2, length) \
+   memcpy(v1, v2, (length) * sizeof(WebRtc_Word16))
+
+#define WEBRTC_SPL_MEMMOVE_W16(v1, v2, length) \
+   memmove(v1, v2, (length) * sizeof(WebRtc_Word16))
+
+// inline functions:
+#include "spl_inl.h"
+
+// Get SPL Version
+WebRtc_Word16 WebRtcSpl_get_version(char* version,
+                                    WebRtc_Word16 length_in_bytes);
+
+int WebRtcSpl_GetScalingSquare(WebRtc_Word16* in_vector,
+                               int in_vector_length,
+                               int times);
+
+// Copy and set operations. Implementation in copy_set_operations.c.
+// Descriptions at bottom of file.
+void WebRtcSpl_MemSetW16(WebRtc_Word16* vector,
+                         WebRtc_Word16 set_value,
+                         int vector_length);
+void WebRtcSpl_MemSetW32(WebRtc_Word32* vector,
+                         WebRtc_Word32 set_value,
+                         int vector_length);
+void WebRtcSpl_MemCpyReversedOrder(WebRtc_Word16* out_vector,
+                                   WebRtc_Word16* in_vector,
+                                   int vector_length);
+WebRtc_Word16 WebRtcSpl_CopyFromEndW16(G_CONST WebRtc_Word16* in_vector,
+                                       WebRtc_Word16 in_vector_length,
+                                       WebRtc_Word16 samples,
+                                       WebRtc_Word16* out_vector);
+WebRtc_Word16 WebRtcSpl_ZerosArrayW16(WebRtc_Word16* vector,
+                                      WebRtc_Word16 vector_length);
+WebRtc_Word16 WebRtcSpl_ZerosArrayW32(WebRtc_Word32* vector,
+                                      WebRtc_Word16 vector_length);
+WebRtc_Word16 WebRtcSpl_OnesArrayW16(WebRtc_Word16* vector,
+                                     WebRtc_Word16 vector_length);
+WebRtc_Word16 WebRtcSpl_OnesArrayW32(WebRtc_Word32* vector,
+                                     WebRtc_Word16 vector_length);
+// End: Copy and set operations.
+
+// Minimum and maximum operations. Implementation in min_max_operations.c.
+// Descriptions at bottom of file.
+WebRtc_Word16 WebRtcSpl_MaxAbsValueW16(const WebRtc_Word16* vector,
+                                       WebRtc_Word16 length);
+WebRtc_Word32 WebRtcSpl_MaxAbsValueW32(G_CONST WebRtc_Word32* vector,
+                                       WebRtc_Word16 length);
+WebRtc_Word16 WebRtcSpl_MinValueW16(G_CONST WebRtc_Word16* vector,
+                                    WebRtc_Word16 length);
+WebRtc_Word32 WebRtcSpl_MinValueW32(G_CONST WebRtc_Word32* vector,
+                                    WebRtc_Word16 length);
+WebRtc_Word16 WebRtcSpl_MaxValueW16(G_CONST WebRtc_Word16* vector,
+                                    WebRtc_Word16 length);
+
+WebRtc_Word16 WebRtcSpl_MaxAbsIndexW16(G_CONST WebRtc_Word16* vector,
+                                       WebRtc_Word16 length);
+WebRtc_Word32 WebRtcSpl_MaxValueW32(G_CONST WebRtc_Word32* vector,
+                                    WebRtc_Word16 length);
+WebRtc_Word16 WebRtcSpl_MinIndexW16(G_CONST WebRtc_Word16* vector,
+                                    WebRtc_Word16 length);
+WebRtc_Word16 WebRtcSpl_MinIndexW32(G_CONST WebRtc_Word32* vector,
+                                    WebRtc_Word16 length);
+WebRtc_Word16 WebRtcSpl_MaxIndexW16(G_CONST WebRtc_Word16* vector,
+                                    WebRtc_Word16 length);
+WebRtc_Word16 WebRtcSpl_MaxIndexW32(G_CONST WebRtc_Word32* vector,
+                                    WebRtc_Word16 length);
+// End: Minimum and maximum operations.
+
+// Vector scaling operations. Implementation in vector_scaling_operations.c.
+// Description at bottom of file.
+void WebRtcSpl_VectorBitShiftW16(WebRtc_Word16* out_vector,
+                                 WebRtc_Word16 vector_length,
+                                 G_CONST WebRtc_Word16* in_vector,
+                                 WebRtc_Word16 right_shifts);
+void WebRtcSpl_VectorBitShiftW32(WebRtc_Word32* out_vector,
+                                 WebRtc_Word16 vector_length,
+                                 G_CONST WebRtc_Word32* in_vector,
+                                 WebRtc_Word16 right_shifts);
+void WebRtcSpl_VectorBitShiftW32ToW16(WebRtc_Word16* out_vector,
+                                      WebRtc_Word16 vector_length,
+                                      G_CONST WebRtc_Word32* in_vector,
+                                      WebRtc_Word16 right_shifts);
+
+void WebRtcSpl_ScaleVector(G_CONST WebRtc_Word16* in_vector,
+                           WebRtc_Word16* out_vector,
+                           WebRtc_Word16 gain,
+                           WebRtc_Word16 vector_length,
+                           WebRtc_Word16 right_shifts);
+void WebRtcSpl_ScaleVectorWithSat(G_CONST WebRtc_Word16* in_vector,
+                                  WebRtc_Word16* out_vector,
+                                  WebRtc_Word16 gain,
+                                  WebRtc_Word16 vector_length,
+                                  WebRtc_Word16 right_shifts);
+void WebRtcSpl_ScaleAndAddVectors(G_CONST WebRtc_Word16* in_vector1,
+                                  WebRtc_Word16 gain1, int right_shifts1,
+                                  G_CONST WebRtc_Word16* in_vector2,
+                                  WebRtc_Word16 gain2, int right_shifts2,
+                                  WebRtc_Word16* out_vector,
+                                  int vector_length);
+// End: Vector scaling operations.
+
+// iLBC specific functions. Implementations in ilbc_specific_functions.c.
+// Description at bottom of file.
+void WebRtcSpl_ScaleAndAddVectorsWithRound(WebRtc_Word16* in_vector1,
+                                           WebRtc_Word16 scale1,
+                                           WebRtc_Word16* in_vector2,
+                                           WebRtc_Word16 scale2,
+                                           WebRtc_Word16 right_shifts,
+                                           WebRtc_Word16* out_vector,
+                                           WebRtc_Word16 vector_length);
+void WebRtcSpl_ReverseOrderMultArrayElements(WebRtc_Word16* out_vector,
+                                             G_CONST WebRtc_Word16* in_vector,
+                                             G_CONST WebRtc_Word16* window,
+                                             WebRtc_Word16 vector_length,
+                                             WebRtc_Word16 right_shifts);
+void WebRtcSpl_ElementwiseVectorMult(WebRtc_Word16* out_vector,
+                                     G_CONST WebRtc_Word16* in_vector,
+                                     G_CONST WebRtc_Word16* window,
+                                     WebRtc_Word16 vector_length,
+                                     WebRtc_Word16 right_shifts);
+void WebRtcSpl_AddVectorsAndShift(WebRtc_Word16* out_vector,
+                                  G_CONST WebRtc_Word16* in_vector1,
+                                  G_CONST WebRtc_Word16* in_vector2,
+                                  WebRtc_Word16 vector_length,
+                                  WebRtc_Word16 right_shifts);
+void WebRtcSpl_AddAffineVectorToVector(WebRtc_Word16* out_vector,
+                                       WebRtc_Word16* in_vector,
+                                       WebRtc_Word16 gain,
+                                       WebRtc_Word32 add_constant,
+                                       WebRtc_Word16 right_shifts,
+                                       int vector_length);
+void WebRtcSpl_AffineTransformVector(WebRtc_Word16* out_vector,
+                                     WebRtc_Word16* in_vector,
+                                     WebRtc_Word16 gain,
+                                     WebRtc_Word32 add_constant,
+                                     WebRtc_Word16 right_shifts,
+                                     int vector_length);
+// End: iLBC specific functions.
+
+// Signal processing operations. Descriptions at bottom of this file.
+int WebRtcSpl_AutoCorrelation(G_CONST WebRtc_Word16* vector,
+                              int vector_length, int order,
+                              WebRtc_Word32* result_vector,
+                              int* scale);
+WebRtc_Word16 WebRtcSpl_LevinsonDurbin(WebRtc_Word32* auto_corr,
+                                       WebRtc_Word16* lpc_coef,
+                                       WebRtc_Word16* refl_coef,
+                                       WebRtc_Word16 order);
+void WebRtcSpl_ReflCoefToLpc(G_CONST WebRtc_Word16* refl_coef,
+                             int use_order,
+                             WebRtc_Word16* lpc_coef);
+void WebRtcSpl_LpcToReflCoef(WebRtc_Word16* lpc_coef,
+                             int use_order,
+                             WebRtc_Word16* refl_coef);
+void WebRtcSpl_AutoCorrToReflCoef(G_CONST WebRtc_Word32* auto_corr,
+                                  int use_order,
+                                  WebRtc_Word16* refl_coef);
+void WebRtcSpl_CrossCorrelation(WebRtc_Word32* cross_corr,
+                                WebRtc_Word16* vector1,
+                                WebRtc_Word16* vector2,
+                                WebRtc_Word16 dim_vector,
+                                WebRtc_Word16 dim_cross_corr,
+                                WebRtc_Word16 right_shifts,
+                                WebRtc_Word16 step_vector2);
+void WebRtcSpl_GetHanningWindow(WebRtc_Word16* window, WebRtc_Word16 size);
+void WebRtcSpl_SqrtOfOneMinusXSquared(WebRtc_Word16* in_vector,
+                                      int vector_length,
+                                      WebRtc_Word16* out_vector);
+// End: Signal processing operations.
+
+// Randomization functions. Implementations collected in randomization_functions.c and
+// descriptions at bottom of this file.
+WebRtc_UWord32 WebRtcSpl_IncreaseSeed(WebRtc_UWord32* seed);
+WebRtc_Word16 WebRtcSpl_RandU(WebRtc_UWord32* seed);
+WebRtc_Word16 WebRtcSpl_RandN(WebRtc_UWord32* seed);
+WebRtc_Word16 WebRtcSpl_RandUArray(WebRtc_Word16* vector,
+                                   WebRtc_Word16 vector_length,
+                                   WebRtc_UWord32* seed);
+// End: Randomization functions.
+
+// Math functions
+WebRtc_Word32 WebRtcSpl_Sqrt(WebRtc_Word32 value);
+WebRtc_Word32 WebRtcSpl_SqrtFloor(WebRtc_Word32 value);
+
+// Divisions. Implementations collected in division_operations.c and
+// descriptions at bottom of this file.
+WebRtc_UWord32 WebRtcSpl_DivU32U16(WebRtc_UWord32 num, WebRtc_UWord16 den);
+WebRtc_Word32 WebRtcSpl_DivW32W16(WebRtc_Word32 num, WebRtc_Word16 den);
+WebRtc_Word16 WebRtcSpl_DivW32W16ResW16(WebRtc_Word32 num, WebRtc_Word16 den);
+WebRtc_Word32 WebRtcSpl_DivResultInQ31(WebRtc_Word32 num, WebRtc_Word32 den);
+WebRtc_Word32 WebRtcSpl_DivW32HiLow(WebRtc_Word32 num, WebRtc_Word16 den_hi,
+                                    WebRtc_Word16 den_low);
+// End: Divisions.
+
+WebRtc_Word32 WebRtcSpl_Energy(WebRtc_Word16* vector,
+                               int vector_length,
+                               int* scale_factor);
+
+WebRtc_Word32 WebRtcSpl_DotProductWithScale(WebRtc_Word16* vector1,
+                                            WebRtc_Word16* vector2,
+                                            int vector_length,
+                                            int scaling);
+
+// Filter operations.
+int WebRtcSpl_FilterAR(G_CONST WebRtc_Word16* ar_coef, int ar_coef_length,
+                       G_CONST WebRtc_Word16* in_vector, int in_vector_length,
+                       WebRtc_Word16* filter_state, int filter_state_length,
+                       WebRtc_Word16* filter_state_low,
+                       int filter_state_low_length, WebRtc_Word16* out_vector,
+                       WebRtc_Word16* out_vector_low, int out_vector_low_length);
+
+void WebRtcSpl_FilterMAFastQ12(WebRtc_Word16* in_vector,
+                               WebRtc_Word16* out_vector,
+                               WebRtc_Word16* ma_coef,
+                               WebRtc_Word16 ma_coef_length,
+                               WebRtc_Word16 vector_length);
+
+// Performs a AR filtering on a vector in Q12
+// Input:
+//      - data_in            : Input samples
+//      - data_out           : State information in positions
+//                               data_out[-order] .. data_out[-1]
+//      - coefficients       : Filter coefficients (in Q12)
+//      - coefficients_length: Number of coefficients (order+1)
+//      - data_length        : Number of samples to be filtered
+// Output:
+//      - data_out           : Filtered samples
+void WebRtcSpl_FilterARFastQ12(const int16_t* data_in,
+                               int16_t* data_out,
+                               const int16_t* __restrict coefficients,
+                               int coefficients_length,
+                               int data_length);
+
+// Performs a MA down sampling filter on a vector
+// Input:
+//      - data_in            : Input samples (state in positions
+//                               data_in[-order] .. data_in[-1])
+//      - data_in_length     : Number of samples in |data_in| to be filtered.
+//                               This must be at least
+//                               |delay| + |factor|*(|out_vector_length|-1) + 1)
+//      - data_out_length    : Number of down sampled samples desired
+//      - coefficients       : Filter coefficients (in Q12)
+//      - coefficients_length: Number of coefficients (order+1)
+//      - factor             : Decimation factor
+//      - delay              : Delay of filter (compensated for in out_vector)
+// Output:
+//      - data_out           : Filtered samples
+// Return value              : 0 if OK, -1 if |in_vector| is too short
+int WebRtcSpl_DownsampleFast(const int16_t* data_in,
+                             int data_in_length,
+                             int16_t* data_out,
+                             int data_out_length,
+                             const int16_t* __restrict coefficients,
+                             int coefficients_length,
+                             int factor,
+                             int delay);
+
+// End: Filter operations.
+
+// FFT operations
+
+int WebRtcSpl_ComplexFFT(WebRtc_Word16 vector[], int stages, int mode);
+int WebRtcSpl_ComplexIFFT(WebRtc_Word16 vector[], int stages, int mode);
+
+// Treat a 16-bit complex data buffer |complex_data| as an array of 32-bit
+// values, and swap elements whose indexes are bit-reverses of each other.
+//
+// Input:
+//      - complex_data  : Complex data buffer containing 2^|stages| real
+//                        elements interleaved with 2^|stages| imaginary
+//                        elements: [Re Im Re Im Re Im....]
+//      - stages        : Number of FFT stages. Must be at least 3 and at most
+//                        10, since the table WebRtcSpl_kSinTable1024[] is 1024
+//                        elements long.
+//
+// Output:
+//      - complex_data  : The complex data buffer.
+
+void WebRtcSpl_ComplexBitReverse(int16_t* __restrict complex_data, int stages);
+
+// End: FFT operations
+
+/************************************************************
+ *
+ * RESAMPLING FUNCTIONS AND THEIR STRUCTS ARE DEFINED BELOW
+ *
+ ************************************************************/
+
+/*******************************************************************
+ * resample.c
+ *
+ * Includes the following resampling combinations
+ * 22 kHz -> 16 kHz
+ * 16 kHz -> 22 kHz
+ * 22 kHz ->  8 kHz
+ *  8 kHz -> 22 kHz
+ *
+ ******************************************************************/
+
+// state structure for 22 -> 16 resampler
+typedef struct
+{
+    WebRtc_Word32 S_22_44[8];
+    WebRtc_Word32 S_44_32[8];
+    WebRtc_Word32 S_32_16[8];
+} WebRtcSpl_State22khzTo16khz;
+
+void WebRtcSpl_Resample22khzTo16khz(const WebRtc_Word16* in,
+                                    WebRtc_Word16* out,
+                                    WebRtcSpl_State22khzTo16khz* state,
+                                    WebRtc_Word32* tmpmem);
+
+void WebRtcSpl_ResetResample22khzTo16khz(WebRtcSpl_State22khzTo16khz* state);
+
+// state structure for 16 -> 22 resampler
+typedef struct
+{
+    WebRtc_Word32 S_16_32[8];
+    WebRtc_Word32 S_32_22[8];
+} WebRtcSpl_State16khzTo22khz;
+
+void WebRtcSpl_Resample16khzTo22khz(const WebRtc_Word16* in,
+                                    WebRtc_Word16* out,
+                                    WebRtcSpl_State16khzTo22khz* state,
+                                    WebRtc_Word32* tmpmem);
+
+void WebRtcSpl_ResetResample16khzTo22khz(WebRtcSpl_State16khzTo22khz* state);
+
+// state structure for 22 -> 8 resampler
+typedef struct
+{
+    WebRtc_Word32 S_22_22[16];
+    WebRtc_Word32 S_22_16[8];
+    WebRtc_Word32 S_16_8[8];
+} WebRtcSpl_State22khzTo8khz;
+
+void WebRtcSpl_Resample22khzTo8khz(const WebRtc_Word16* in, WebRtc_Word16* out,
+                                   WebRtcSpl_State22khzTo8khz* state,
+                                   WebRtc_Word32* tmpmem);
+
+void WebRtcSpl_ResetResample22khzTo8khz(WebRtcSpl_State22khzTo8khz* state);
+
+// state structure for 8 -> 22 resampler
+typedef struct
+{
+    WebRtc_Word32 S_8_16[8];
+    WebRtc_Word32 S_16_11[8];
+    WebRtc_Word32 S_11_22[8];
+} WebRtcSpl_State8khzTo22khz;
+
+void WebRtcSpl_Resample8khzTo22khz(const WebRtc_Word16* in, WebRtc_Word16* out,
+                                   WebRtcSpl_State8khzTo22khz* state,
+                                   WebRtc_Word32* tmpmem);
+
+void WebRtcSpl_ResetResample8khzTo22khz(WebRtcSpl_State8khzTo22khz* state);
+
+/*******************************************************************
+ * resample_fractional.c
+ * Functions for internal use in the other resample functions
+ *
+ * Includes the following resampling combinations
+ * 48 kHz -> 32 kHz
+ * 32 kHz -> 24 kHz
+ * 44 kHz -> 32 kHz
+ *
+ ******************************************************************/
+
+void WebRtcSpl_Resample48khzTo32khz(const WebRtc_Word32* In, WebRtc_Word32* Out,
+                                    const WebRtc_Word32 K);
+
+void WebRtcSpl_Resample32khzTo24khz(const WebRtc_Word32* In, WebRtc_Word32* Out,
+                                    const WebRtc_Word32 K);
+
+void WebRtcSpl_Resample44khzTo32khz(const WebRtc_Word32* In, WebRtc_Word32* Out,
+                                    const WebRtc_Word32 K);
+
+/*******************************************************************
+ * resample_48khz.c
+ *
+ * Includes the following resampling combinations
+ * 48 kHz -> 16 kHz
+ * 16 kHz -> 48 kHz
+ * 48 kHz ->  8 kHz
+ *  8 kHz -> 48 kHz
+ *
+ ******************************************************************/
+
+typedef struct
+{
+    WebRtc_Word32 S_48_48[16];
+    WebRtc_Word32 S_48_32[8];
+    WebRtc_Word32 S_32_16[8];
+} WebRtcSpl_State48khzTo16khz;
+
+void WebRtcSpl_Resample48khzTo16khz(const WebRtc_Word16* in, WebRtc_Word16* out,
+                                    WebRtcSpl_State48khzTo16khz* state,
+                                    WebRtc_Word32* tmpmem);
+
+void WebRtcSpl_ResetResample48khzTo16khz(WebRtcSpl_State48khzTo16khz* state);
+
+typedef struct
+{
+    WebRtc_Word32 S_16_32[8];
+    WebRtc_Word32 S_32_24[8];
+    WebRtc_Word32 S_24_48[8];
+} WebRtcSpl_State16khzTo48khz;
+
+void WebRtcSpl_Resample16khzTo48khz(const WebRtc_Word16* in, WebRtc_Word16* out,
+                                    WebRtcSpl_State16khzTo48khz* state,
+                                    WebRtc_Word32* tmpmem);
+
+void WebRtcSpl_ResetResample16khzTo48khz(WebRtcSpl_State16khzTo48khz* state);
+
+typedef struct
+{
+    WebRtc_Word32 S_48_24[8];
+    WebRtc_Word32 S_24_24[16];
+    WebRtc_Word32 S_24_16[8];
+    WebRtc_Word32 S_16_8[8];
+} WebRtcSpl_State48khzTo8khz;
+
+void WebRtcSpl_Resample48khzTo8khz(const WebRtc_Word16* in, WebRtc_Word16* out,
+                                   WebRtcSpl_State48khzTo8khz* state,
+                                   WebRtc_Word32* tmpmem);
+
+void WebRtcSpl_ResetResample48khzTo8khz(WebRtcSpl_State48khzTo8khz* state);
+
+typedef struct
+{
+    WebRtc_Word32 S_8_16[8];
+    WebRtc_Word32 S_16_12[8];
+    WebRtc_Word32 S_12_24[8];
+    WebRtc_Word32 S_24_48[8];
+} WebRtcSpl_State8khzTo48khz;
+
+void WebRtcSpl_Resample8khzTo48khz(const WebRtc_Word16* in, WebRtc_Word16* out,
+                                   WebRtcSpl_State8khzTo48khz* state,
+                                   WebRtc_Word32* tmpmem);
+
+void WebRtcSpl_ResetResample8khzTo48khz(WebRtcSpl_State8khzTo48khz* state);
+
+/*******************************************************************
+ * resample_by_2.c
+ *
+ * Includes down and up sampling by a factor of two.
+ *
+ ******************************************************************/
+
+void WebRtcSpl_DownsampleBy2(const WebRtc_Word16* in, const WebRtc_Word16 len,
+                             WebRtc_Word16* out, WebRtc_Word32* filtState);
+
+void WebRtcSpl_UpsampleBy2(const WebRtc_Word16* in, WebRtc_Word16 len, WebRtc_Word16* out,
+                           WebRtc_Word32* filtState);
+
+/************************************************************
+ * END OF RESAMPLING FUNCTIONS
+ ************************************************************/
+void WebRtcSpl_AnalysisQMF(const WebRtc_Word16* in_data,
+                           WebRtc_Word16* low_band,
+                           WebRtc_Word16* high_band,
+                           WebRtc_Word32* filter_state1,
+                           WebRtc_Word32* filter_state2);
+void WebRtcSpl_SynthesisQMF(const WebRtc_Word16* low_band,
+                            const WebRtc_Word16* high_band,
+                            WebRtc_Word16* out_data,
+                            WebRtc_Word32* filter_state1,
+                            WebRtc_Word32* filter_state2);
+
+#ifdef __cplusplus
+}
+#endif // __cplusplus
+#endif // WEBRTC_SPL_SIGNAL_PROCESSING_LIBRARY_H_
+
+//
+// WebRtcSpl_AddSatW16(...)
+// WebRtcSpl_AddSatW32(...)
+//
+// Returns the result of a saturated 16-bit, respectively 32-bit, addition of
+// the numbers specified by the |var1| and |var2| parameters.
+//
+// Input:
+//      - var1      : Input variable 1
+//      - var2      : Input variable 2
+//
+// Return value     : Added and saturated value
+//
+
+//
+// WebRtcSpl_SubSatW16(...)
+// WebRtcSpl_SubSatW32(...)
+//
+// Returns the result of a saturated 16-bit, respectively 32-bit, subtraction
+// of the numbers specified by the |var1| and |var2| parameters.
+//
+// Input:
+//      - var1      : Input variable 1
+//      - var2      : Input variable 2
+//
+// Returned value   : Subtracted and saturated value
+//
+
+//
+// WebRtcSpl_GetSizeInBits(...)
+//
+// Returns the # of bits that are needed at the most to represent the number
+// specified by the |value| parameter.
+//
+// Input:
+//      - value     : Input value
+//
+// Return value     : Number of bits needed to represent |value|
+//
+
+//
+// WebRtcSpl_NormW32(...)
+//
+// Norm returns the # of left shifts required to 32-bit normalize the 32-bit
+// signed number specified by the |value| parameter.
+//
+// Input:
+//      - value     : Input value
+//
+// Return value     : Number of bit shifts needed to 32-bit normalize |value|
+//
+
+//
+// WebRtcSpl_NormW16(...)
+//
+// Norm returns the # of left shifts required to 16-bit normalize the 16-bit
+// signed number specified by the |value| parameter.
+//
+// Input:
+//      - value     : Input value
+//
+// Return value     : Number of bit shifts needed to 32-bit normalize |value|
+//
+
+//
+// WebRtcSpl_NormU32(...)
+//
+// Norm returns the # of left shifts required to 32-bit normalize the unsigned
+// 32-bit number specified by the |value| parameter.
+//
+// Input:
+//      - value     : Input value
+//
+// Return value     : Number of bit shifts needed to 32-bit normalize |value|
+//
+
+//
+// WebRtcSpl_GetScalingSquare(...)
+//
+// Returns the # of bits required to scale the samples specified in the
+// |in_vector| parameter so that, if the squares of the samples are added the
+// # of times specified by the |times| parameter, the 32-bit addition will not
+// overflow (result in WebRtc_Word32).
+//
+// Input:
+//      - in_vector         : Input vector to check scaling on
+//      - in_vector_length  : Samples in |in_vector|
+//      - times             : Number of additions to be performed
+//
+// Return value             : Number of right bit shifts needed to avoid
+//                            overflow in the addition calculation
+//
+
+//
+// WebRtcSpl_MemSetW16(...)
+//
+// Sets all the values in the WebRtc_Word16 vector |vector| of length
+// |vector_length| to the specified value |set_value|
+//
+// Input:
+//      - vector        : Pointer to the WebRtc_Word16 vector
+//      - set_value     : Value specified
+//      - vector_length : Length of vector
+//
+
+//
+// WebRtcSpl_MemSetW32(...)
+//
+// Sets all the values in the WebRtc_Word32 vector |vector| of length
+// |vector_length| to the specified value |set_value|
+//
+// Input:
+//      - vector        : Pointer to the WebRtc_Word16 vector
+//      - set_value     : Value specified
+//      - vector_length : Length of vector
+//
+
+//
+// WebRtcSpl_MemCpyReversedOrder(...)
+//
+// Copies all the values from the source WebRtc_Word16 vector |in_vector| to a
+// destination WebRtc_Word16 vector |out_vector|. It is done in reversed order,
+// meaning that the first sample of |in_vector| is copied to the last sample of
+// the |out_vector|. The procedure continues until the last sample of
+// |in_vector| has been copied to the first sample of |out_vector|. This
+// creates a reversed vector. Used in e.g. prediction in iLBC.
+//
+// Input:
+//      - in_vector     : Pointer to the first sample in a WebRtc_Word16 vector
+//                        of length |length|
+//      - vector_length : Number of elements to copy
+//
+// Output:
+//      - out_vector    : Pointer to the last sample in a WebRtc_Word16 vector
+//                        of length |length|
+//
+
+//
+// WebRtcSpl_CopyFromEndW16(...)
+//
+// Copies the rightmost |samples| of |in_vector| (of length |in_vector_length|)
+// to the vector |out_vector|.
+//
+// Input:
+//      - in_vector         : Input vector
+//      - in_vector_length  : Number of samples in |in_vector|
+//      - samples           : Number of samples to extract (from right side)
+//                            from |in_vector|
+//
+// Output:
+//      - out_vector        : Vector with the requested samples
+//
+// Return value             : Number of copied samples in |out_vector|
+//
+
+//
+// WebRtcSpl_ZerosArrayW16(...)
+// WebRtcSpl_ZerosArrayW32(...)
+//
+// Inserts the value "zero" in all positions of a w16 and a w32 vector
+// respectively.
+//
+// Input:
+//      - vector_length : Number of samples in vector
+//
+// Output:
+//      - vector        : Vector containing all zeros
+//
+// Return value         : Number of samples in vector
+//
+
+//
+// WebRtcSpl_OnesArrayW16(...)
+// WebRtcSpl_OnesArrayW32(...)
+//
+// Inserts the value "one" in all positions of a w16 and a w32 vector
+// respectively.
+//
+// Input:
+//      - vector_length : Number of samples in vector
+//
+// Output:
+//      - vector        : Vector containing all ones
+//
+// Return value         : Number of samples in vector
+//
+
+//
+// WebRtcSpl_MinValueW16(...)
+// WebRtcSpl_MinValueW32(...)
+//
+// Returns the minimum value of a vector
+//
+// Input:
+//      - vector        : Input vector
+//      - vector_length : Number of samples in vector
+//
+// Return value         : Minimum sample value in vector
+//
+
+//
+// WebRtcSpl_MaxValueW16(...)
+// WebRtcSpl_MaxValueW32(...)
+//
+// Returns the maximum value of a vector
+//
+// Input:
+//      - vector        : Input vector
+//      - vector_length : Number of samples in vector
+//
+// Return value         : Maximum sample value in vector
+//
+
+//
+// WebRtcSpl_MaxAbsValueW16(...)
+// WebRtcSpl_MaxAbsValueW32(...)
+//
+// Returns the largest absolute value of a vector
+//
+// Input:
+//      - vector        : Input vector
+//      - vector_length : Number of samples in vector
+//
+// Return value         : Maximum absolute value in vector
+//
+
+//
+// WebRtcSpl_MaxAbsIndexW16(...)
+//
+// Returns the vector index to the largest absolute value of a vector
+//
+// Input:
+//      - vector        : Input vector
+//      - vector_length : Number of samples in vector
+//
+// Return value         : Index to maximum absolute value in vector
+//
+
+//
+// WebRtcSpl_MinIndexW16(...)
+// WebRtcSpl_MinIndexW32(...)
+//
+// Returns the vector index to the minimum sample value of a vector
+//
+// Input:
+//      - vector        : Input vector
+//      - vector_length : Number of samples in vector
+//
+// Return value         : Index to minimum sample value in vector
+//
+
+//
+// WebRtcSpl_MaxIndexW16(...)
+// WebRtcSpl_MaxIndexW32(...)
+//
+// Returns the vector index to the maximum sample value of a vector
+//
+// Input:
+//      - vector        : Input vector
+//      - vector_length : Number of samples in vector
+//
+// Return value         : Index to maximum sample value in vector
+//
+
+//
+// WebRtcSpl_VectorBitShiftW16(...)
+// WebRtcSpl_VectorBitShiftW32(...)
+//
+// Bit shifts all the values in a vector up or downwards. Different calls for
+// WebRtc_Word16 and WebRtc_Word32 vectors respectively.
+//
+// Input:
+//      - vector_length : Length of vector
+//      - in_vector     : Pointer to the vector that should be bit shifted
+//      - right_shifts  : Number of right bit shifts (negative value gives left
+//                        shifts)
+//
+// Output:
+//      - out_vector    : Pointer to the result vector (can be the same as
+//                        |in_vector|)
+//
+
+//
+// WebRtcSpl_VectorBitShiftW32ToW16(...)
+//
+// Bit shifts all the values in a WebRtc_Word32 vector up or downwards and
+// stores the result as a WebRtc_Word16 vector
+//
+// Input:
+//      - vector_length : Length of vector
+//      - in_vector     : Pointer to the vector that should be bit shifted
+//      - right_shifts  : Number of right bit shifts (negative value gives left
+//                        shifts)
+//
+// Output:
+//      - out_vector    : Pointer to the result vector (can be the same as
+//                        |in_vector|)
+//
+
+//
+// WebRtcSpl_ScaleVector(...)
+//
+// Performs the vector operation:
+//  out_vector[k] = (gain*in_vector[k])>>right_shifts
+//
+// Input:
+//      - in_vector     : Input vector
+//      - gain          : Scaling gain
+//      - vector_length : Elements in the |in_vector|
+//      - right_shifts  : Number of right bit shifts applied
+//
+// Output:
+//      - out_vector    : Output vector (can be the same as |in_vector|)
+//
+
+//
+// WebRtcSpl_ScaleVectorWithSat(...)
+//
+// Performs the vector operation:
+//  out_vector[k] = SATURATE( (gain*in_vector[k])>>right_shifts )
+//
+// Input:
+//      - in_vector     : Input vector
+//      - gain          : Scaling gain
+//      - vector_length : Elements in the |in_vector|
+//      - right_shifts  : Number of right bit shifts applied
+//
+// Output:
+//      - out_vector    : Output vector (can be the same as |in_vector|)
+//
+
+//
+// WebRtcSpl_ScaleAndAddVectors(...)
+//
+// Performs the vector operation:
+//  out_vector[k] = (gain1*in_vector1[k])>>right_shifts1
+//                  + (gain2*in_vector2[k])>>right_shifts2
+//
+// Input:
+//      - in_vector1    : Input vector 1
+//      - gain1         : Gain to be used for vector 1
+//      - right_shifts1 : Right bit shift to be used for vector 1
+//      - in_vector2    : Input vector 2
+//      - gain2         : Gain to be used for vector 2
+//      - right_shifts2 : Right bit shift to be used for vector 2
+//      - vector_length : Elements in the input vectors
+//
+// Output:
+//      - out_vector    : Output vector
+//
+
+//
+// WebRtcSpl_ScaleAndAddVectorsWithRound(...)
+//
+// Performs the vector operation:
+//
+//  out_vector[k] = ((scale1*in_vector1[k]) + (scale2*in_vector2[k])
+//                      + round_value) >> right_shifts
+//
+//      where:
+//
+//  round_value = (1<<right_shifts)>>1
+//
+// Input:
+//      - in_vector1    : Input vector 1
+//      - scale1        : Gain to be used for vector 1
+//      - in_vector2    : Input vector 2
+//      - scale2        : Gain to be used for vector 2
+//      - right_shifts  : Number of right bit shifts to be applied
+//      - vector_length : Number of elements in the input vectors
+//
+// Output:
+//      - out_vector    : Output vector
+//
+
+//
+// WebRtcSpl_ReverseOrderMultArrayElements(...)
+//
+// Performs the vector operation:
+//  out_vector[n] = (in_vector[n]*window[-n])>>right_shifts
+//
+// Input:
+//      - in_vector     : Input vector
+//      - window        : Window vector (should be reversed). The pointer
+//                        should be set to the last value in the vector
+//      - right_shifts  : Number of right bit shift to be applied after the
+//                        multiplication
+//      - vector_length : Number of elements in |in_vector|
+//
+// Output:
+//      - out_vector    : Output vector (can be same as |in_vector|)
+//
+
+//
+// WebRtcSpl_ElementwiseVectorMult(...)
+//
+// Performs the vector operation:
+//  out_vector[n] = (in_vector[n]*window[n])>>right_shifts
+//
+// Input:
+//      - in_vector     : Input vector
+//      - window        : Window vector.
+//      - right_shifts  : Number of right bit shift to be applied after the
+//                        multiplication
+//      - vector_length : Number of elements in |in_vector|
+//
+// Output:
+//      - out_vector    : Output vector (can be same as |in_vector|)
+//
+
+//
+// WebRtcSpl_AddVectorsAndShift(...)
+//
+// Performs the vector operation:
+//  out_vector[k] = (in_vector1[k] + in_vector2[k])>>right_shifts
+//
+// Input:
+//      - in_vector1    : Input vector 1
+//      - in_vector2    : Input vector 2
+//      - right_shifts  : Number of right bit shift to be applied after the
+//                        multiplication
+//      - vector_length : Number of elements in |in_vector1| and |in_vector2|
+//
+// Output:
+//      - out_vector    : Output vector (can be same as |in_vector1|)
+//
+
+//
+// WebRtcSpl_AddAffineVectorToVector(...)
+//
+// Adds an affine transformed vector to another vector |out_vector|, i.e,
+// performs
+//  out_vector[k] += (in_vector[k]*gain+add_constant)>>right_shifts
+//
+// Input:
+//      - in_vector     : Input vector
+//      - gain          : Gain value, used to multiply the in vector with
+//      - add_constant  : Constant value to add (usually 1<<(right_shifts-1),
+//                        but others can be used as well
+//      - right_shifts  : Number of right bit shifts (0-16)
+//      - vector_length : Number of samples in |in_vector| and |out_vector|
+//
+// Output:
+//      - out_vector    : Vector with the output
+//
+
+//
+// WebRtcSpl_AffineTransformVector(...)
+//
+// Affine transforms a vector, i.e, performs
+//  out_vector[k] = (in_vector[k]*gain+add_constant)>>right_shifts
+//
+// Input:
+//      - in_vector     : Input vector
+//      - gain          : Gain value, used to multiply the in vector with
+//      - add_constant  : Constant value to add (usually 1<<(right_shifts-1),
+//                        but others can be used as well
+//      - right_shifts  : Number of right bit shifts (0-16)
+//      - vector_length : Number of samples in |in_vector| and |out_vector|
+//
+// Output:
+//      - out_vector    : Vector with the output
+//
+
+//
+// WebRtcSpl_AutoCorrelation(...)
+//
+// A 32-bit fix-point implementation of auto-correlation computation
+//
+// Input:
+//      - vector        : Vector to calculate autocorrelation upon
+//      - vector_length : Length (in samples) of |vector|
+//      - order         : The order up to which the autocorrelation should be
+//                        calculated
+//
+// Output:
+//      - result_vector : auto-correlation values (values should be seen
+//                        relative to each other since the absolute values
+//                        might have been down shifted to avoid overflow)
+//
+//      - scale         : The number of left shifts required to obtain the
+//                        auto-correlation in Q0
+//
+// Return value         : Number of samples in |result_vector|, i.e., (order+1)
+//
+
+//
+// WebRtcSpl_LevinsonDurbin(...)
+//
+// A 32-bit fix-point implementation of the Levinson-Durbin algorithm that
+// does NOT use the 64 bit class
+//
+// Input:
+//      - auto_corr : Vector with autocorrelation values of length >=
+//                    |use_order|+1
+//      - use_order : The LPC filter order (support up to order 20)
+//
+// Output:
+//      - lpc_coef  : lpc_coef[0..use_order] LPC coefficients in Q12
+//      - refl_coef : refl_coef[0...use_order-1]| Reflection coefficients in
+//                    Q15
+//
+// Return value     : 1 for stable 0 for unstable
+//
+
+//
+// WebRtcSpl_ReflCoefToLpc(...)
+//
+// Converts reflection coefficients |refl_coef| to LPC coefficients |lpc_coef|.
+// This version is a 16 bit operation.
+//
+// NOTE: The 16 bit refl_coef -> lpc_coef conversion might result in a
+// "slightly unstable" filter (i.e., a pole just outside the unit circle) in
+// "rare" cases even if the reflection coefficients are stable.
+//
+// Input:
+//      - refl_coef : Reflection coefficients in Q15 that should be converted
+//                    to LPC coefficients
+//      - use_order : Number of coefficients in |refl_coef|
+//
+// Output:
+//      - lpc_coef  : LPC coefficients in Q12
+//
+
+//
+// WebRtcSpl_LpcToReflCoef(...)
+//
+// Converts LPC coefficients |lpc_coef| to reflection coefficients |refl_coef|.
+// This version is a 16 bit operation.
+// The conversion is implemented by the step-down algorithm.
+//
+// Input:
+//      - lpc_coef  : LPC coefficients in Q12, that should be converted to
+//                    reflection coefficients
+//      - use_order : Number of coefficients in |lpc_coef|
+//
+// Output:
+//      - refl_coef : Reflection coefficients in Q15.
+//
+
+//
+// WebRtcSpl_AutoCorrToReflCoef(...)
+//
+// Calculates reflection coefficients (16 bit) from auto-correlation values
+//
+// Input:
+//      - auto_corr : Auto-correlation values
+//      - use_order : Number of coefficients wanted be calculated
+//
+// Output:
+//      - refl_coef : Reflection coefficients in Q15.
+//
+
+//
+// WebRtcSpl_CrossCorrelation(...)
+//
+// Calculates the cross-correlation between two sequences |vector1| and
+// |vector2|. |vector1| is fixed and |vector2| slides as the pointer is
+// increased with the amount |step_vector2|
+//
+// Input:
+//      - vector1           : First sequence (fixed throughout the correlation)
+//      - vector2           : Second sequence (slides |step_vector2| for each
+//                            new correlation)
+//      - dim_vector        : Number of samples to use in the cross-correlation
+//      - dim_cross_corr    : Number of cross-correlations to calculate (the
+//                            start position for |vector2| is updated for each
+//                            new one)
+//      - right_shifts      : Number of right bit shifts to use. This will
+//                            become the output Q-domain.
+//      - step_vector2      : How many (positive or negative) steps the
+//                            |vector2| pointer should be updated for each new
+//                            cross-correlation value.
+//
+// Output:
+//      - cross_corr        : The cross-correlation in Q(-right_shifts)
+//
+
+//
+// WebRtcSpl_GetHanningWindow(...)
+//
+// Creates (the first half of) a Hanning window. Size must be at least 1 and
+// at most 512.
+//
+// Input:
+//      - size      : Length of the requested Hanning window (1 to 512)
+//
+// Output:
+//      - window    : Hanning vector in Q14.
+//
+
+//
+// WebRtcSpl_SqrtOfOneMinusXSquared(...)
+//
+// Calculates y[k] = sqrt(1 - x[k]^2) for each element of the input vector
+// |in_vector|. Input and output values are in Q15.
+//
+// Inputs:
+//      - in_vector     : Values to calculate sqrt(1 - x^2) of
+//      - vector_length : Length of vector |in_vector|
+//
+// Output:
+//      - out_vector    : Output values in Q15
+//
+
+//
+// WebRtcSpl_IncreaseSeed(...)
+//
+// Increases the seed (and returns the new value)
+//
+// Input:
+//      - seed      : Seed for random calculation
+//
+// Output:
+//      - seed      : Updated seed value
+//
+// Return value     : The new seed value
+//
+
+//
+// WebRtcSpl_RandU(...)
+//
+// Produces a uniformly distributed value in the WebRtc_Word16 range
+//
+// Input:
+//      - seed      : Seed for random calculation
+//
+// Output:
+//      - seed      : Updated seed value
+//
+// Return value     : Uniformly distributed value in the range
+//                    [Word16_MIN...Word16_MAX]
+//
+
+//
+// WebRtcSpl_RandN(...)
+//
+// Produces a normal distributed value in the WebRtc_Word16 range
+//
+// Input:
+//      - seed      : Seed for random calculation
+//
+// Output:
+//      - seed      : Updated seed value
+//
+// Return value     : N(0,1) value in the Q13 domain
+//
+
+//
+// WebRtcSpl_RandUArray(...)
+//
+// Produces a uniformly distributed vector with elements in the WebRtc_Word16
+// range
+//
+// Input:
+//      - vector_length : Samples wanted in the vector
+//      - seed          : Seed for random calculation
+//
+// Output:
+//      - vector        : Vector with the uniform values
+//      - seed          : Updated seed value
+//
+// Return value         : Number of samples in vector, i.e., |vector_length|
+//
+
+//
+// WebRtcSpl_Sqrt(...)
+//
+// Returns the square root of the input value |value|. The precision of this
+// function is integer precision, i.e., sqrt(8) gives 2 as answer.
+// If |value| is a negative number then 0 is returned.
+//
+// Algorithm:
+//
+// A sixth order Taylor Series expansion is used here to compute the square
+// root of a number y^0.5 = (1+x)^0.5
+// where
+// x = y-1
+//   = 1+(x/2)-0.5*((x/2)^2+0.5*((x/2)^3-0.625*((x/2)^4+0.875*((x/2)^5)
+// 0.5 <= x < 1
+//
+// Input:
+//      - value     : Value to calculate sqrt of
+//
+// Return value     : Result of the sqrt calculation
+//
+
+//
+// WebRtcSpl_SqrtFloor(...)
+//
+// Returns the square root of the input value |value|. The precision of this
+// function is rounding down integer precision, i.e., sqrt(8) gives 2 as answer.
+// If |value| is a negative number then 0 is returned.
+//
+// Algorithm:
+//
+// An iterative 4 cylce/bit routine
+//
+// Input:
+//      - value     : Value to calculate sqrt of
+//
+// Return value     : Result of the sqrt calculation
+//
+
+//
+// WebRtcSpl_DivU32U16(...)
+//
+// Divides a WebRtc_UWord32 |num| by a WebRtc_UWord16 |den|.
+//
+// If |den|==0, (WebRtc_UWord32)0xFFFFFFFF is returned.
+//
+// Input:
+//      - num       : Numerator
+//      - den       : Denominator
+//
+// Return value     : Result of the division (as a WebRtc_UWord32), i.e., the
+//                    integer part of num/den.
+//
+
+//
+// WebRtcSpl_DivW32W16(...)
+//
+// Divides a WebRtc_Word32 |num| by a WebRtc_Word16 |den|.
+//
+// If |den|==0, (WebRtc_Word32)0x7FFFFFFF is returned.
+//
+// Input:
+//      - num       : Numerator
+//      - den       : Denominator
+//
+// Return value     : Result of the division (as a WebRtc_Word32), i.e., the
+//                    integer part of num/den.
+//
+
+//
+// WebRtcSpl_DivW32W16ResW16(...)
+//
+// Divides a WebRtc_Word32 |num| by a WebRtc_Word16 |den|, assuming that the
+// result is less than 32768, otherwise an unpredictable result will occur.
+//
+// If |den|==0, (WebRtc_Word16)0x7FFF is returned.
+//
+// Input:
+//      - num       : Numerator
+//      - den       : Denominator
+//
+// Return value     : Result of the division (as a WebRtc_Word16), i.e., the
+//                    integer part of num/den.
+//
+
+//
+// WebRtcSpl_DivResultInQ31(...)
+//
+// Divides a WebRtc_Word32 |num| by a WebRtc_Word16 |den|, assuming that the
+// absolute value of the denominator is larger than the numerator, otherwise
+// an unpredictable result will occur.
+//
+// Input:
+//      - num       : Numerator
+//      - den       : Denominator
+//
+// Return value     : Result of the division in Q31.
+//
+
+//
+// WebRtcSpl_DivW32HiLow(...)
+//
+// Divides a WebRtc_Word32 |num| by a denominator in hi, low format. The
+// absolute value of the denominator has to be larger (or equal to) the
+// numerator.
+//
+// Input:
+//      - num       : Numerator
+//      - den_hi    : High part of denominator
+//      - den_low   : Low part of denominator
+//
+// Return value     : Divided value in Q31
+//
+
+//
+// WebRtcSpl_Energy(...)
+//
+// Calculates the energy of a vector
+//
+// Input:
+//      - vector        : Vector which the energy should be calculated on
+//      - vector_length : Number of samples in vector
+//
+// Output:
+//      - scale_factor  : Number of left bit shifts needed to get the physical
+//                        energy value, i.e, to get the Q0 value
+//
+// Return value         : Energy value in Q(-|scale_factor|)
+//
+
+//
+// WebRtcSpl_FilterAR(...)
+//
+// Performs a 32-bit AR filtering on a vector in Q12
+//
+// Input:
+//  - ar_coef                   : AR-coefficient vector (values in Q12),
+//                                ar_coef[0] must be 4096.
+//  - ar_coef_length            : Number of coefficients in |ar_coef|.
+//  - in_vector                 : Vector to be filtered.
+//  - in_vector_length          : Number of samples in |in_vector|.
+//  - filter_state              : Current state (higher part) of the filter.
+//  - filter_state_length       : Length (in samples) of |filter_state|.
+//  - filter_state_low          : Current state (lower part) of the filter.
+//  - filter_state_low_length   : Length (in samples) of |filter_state_low|.
+//  - out_vector_low_length     : Maximum length (in samples) of
+//                                |out_vector_low|.
+//
+// Output:
+//  - filter_state              : Updated state (upper part) vector.
+//  - filter_state_low          : Updated state (lower part) vector.
+//  - out_vector                : Vector containing the upper part of the
+//                                filtered values.
+//  - out_vector_low            : Vector containing the lower part of the
+//                                filtered values.
+//
+// Return value                 : Number of samples in the |out_vector|.
+//
+
+//
+// WebRtcSpl_FilterMAFastQ12(...)
+//
+// Performs a MA filtering on a vector in Q12
+//
+// Input:
+//      - in_vector         : Input samples (state in positions
+//                            in_vector[-order] .. in_vector[-1])
+//      - ma_coef           : Filter coefficients (in Q12)
+//      - ma_coef_length    : Number of B coefficients (order+1)
+//      - vector_length     : Number of samples to be filtered
+//
+// Output:
+//      - out_vector        : Filtered samples
+//
+
+
+//
+// WebRtcSpl_DotProductWithScale(...)
+//
+// Calculates the dot product between two (WebRtc_Word16) vectors
+//
+// Input:
+//      - vector1       : Vector 1
+//      - vector2       : Vector 2
+//      - vector_length : Number of samples used in the dot product
+//      - scaling       : The number of right bit shifts to apply on each term
+//                        during calculation to avoid overflow, i.e., the
+//                        output will be in Q(-|scaling|)
+//
+// Return value         : The dot product in Q(-scaling)
+//
+
+//
+// WebRtcSpl_ComplexIFFT(...)
+//
+// Complex Inverse FFT
+//
+// Computes an inverse complex 2^|stages|-point FFT on the input vector, which
+// is in bit-reversed order. The original content of the vector is destroyed in
+// the process, since the input is overwritten by the output, normal-ordered,
+// FFT vector. With X as the input complex vector, y as the output complex
+// vector and with M = 2^|stages|, the following is computed:
+//
+//        M-1
+// y(k) = sum[X(i)*[cos(2*pi*i*k/M) + j*sin(2*pi*i*k/M)]]
+//        i=0
+//
+// The implementations are optimized for speed, not for code size. It uses the
+// decimation-in-time algorithm with radix-2 butterfly technique.
+//
+// Input:
+//      - vector    : In pointer to complex vector containing 2^|stages|
+//                    real elements interleaved with 2^|stages| imaginary
+//                    elements.
+//                    [ReImReImReIm....]
+//                    The elements are in Q(-scale) domain, see more on Return
+//                    Value below.
+//
+//      - stages    : Number of FFT stages. Must be at least 3 and at most 10,
+//                    since the table WebRtcSpl_kSinTable1024[] is 1024
+//                    elements long.
+//
+//      - mode      : This parameter gives the user to choose how the FFT
+//                    should work.
+//                    mode==0: Low-complexity and Low-accuracy mode
+//                    mode==1: High-complexity and High-accuracy mode
+//
+// Output:
+//      - vector    : Out pointer to the FFT vector (the same as input).
+//
+// Return Value     : The scale value that tells the number of left bit shifts
+//                    that the elements in the |vector| should be shifted with
+//                    in order to get Q0 values, i.e. the physically correct
+//                    values. The scale parameter is always 0 or positive,
+//                    except if N>1024 (|stages|>10), which returns a scale
+//                    value of -1, indicating error.
+//
+
+//
+// WebRtcSpl_ComplexFFT(...)
+//
+// Complex FFT
+//
+// Computes a complex 2^|stages|-point FFT on the input vector, which is in
+// bit-reversed order. The original content of the vector is destroyed in
+// the process, since the input is overwritten by the output, normal-ordered,
+// FFT vector. With x as the input complex vector, Y as the output complex
+// vector and with M = 2^|stages|, the following is computed:
+//
+//              M-1
+// Y(k) = 1/M * sum[x(i)*[cos(2*pi*i*k/M) + j*sin(2*pi*i*k/M)]]
+//              i=0
+//
+// The implementations are optimized for speed, not for code size. It uses the
+// decimation-in-time algorithm with radix-2 butterfly technique.
+//
+// This routine prevents overflow by scaling by 2 before each FFT stage. This is
+// a fixed scaling, for proper normalization - there will be log2(n) passes, so
+// this results in an overall factor of 1/n, distributed to maximize arithmetic
+// accuracy.
+//
+// Input:
+//      - vector    : In pointer to complex vector containing 2^|stages| real
+//                    elements interleaved with 2^|stages| imaginary elements.
+//                    [ReImReImReIm....]
+//                    The output is in the Q0 domain.
+//
+//      - stages    : Number of FFT stages. Must be at least 3 and at most 10,
+//                    since the table WebRtcSpl_kSinTable1024[] is 1024
+//                    elements long.
+//
+//      - mode      : This parameter gives the user to choose how the FFT
+//                    should work.
+//                    mode==0: Low-complexity and Low-accuracy mode
+//                    mode==1: High-complexity and High-accuracy mode
+//
+// Output:
+//      - vector    : The output FFT vector is in the Q0 domain.
+//
+// Return value     : The scale parameter is always 0, except if N>1024,
+//                    which returns a scale value of -1, indicating error.
+//
+
+//
+// WebRtcSpl_AnalysisQMF(...)
+//
+// Splits a 0-2*F Hz signal into two sub bands: 0-F Hz and F-2*F Hz. The
+// current version has F = 8000, therefore, a super-wideband audio signal is
+// split to lower-band 0-8 kHz and upper-band 8-16 kHz.
+//
+// Input:
+//      - in_data       : Wide band speech signal, 320 samples (10 ms)
+//
+// Input & Output:
+//      - filter_state1 : Filter state for first All-pass filter
+//      - filter_state2 : Filter state for second All-pass filter
+//
+// Output:
+//      - low_band      : Lower-band signal 0-8 kHz band, 160 samples (10 ms)
+//      - high_band     : Upper-band signal 8-16 kHz band (flipped in frequency
+//                        domain), 160 samples (10 ms)
+//
+
+//
+// WebRtcSpl_SynthesisQMF(...)
+//
+// Combines the two sub bands (0-F and F-2*F Hz) into a signal of 0-2*F
+// Hz, (current version has F = 8000 Hz). So the filter combines lower-band
+// (0-8 kHz) and upper-band (8-16 kHz) channels to obtain super-wideband 0-16
+// kHz audio.
+//
+// Input:
+//      - low_band      : The signal with the 0-8 kHz band, 160 samples (10 ms)
+//      - high_band     : The signal with the 8-16 kHz band, 160 samples (10 ms)
+//
+// Input & Output:
+//      - filter_state1 : Filter state for first All-pass filter
+//      - filter_state2 : Filter state for second All-pass filter
+//
+// Output:
+//      - out_data      : Super-wideband speech signal, 0-16 kHz
+//
+
+// WebRtc_Word16 WebRtcSpl_SatW32ToW16(...)
+//
+// This function saturates a 32-bit word into a 16-bit word.
+// 
+// Input:
+//      - value32   : The value of a 32-bit word.
+//
+// Output:
+//      - out16     : the saturated 16-bit word.
+//
+
+// int32_t WebRtc_MulAccumW16(...)
+//
+// This function multiply a 16-bit word by a 16-bit word, and accumulate this
+// value to a 32-bit integer.
+// 
+// Input:
+//      - a    : The value of the first 16-bit word.
+//      - b    : The value of the second 16-bit word.
+//      - c    : The value of an 32-bit integer.
+//
+// Return Value: The value of a * b + c.
+//
+
+// WebRtc_Word16 WebRtcSpl_get_version(...)
+//
+// This function gives the version string of the Signal Processing Library.
+//
+// Input:
+//      - length_in_bytes   : The size of Allocated space (in Bytes) where
+//                            the version number is written to (in string format).
+//
+// Output:
+//      - version           : Pointer to a buffer where the version number is written to.
+//
diff --git a/trunk/src/common_audio/signal_processing/include/spl_inl.h b/trunk/src/common_audio/signal_processing/include/spl_inl.h
new file mode 100644
index 0000000..23b3209
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/include/spl_inl.h
@@ -0,0 +1,159 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+// This header file includes the inline functions in
+// the fix point signal processing library.
+
+#ifndef WEBRTC_SPL_SPL_INL_H_
+#define WEBRTC_SPL_SPL_INL_H_
+
+#ifdef WEBRTC_ARCH_ARM_V7A
+#include "spl_inl_armv7.h"
+#else
+
+static __inline WebRtc_Word16 WebRtcSpl_SatW32ToW16(WebRtc_Word32 value32) {
+  WebRtc_Word16 out16 = (WebRtc_Word16) value32;
+
+  if (value32 > 32767)
+    out16 = 32767;
+  else if (value32 < -32768)
+    out16 = -32768;
+
+  return out16;
+}
+
+static __inline WebRtc_Word16 WebRtcSpl_AddSatW16(WebRtc_Word16 a,
+                                                  WebRtc_Word16 b) {
+  return WebRtcSpl_SatW32ToW16((WebRtc_Word32) a + (WebRtc_Word32) b);
+}
+
+static __inline WebRtc_Word32 WebRtcSpl_AddSatW32(WebRtc_Word32 l_var1,
+                                                  WebRtc_Word32 l_var2) {
+  WebRtc_Word32 l_sum;
+
+  // perform long addition
+  l_sum = l_var1 + l_var2;
+
+  // check for under or overflow
+  if (WEBRTC_SPL_IS_NEG(l_var1)) {
+    if (WEBRTC_SPL_IS_NEG(l_var2) && !WEBRTC_SPL_IS_NEG(l_sum)) {
+        l_sum = (WebRtc_Word32)0x80000000;
+    }
+  } else {
+    if (!WEBRTC_SPL_IS_NEG(l_var2) && WEBRTC_SPL_IS_NEG(l_sum)) {
+        l_sum = (WebRtc_Word32)0x7FFFFFFF;
+    }
+  }
+
+  return l_sum;
+}
+
+static __inline WebRtc_Word16 WebRtcSpl_SubSatW16(WebRtc_Word16 var1,
+                                                  WebRtc_Word16 var2) {
+  return WebRtcSpl_SatW32ToW16((WebRtc_Word32) var1 - (WebRtc_Word32) var2);
+}
+
+static __inline WebRtc_Word32 WebRtcSpl_SubSatW32(WebRtc_Word32 l_var1,
+                                                  WebRtc_Word32 l_var2) {
+  WebRtc_Word32 l_diff;
+
+  // perform subtraction
+  l_diff = l_var1 - l_var2;
+
+  // check for underflow
+  if ((l_var1 < 0) && (l_var2 > 0) && (l_diff > 0))
+    l_diff = (WebRtc_Word32)0x80000000;
+  // check for overflow
+  if ((l_var1 > 0) && (l_var2 < 0) && (l_diff < 0))
+    l_diff = (WebRtc_Word32)0x7FFFFFFF;
+
+  return l_diff;
+}
+
+static __inline WebRtc_Word16 WebRtcSpl_GetSizeInBits(WebRtc_UWord32 n) {
+  int bits;
+
+  if (0xFFFF0000 & n) {
+    bits = 16;
+  } else {
+    bits = 0;
+  }
+  if (0x0000FF00 & (n >> bits)) bits += 8;
+  if (0x000000F0 & (n >> bits)) bits += 4;
+  if (0x0000000C & (n >> bits)) bits += 2;
+  if (0x00000002 & (n >> bits)) bits += 1;
+  if (0x00000001 & (n >> bits)) bits += 1;
+
+  return bits;
+}
+
+static __inline int WebRtcSpl_NormW32(WebRtc_Word32 a) {
+  int zeros;
+
+  if (a <= 0) a ^= 0xFFFFFFFF;
+
+  if (!(0xFFFF8000 & a)) {
+    zeros = 16;
+  } else {
+    zeros = 0;
+  }
+  if (!(0xFF800000 & (a << zeros))) zeros += 8;
+  if (!(0xF8000000 & (a << zeros))) zeros += 4;
+  if (!(0xE0000000 & (a << zeros))) zeros += 2;
+  if (!(0xC0000000 & (a << zeros))) zeros += 1;
+
+  return zeros;
+}
+
+static __inline int WebRtcSpl_NormU32(WebRtc_UWord32 a) {
+  int zeros;
+
+  if (a == 0) return 0;
+
+  if (!(0xFFFF0000 & a)) {
+    zeros = 16;
+  } else {
+    zeros = 0;
+  }
+  if (!(0xFF000000 & (a << zeros))) zeros += 8;
+  if (!(0xF0000000 & (a << zeros))) zeros += 4;
+  if (!(0xC0000000 & (a << zeros))) zeros += 2;
+  if (!(0x80000000 & (a << zeros))) zeros += 1;
+
+  return zeros;
+}
+
+static __inline int WebRtcSpl_NormW16(WebRtc_Word16 a) {
+  int zeros;
+
+  if (a <= 0) a ^= 0xFFFF;
+
+  if (!(0xFF80 & a)) {
+    zeros = 8;
+  } else {
+    zeros = 0;
+  }
+  if (!(0xF800 & (a << zeros))) zeros += 4;
+  if (!(0xE000 & (a << zeros))) zeros += 2;
+  if (!(0xC000 & (a << zeros))) zeros += 1;
+
+  return zeros;
+}
+
+static __inline int32_t WebRtc_MulAccumW16(int16_t a,
+                                          int16_t b,
+                                          int32_t c) {
+  return (a * b + c);
+}
+
+#endif  // WEBRTC_ARCH_ARM_V7A
+
+#endif  // WEBRTC_SPL_SPL_INL_H_
diff --git a/trunk/src/common_audio/signal_processing/include/spl_inl_armv7.h b/trunk/src/common_audio/signal_processing/include/spl_inl_armv7.h
new file mode 100644
index 0000000..5b19c2c
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/include/spl_inl_armv7.h
@@ -0,0 +1,140 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+// This header file includes the inline functions for ARM processors in
+// the fix point signal processing library.
+
+#ifndef WEBRTC_SPL_SPL_INL_ARMV7_H_
+#define WEBRTC_SPL_SPL_INL_ARMV7_H_
+
+// TODO(kma): Replace some assembly code with GCC intrinsics
+// (e.g. __builtin_clz).
+
+static __inline WebRtc_Word32 WEBRTC_SPL_MUL_16_32_RSFT16(WebRtc_Word16 a,
+                                                          WebRtc_Word32 b) {
+  WebRtc_Word32 tmp;
+  __asm__("smulwb %0, %1, %2":"=r"(tmp):"r"(b), "r"(a));
+  return tmp;
+}
+
+static __inline WebRtc_Word32 WEBRTC_SPL_MUL_32_32_RSFT32(WebRtc_Word16 a,
+                                                          WebRtc_Word16 b,
+                                                          WebRtc_Word32 c) {
+  WebRtc_Word32 tmp;
+  __asm__("pkhbt %0, %1, %2, lsl #16" : "=r"(tmp) : "r"(b), "r"(a));
+  __asm__("smmul %0, %1, %2":"=r"(tmp):"r"(tmp), "r"(c));
+  return tmp;
+}
+
+static __inline WebRtc_Word32 WEBRTC_SPL_MUL_32_32_RSFT32BI(WebRtc_Word32 a,
+                                                            WebRtc_Word32 b) {
+  WebRtc_Word32 tmp;
+  __asm__("smmul %0, %1, %2":"=r"(tmp):"r"(a), "r"(b));
+  return tmp;
+}
+
+static __inline WebRtc_Word32 WEBRTC_SPL_MUL_16_16(WebRtc_Word16 a,
+                                                   WebRtc_Word16 b) {
+  WebRtc_Word32 tmp;
+  __asm__("smulbb %0, %1, %2":"=r"(tmp):"r"(a), "r"(b));
+  return tmp;
+}
+
+static __inline int32_t WebRtc_MulAccumW16(int16_t a,
+                                          int16_t b,
+                                          int32_t c) {
+  int32_t tmp = 0;
+  __asm__("smlabb %0, %1, %2, %3":"=r"(tmp):"r"(a), "r"(b), "r"(c));
+  return tmp;
+}
+
+static __inline WebRtc_Word16 WebRtcSpl_AddSatW16(WebRtc_Word16 a,
+                                                  WebRtc_Word16 b) {
+  WebRtc_Word32 s_sum;
+
+  __asm__("qadd16 %0, %1, %2":"=r"(s_sum):"r"(a), "r"(b));
+
+  return (WebRtc_Word16) s_sum;
+}
+
+static __inline WebRtc_Word32 WebRtcSpl_AddSatW32(WebRtc_Word32 l_var1,
+                                                  WebRtc_Word32 l_var2) {
+  WebRtc_Word32 l_sum;
+
+  __asm__("qadd %0, %1, %2":"=r"(l_sum):"r"(l_var1), "r"(l_var2));
+
+  return l_sum;
+}
+
+static __inline WebRtc_Word16 WebRtcSpl_SubSatW16(WebRtc_Word16 var1,
+                                                  WebRtc_Word16 var2) {
+  WebRtc_Word32 s_sub;
+
+  __asm__("qsub16 %0, %1, %2":"=r"(s_sub):"r"(var1), "r"(var2));
+
+  return (WebRtc_Word16)s_sub;
+}
+
+static __inline WebRtc_Word32 WebRtcSpl_SubSatW32(WebRtc_Word32 l_var1,
+                                                  WebRtc_Word32 l_var2) {
+  WebRtc_Word32 l_sub;
+
+  __asm__("qsub %0, %1, %2":"=r"(l_sub):"r"(l_var1), "r"(l_var2));
+
+  return l_sub;
+}
+
+static __inline WebRtc_Word16 WebRtcSpl_GetSizeInBits(WebRtc_UWord32 n) {
+  WebRtc_Word32 tmp;
+
+  __asm__("clz %0, %1":"=r"(tmp):"r"(n));
+
+  return (WebRtc_Word16)(32 - tmp);
+}
+
+static __inline int WebRtcSpl_NormW32(WebRtc_Word32 a) {
+  WebRtc_Word32 tmp;
+
+  if (a <= 0) a ^= 0xFFFFFFFF;
+
+  __asm__("clz %0, %1":"=r"(tmp):"r"(a));
+
+  return tmp - 1;
+}
+
+static __inline int WebRtcSpl_NormU32(WebRtc_UWord32 a) {
+  int tmp;
+
+  if (a == 0) return 0;
+
+  __asm__("clz %0, %1":"=r"(tmp):"r"(a));
+
+  return tmp;
+}
+
+static __inline int WebRtcSpl_NormW16(WebRtc_Word16 a) {
+  WebRtc_Word32 tmp;
+
+  if (a <= 0) a ^= 0xFFFFFFFF;
+
+  __asm__("clz %0, %1":"=r"(tmp):"r"(a));
+
+  return tmp - 17;
+}
+
+static __inline WebRtc_Word16 WebRtcSpl_SatW32ToW16(WebRtc_Word32 value32) {
+  WebRtc_Word16 out16;
+
+  __asm__("ssat %r0, #16, %r1" : "=r"(out16) : "r"(value32));
+
+  return out16;
+}
+#endif  // WEBRTC_SPL_SPL_INL_ARMV7_H_
diff --git a/trunk/src/common_audio/signal_processing/levinson_durbin.c b/trunk/src/common_audio/signal_processing/levinson_durbin.c
new file mode 100644
index 0000000..4e11cdb
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/levinson_durbin.c
@@ -0,0 +1,259 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains the function WebRtcSpl_LevinsonDurbin().
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "signal_processing_library.h"
+
+#define SPL_LEVINSON_MAXORDER 20
+
+WebRtc_Word16 WebRtcSpl_LevinsonDurbin(WebRtc_Word32 *R, WebRtc_Word16 *A, WebRtc_Word16 *K,
+                                       WebRtc_Word16 order)
+{
+    WebRtc_Word16 i, j;
+    // Auto-correlation coefficients in high precision
+    WebRtc_Word16 R_hi[SPL_LEVINSON_MAXORDER + 1], R_low[SPL_LEVINSON_MAXORDER + 1];
+    // LPC coefficients in high precision
+    WebRtc_Word16 A_hi[SPL_LEVINSON_MAXORDER + 1], A_low[SPL_LEVINSON_MAXORDER + 1];
+    // LPC coefficients for next iteration
+    WebRtc_Word16 A_upd_hi[SPL_LEVINSON_MAXORDER + 1], A_upd_low[SPL_LEVINSON_MAXORDER + 1];
+    // Reflection coefficient in high precision
+    WebRtc_Word16 K_hi, K_low;
+    // Prediction gain Alpha in high precision and with scale factor
+    WebRtc_Word16 Alpha_hi, Alpha_low, Alpha_exp;
+    WebRtc_Word16 tmp_hi, tmp_low;
+    WebRtc_Word32 temp1W32, temp2W32, temp3W32;
+    WebRtc_Word16 norm;
+
+    // Normalize the autocorrelation R[0]...R[order+1]
+
+    norm = WebRtcSpl_NormW32(R[0]);
+
+    for (i = order; i >= 0; i--)
+    {
+        temp1W32 = WEBRTC_SPL_LSHIFT_W32(R[i], norm);
+        // Put R in hi and low format
+        R_hi[i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
+        R_low[i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32
+                - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)R_hi[i], 16)), 1);
+    }
+
+    // K = A[1] = -R[1] / R[0]
+
+    temp2W32 = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)R_hi[1],16)
+            + WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)R_low[1],1); // R[1] in Q31
+    temp3W32 = WEBRTC_SPL_ABS_W32(temp2W32); // abs R[1]
+    temp1W32 = WebRtcSpl_DivW32HiLow(temp3W32, R_hi[0], R_low[0]); // abs(R[1])/R[0] in Q31
+    // Put back the sign on R[1]
+    if (temp2W32 > 0)
+    {
+        temp1W32 = -temp1W32;
+    }
+
+    // Put K in hi and low format
+    K_hi = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
+    K_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32
+            - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)K_hi, 16)), 1);
+
+    // Store first reflection coefficient
+    K[0] = K_hi;
+
+    temp1W32 = WEBRTC_SPL_RSHIFT_W32(temp1W32, 4); // A[1] in Q27
+
+    // Put A[1] in hi and low format
+    A_hi[1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
+    A_low[1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32
+            - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_hi[1], 16)), 1);
+
+    // Alpha = R[0] * (1-K^2)
+
+    temp1W32 = (((WEBRTC_SPL_MUL_16_16(K_hi, K_low) >> 14) + WEBRTC_SPL_MUL_16_16(K_hi, K_hi))
+            << 1); // temp1W32 = k^2 in Q31
+
+    temp1W32 = WEBRTC_SPL_ABS_W32(temp1W32); // Guard against <0
+    temp1W32 = (WebRtc_Word32)0x7fffffffL - temp1W32; // temp1W32 = (1 - K[0]*K[0]) in Q31
+
+    // Store temp1W32 = 1 - K[0]*K[0] on hi and low format
+    tmp_hi = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
+    tmp_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32
+            - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)tmp_hi, 16)), 1);
+
+    // Calculate Alpha in Q31
+    temp1W32 = ((WEBRTC_SPL_MUL_16_16(R_hi[0], tmp_hi)
+            + (WEBRTC_SPL_MUL_16_16(R_hi[0], tmp_low) >> 15)
+            + (WEBRTC_SPL_MUL_16_16(R_low[0], tmp_hi) >> 15)) << 1);
+
+    // Normalize Alpha and put it in hi and low format
+
+    Alpha_exp = WebRtcSpl_NormW32(temp1W32);
+    temp1W32 = WEBRTC_SPL_LSHIFT_W32(temp1W32, Alpha_exp);
+    Alpha_hi = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
+    Alpha_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32
+            - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)Alpha_hi, 16)), 1);
+
+    // Perform the iterative calculations in the Levinson-Durbin algorithm
+
+    for (i = 2; i <= order; i++)
+    {
+        /*                    ----
+         temp1W32 =  R[i] + > R[j]*A[i-j]
+         /
+         ----
+         j=1..i-1
+         */
+
+        temp1W32 = 0;
+
+        for (j = 1; j < i; j++)
+        {
+            // temp1W32 is in Q31
+            temp1W32 += ((WEBRTC_SPL_MUL_16_16(R_hi[j], A_hi[i-j]) << 1)
+                    + (((WEBRTC_SPL_MUL_16_16(R_hi[j], A_low[i-j]) >> 15)
+                            + (WEBRTC_SPL_MUL_16_16(R_low[j], A_hi[i-j]) >> 15)) << 1));
+        }
+
+        temp1W32 = WEBRTC_SPL_LSHIFT_W32(temp1W32, 4);
+        temp1W32 += (WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)R_hi[i], 16)
+                + WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)R_low[i], 1));
+
+        // K = -temp1W32 / Alpha
+        temp2W32 = WEBRTC_SPL_ABS_W32(temp1W32); // abs(temp1W32)
+        temp3W32 = WebRtcSpl_DivW32HiLow(temp2W32, Alpha_hi, Alpha_low); // abs(temp1W32)/Alpha
+
+        // Put the sign of temp1W32 back again
+        if (temp1W32 > 0)
+        {
+            temp3W32 = -temp3W32;
+        }
+
+        // Use the Alpha shifts from earlier to de-normalize
+        norm = WebRtcSpl_NormW32(temp3W32);
+        if ((Alpha_exp <= norm) || (temp3W32 == 0))
+        {
+            temp3W32 = WEBRTC_SPL_LSHIFT_W32(temp3W32, Alpha_exp);
+        } else
+        {
+            if (temp3W32 > 0)
+            {
+                temp3W32 = (WebRtc_Word32)0x7fffffffL;
+            } else
+            {
+                temp3W32 = (WebRtc_Word32)0x80000000L;
+            }
+        }
+
+        // Put K on hi and low format
+        K_hi = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(temp3W32, 16);
+        K_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp3W32
+                - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)K_hi, 16)), 1);
+
+        // Store Reflection coefficient in Q15
+        K[i - 1] = K_hi;
+
+        // Test for unstable filter.
+        // If unstable return 0 and let the user decide what to do in that case
+
+        if ((WebRtc_Word32)WEBRTC_SPL_ABS_W16(K_hi) > (WebRtc_Word32)32750)
+        {
+            return 0; // Unstable filter
+        }
+
+        /*
+         Compute updated LPC coefficient: Anew[i]
+         Anew[j]= A[j] + K*A[i-j]   for j=1..i-1
+         Anew[i]= K
+         */
+
+        for (j = 1; j < i; j++)
+        {
+            // temp1W32 = A[j] in Q27
+            temp1W32 = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_hi[j],16)
+                    + WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_low[j],1);
+
+            // temp1W32 += K*A[i-j] in Q27
+            temp1W32 += ((WEBRTC_SPL_MUL_16_16(K_hi, A_hi[i-j])
+                    + (WEBRTC_SPL_MUL_16_16(K_hi, A_low[i-j]) >> 15)
+                    + (WEBRTC_SPL_MUL_16_16(K_low, A_hi[i-j]) >> 15)) << 1);
+
+            // Put Anew in hi and low format
+            A_upd_hi[j] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
+            A_upd_low[j] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32
+                    - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_upd_hi[j], 16)), 1);
+        }
+
+        // temp3W32 = K in Q27 (Convert from Q31 to Q27)
+        temp3W32 = WEBRTC_SPL_RSHIFT_W32(temp3W32, 4);
+
+        // Store Anew in hi and low format
+        A_upd_hi[i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(temp3W32, 16);
+        A_upd_low[i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp3W32
+                - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_upd_hi[i], 16)), 1);
+
+        // Alpha = Alpha * (1-K^2)
+
+        temp1W32 = (((WEBRTC_SPL_MUL_16_16(K_hi, K_low) >> 14)
+                + WEBRTC_SPL_MUL_16_16(K_hi, K_hi)) << 1); // K*K in Q31
+
+        temp1W32 = WEBRTC_SPL_ABS_W32(temp1W32); // Guard against <0
+        temp1W32 = (WebRtc_Word32)0x7fffffffL - temp1W32; // 1 - K*K  in Q31
+
+        // Convert 1- K^2 in hi and low format
+        tmp_hi = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
+        tmp_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32
+                - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)tmp_hi, 16)), 1);
+
+        // Calculate Alpha = Alpha * (1-K^2) in Q31
+        temp1W32 = ((WEBRTC_SPL_MUL_16_16(Alpha_hi, tmp_hi)
+                + (WEBRTC_SPL_MUL_16_16(Alpha_hi, tmp_low) >> 15)
+                + (WEBRTC_SPL_MUL_16_16(Alpha_low, tmp_hi) >> 15)) << 1);
+
+        // Normalize Alpha and store it on hi and low format
+
+        norm = WebRtcSpl_NormW32(temp1W32);
+        temp1W32 = WEBRTC_SPL_LSHIFT_W32(temp1W32, norm);
+
+        Alpha_hi = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
+        Alpha_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32
+                - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)Alpha_hi, 16)), 1);
+
+        // Update the total normalization of Alpha
+        Alpha_exp = Alpha_exp + norm;
+
+        // Update A[]
+
+        for (j = 1; j <= i; j++)
+        {
+            A_hi[j] = A_upd_hi[j];
+            A_low[j] = A_upd_low[j];
+        }
+    }
+
+    /*
+     Set A[0] to 1.0 and store the A[i] i=1...order in Q12
+     (Convert from Q27 and use rounding)
+     */
+
+    A[0] = 4096;
+
+    for (i = 1; i <= order; i++)
+    {
+        // temp1W32 in Q27
+        temp1W32 = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_hi[i], 16)
+                + WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_low[i], 1);
+        // Round and store upper word
+        A[i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32<<1)+(WebRtc_Word32)32768, 16);
+    }
+    return 1; // Stable filters
+}
diff --git a/trunk/src/common_audio/signal_processing/lpc_to_refl_coef.c b/trunk/src/common_audio/signal_processing/lpc_to_refl_coef.c
new file mode 100644
index 0000000..2cb83c2
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/lpc_to_refl_coef.c
@@ -0,0 +1,57 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains the function WebRtcSpl_LpcToReflCoef().
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "signal_processing_library.h"
+
+#define SPL_LPC_TO_REFL_COEF_MAX_AR_MODEL_ORDER 50
+
+void WebRtcSpl_LpcToReflCoef(WebRtc_Word16* a16, int use_order, WebRtc_Word16* k16)
+{
+    int m, k;
+    WebRtc_Word32 tmp32[SPL_LPC_TO_REFL_COEF_MAX_AR_MODEL_ORDER];
+    WebRtc_Word32 tmp_inv_denom32;
+    WebRtc_Word16 tmp_inv_denom16;
+
+    k16[use_order - 1] = WEBRTC_SPL_LSHIFT_W16(a16[use_order], 3); //Q12<<3 => Q15
+    for (m = use_order - 1; m > 0; m--)
+    {
+        // (1 - k^2) in Q30
+        tmp_inv_denom32 = ((WebRtc_Word32)1073741823) - WEBRTC_SPL_MUL_16_16(k16[m], k16[m]);
+        // (1 - k^2) in Q15
+        tmp_inv_denom16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp_inv_denom32, 15);
+
+        for (k = 1; k <= m; k++)
+        {
+            // tmp[k] = (a[k] - RC[m] * a[m-k+1]) / (1.0 - RC[m]*RC[m]);
+
+            // [Q12<<16 - (Q15*Q12)<<1] = [Q28 - Q28] = Q28
+            tmp32[k] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)a16[k], 16)
+                    - WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_16(k16[m], a16[m-k+1]), 1);
+
+            tmp32[k] = WebRtcSpl_DivW32W16(tmp32[k], tmp_inv_denom16); //Q28/Q15 = Q13
+        }
+
+        for (k = 1; k < m; k++)
+        {
+            a16[k] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32[k], 1); //Q13>>1 => Q12
+        }
+
+        tmp32[m] = WEBRTC_SPL_SAT(8191, tmp32[m], -8191);
+        k16[m - 1] = (WebRtc_Word16)WEBRTC_SPL_LSHIFT_W32(tmp32[m], 2); //Q13<<2 => Q15
+    }
+    return;
+}
diff --git a/trunk/src/common_audio/signal_processing/min_max_operations.c b/trunk/src/common_audio/signal_processing/min_max_operations.c
new file mode 100644
index 0000000..57eaff7
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/min_max_operations.c
@@ -0,0 +1,265 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * This file contains the implementation of functions
+ * WebRtcSpl_MaxAbsValueW16()
+ * WebRtcSpl_MaxAbsIndexW16()
+ * WebRtcSpl_MaxAbsValueW32()
+ * WebRtcSpl_MaxValueW16()
+ * WebRtcSpl_MaxIndexW16()
+ * WebRtcSpl_MaxValueW32()
+ * WebRtcSpl_MaxIndexW32()
+ * WebRtcSpl_MinValueW16()
+ * WebRtcSpl_MinIndexW16()
+ * WebRtcSpl_MinValueW32()
+ * WebRtcSpl_MinIndexW32()
+ *
+ * The description header can be found in signal_processing_library.h.
+ *
+ */
+
+#include "signal_processing_library.h"
+
+#if !(defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM_NEON))
+
+// Maximum absolute value of word16 vector.
+WebRtc_Word16 WebRtcSpl_MaxAbsValueW16(const WebRtc_Word16 *vector, WebRtc_Word16 length)
+{
+    WebRtc_Word32 tempMax = 0;
+    WebRtc_Word32 absVal;
+    WebRtc_Word16 totMax;
+    int i;
+    G_CONST WebRtc_Word16 *tmpvector = vector;
+
+    for (i = 0; i < length; i++)
+    {
+        absVal = WEBRTC_SPL_ABS_W32((*tmpvector));
+        if (absVal > tempMax)
+        {
+            tempMax = absVal;
+        }
+        tmpvector++;
+    }
+    totMax = (WebRtc_Word16)WEBRTC_SPL_MIN(tempMax, WEBRTC_SPL_WORD16_MAX);
+    return totMax;
+}
+
+#endif
+
+// Index of maximum absolute value in a  word16 vector.
+WebRtc_Word16 WebRtcSpl_MaxAbsIndexW16(G_CONST WebRtc_Word16* vector, WebRtc_Word16 length)
+{
+    WebRtc_Word16 tempMax;
+    WebRtc_Word16 absTemp;
+    WebRtc_Word16 tempMaxIndex = 0;
+    WebRtc_Word16 i = 0;
+    G_CONST WebRtc_Word16 *tmpvector = vector;
+
+    tempMax = WEBRTC_SPL_ABS_W16(*tmpvector);
+    tmpvector++;
+    for (i = 1; i < length; i++)
+    {
+        absTemp = WEBRTC_SPL_ABS_W16(*tmpvector);
+        tmpvector++;
+        if (absTemp > tempMax)
+        {
+            tempMax = absTemp;
+            tempMaxIndex = i;
+        }
+    }
+    return tempMaxIndex;
+}
+
+// Maximum absolute value of word32 vector.
+WebRtc_Word32 WebRtcSpl_MaxAbsValueW32(G_CONST WebRtc_Word32 *vector, WebRtc_Word16 length)
+{
+    WebRtc_UWord32 tempMax = 0;
+    WebRtc_UWord32 absVal;
+    WebRtc_Word32 retval;
+    int i;
+    G_CONST WebRtc_Word32 *tmpvector = vector;
+
+    for (i = 0; i < length; i++)
+    {
+        absVal = WEBRTC_SPL_ABS_W32((*tmpvector));
+        if (absVal > tempMax)
+        {
+            tempMax = absVal;
+        }
+        tmpvector++;
+    }
+    retval = (WebRtc_Word32)(WEBRTC_SPL_MIN(tempMax, WEBRTC_SPL_WORD32_MAX));
+    return retval;
+}
+
+// Maximum value of word16 vector.
+#ifndef XSCALE_OPT
+WebRtc_Word16 WebRtcSpl_MaxValueW16(G_CONST WebRtc_Word16* vector, WebRtc_Word16 length)
+{
+    WebRtc_Word16 tempMax;
+    WebRtc_Word16 i;
+    G_CONST WebRtc_Word16 *tmpvector = vector;
+
+    tempMax = *tmpvector++;
+    for (i = 1; i < length; i++)
+    {
+        if (*tmpvector++ > tempMax)
+            tempMax = vector[i];
+    }
+    return tempMax;
+}
+#else
+#pragma message(">> WebRtcSpl_MaxValueW16 is excluded from this build")
+#endif
+
+// Index of maximum value in a word16 vector.
+WebRtc_Word16 WebRtcSpl_MaxIndexW16(G_CONST WebRtc_Word16 *vector, WebRtc_Word16 length)
+{
+    WebRtc_Word16 tempMax;
+    WebRtc_Word16 tempMaxIndex = 0;
+    WebRtc_Word16 i = 0;
+    G_CONST WebRtc_Word16 *tmpvector = vector;
+
+    tempMax = *tmpvector++;
+    for (i = 1; i < length; i++)
+    {
+        if (*tmpvector++ > tempMax)
+        {
+            tempMax = vector[i];
+            tempMaxIndex = i;
+        }
+    }
+    return tempMaxIndex;
+}
+
+// Maximum value of word32 vector.
+#ifndef XSCALE_OPT
+WebRtc_Word32 WebRtcSpl_MaxValueW32(G_CONST WebRtc_Word32* vector, WebRtc_Word16 length)
+{
+    WebRtc_Word32 tempMax;
+    WebRtc_Word16 i;
+    G_CONST WebRtc_Word32 *tmpvector = vector;
+
+    tempMax = *tmpvector++;
+    for (i = 1; i < length; i++)
+    {
+        if (*tmpvector++ > tempMax)
+            tempMax = vector[i];
+    }
+    return tempMax;
+}
+#else
+#pragma message(">> WebRtcSpl_MaxValueW32 is excluded from this build")
+#endif
+
+// Index of maximum value in a word32 vector.
+WebRtc_Word16 WebRtcSpl_MaxIndexW32(G_CONST WebRtc_Word32* vector, WebRtc_Word16 length)
+{
+    WebRtc_Word32 tempMax;
+    WebRtc_Word16 tempMaxIndex = 0;
+    WebRtc_Word16 i = 0;
+    G_CONST WebRtc_Word32 *tmpvector = vector;
+
+    tempMax = *tmpvector++;
+    for (i = 1; i < length; i++)
+    {
+        if (*tmpvector++ > tempMax)
+        {
+            tempMax = vector[i];
+            tempMaxIndex = i;
+        }
+    }
+    return tempMaxIndex;
+}
+
+// Minimum value of word16 vector.
+WebRtc_Word16 WebRtcSpl_MinValueW16(G_CONST WebRtc_Word16 *vector, WebRtc_Word16 length)
+{
+    WebRtc_Word16 tempMin;
+    WebRtc_Word16 i;
+    G_CONST WebRtc_Word16 *tmpvector = vector;
+
+    // Find the minimum value
+    tempMin = *tmpvector++;
+    for (i = 1; i < length; i++)
+    {
+        if (*tmpvector++ < tempMin)
+            tempMin = (vector[i]);
+    }
+    return tempMin;
+}
+
+// Index of minimum value in a word16 vector.
+#ifndef XSCALE_OPT
+WebRtc_Word16 WebRtcSpl_MinIndexW16(G_CONST WebRtc_Word16* vector, WebRtc_Word16 length)
+{
+    WebRtc_Word16 tempMin;
+    WebRtc_Word16 tempMinIndex = 0;
+    WebRtc_Word16 i = 0;
+    G_CONST WebRtc_Word16* tmpvector = vector;
+
+    // Find index of smallest value
+    tempMin = *tmpvector++;
+    for (i = 1; i < length; i++)
+    {
+        if (*tmpvector++ < tempMin)
+        {
+            tempMin = vector[i];
+            tempMinIndex = i;
+        }
+    }
+    return tempMinIndex;
+}
+#else
+#pragma message(">> WebRtcSpl_MinIndexW16 is excluded from this build")
+#endif
+
+// Minimum value of word32 vector.
+WebRtc_Word32 WebRtcSpl_MinValueW32(G_CONST WebRtc_Word32 *vector, WebRtc_Word16 length)
+{
+    WebRtc_Word32 tempMin;
+    WebRtc_Word16 i;
+    G_CONST WebRtc_Word32 *tmpvector = vector;
+
+    // Find the minimum value
+    tempMin = *tmpvector++;
+    for (i = 1; i < length; i++)
+    {
+        if (*tmpvector++ < tempMin)
+            tempMin = (vector[i]);
+    }
+    return tempMin;
+}
+
+// Index of minimum value in a word32 vector.
+#ifndef XSCALE_OPT
+WebRtc_Word16 WebRtcSpl_MinIndexW32(G_CONST WebRtc_Word32* vector, WebRtc_Word16 length)
+{
+    WebRtc_Word32 tempMin;
+    WebRtc_Word16 tempMinIndex = 0;
+    WebRtc_Word16 i = 0;
+    G_CONST WebRtc_Word32 *tmpvector = vector;
+
+    // Find index of smallest value
+    tempMin = *tmpvector++;
+    for (i = 1; i < length; i++)
+    {
+        if (*tmpvector++ < tempMin)
+        {
+            tempMin = vector[i];
+            tempMinIndex = i;
+        }
+    }
+    return tempMinIndex;
+}
+#else
+#pragma message(">> WebRtcSpl_MinIndexW32 is excluded from this build")
+#endif
diff --git a/trunk/src/common_audio/signal_processing/min_max_operations_neon.c b/trunk/src/common_audio/signal_processing/min_max_operations_neon.c
new file mode 100644
index 0000000..158bcc1
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/min_max_operations_neon.c
@@ -0,0 +1,47 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#if (defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM_NEON))
+
+#include <arm_neon.h>
+
+#include "signal_processing_library.h"
+
+// Maximum absolute value of word16 vector.
+WebRtc_Word16 WebRtcSpl_MaxAbsValueW16(const WebRtc_Word16* vector,
+                                       WebRtc_Word16 length) {
+  WebRtc_Word32 temp_max = 0;
+  WebRtc_Word32 abs_val;
+  WebRtc_Word16 tot_max;
+  int i;
+
+  __asm__("vmov.i16 d25, #0" : : : "d25");
+
+  for (i = 0; i < length - 7; i += 8) {
+    __asm__("vld1.16 {d26, d27}, [%0]" : : "r"(&vector[i]) : "q13");
+    __asm__("vabs.s16 q13, q13" : : : "q13");
+    __asm__("vpmax.s16 d26, d27" : : : "q13");
+    __asm__("vpmax.s16 d25, d26" : : : "d25", "d26");
+  }
+  __asm__("vpmax.s16 d25, d25" : : : "d25");
+  __asm__("vpmax.s16 d25, d25" : : : "d25");
+  __asm__("vmov.s16 %0, d25[0]" : "=r"(temp_max): : "d25");
+
+  for (; i < length; i++) {
+    abs_val = WEBRTC_SPL_ABS_W32((vector[i]));
+    if (abs_val > temp_max) {
+      temp_max = abs_val;
+    }
+  }
+  tot_max = (WebRtc_Word16)WEBRTC_SPL_MIN(temp_max, WEBRTC_SPL_WORD16_MAX);
+  return tot_max;
+}
+
+#endif
diff --git a/trunk/src/common_audio/signal_processing/randomization_functions.c b/trunk/src/common_audio/signal_processing/randomization_functions.c
new file mode 100644
index 0000000..04271ad
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/randomization_functions.c
@@ -0,0 +1,119 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains implementations of the randomization functions
+ * WebRtcSpl_IncreaseSeed()
+ * WebRtcSpl_RandU()
+ * WebRtcSpl_RandN()
+ * WebRtcSpl_RandUArray()
+ *
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "signal_processing_library.h"
+
+static const WebRtc_Word16 kRandNTable[] = {
+    9178,    -7260,       40,    10189,     4894,    -3531,   -13779,    14764,
+   -4008,    -8884,    -8990,     1008,     7368,     5184,     3251,    -5817,
+   -9786,     5963,     1770,     8066,    -7135,    10772,    -2298,     1361,
+    6484,     2241,    -8633,      792,      199,    -3344,     6553,   -10079,
+  -15040,       95,    11608,   -12469,    14161,    -4176,     2476,     6403,
+   13685,   -16005,     6646,     2239,    10916,    -3004,     -602,    -3141,
+    2142,    14144,    -5829,     5305,     8209,     4713,     2697,    -5112,
+   16092,    -1210,    -2891,    -6631,    -5360,   -11878,    -6781,    -2739,
+   -6392,      536,    10923,    10872,     5059,    -4748,    -7770,     5477,
+      38,    -1025,    -2892,     1638,     6304,    14375,   -11028,     1553,
+   -1565,    10762,     -393,     4040,     5257,    12310,     6554,    -4799,
+    4899,    -6354,     1603,    -1048,    -2220,     8247,     -186,    -8944,
+  -12004,     2332,     4801,    -4933,     6371,      131,     8614,    -5927,
+   -8287,   -22760,     4033,   -15162,     3385,     3246,     3153,    -5250,
+    3766,      784,     6494,      -62,     3531,    -1582,    15572,      662,
+   -3952,     -330,    -3196,      669,     7236,    -2678,    -6569,    23319,
+   -8645,     -741,    14830,   -15976,     4903,      315,   -11342,    10311,
+    1858,    -7777,     2145,     5436,     5677,     -113,   -10033,      826,
+   -1353,    17210,     7768,      986,    -1471,     8291,    -4982,     8207,
+  -14911,    -6255,    -2449,   -11881,    -7059,   -11703,    -4338,     8025,
+    7538,    -2823,   -12490,     9470,    -1613,    -2529,   -10092,    -7807,
+    9480,     6970,   -12844,     5123,     3532,     4816,     4803,    -8455,
+   -5045,    14032,    -4378,    -1643,     5756,   -11041,    -2732,   -16618,
+   -6430,   -18375,    -3320,     6098,     5131,    -4269,    -8840,     2482,
+   -7048,     1547,   -21890,    -6505,    -7414,     -424,   -11722,     7955,
+    1653,   -17299,     1823,      473,    -9232,     3337,     1111,      873,
+    4018,    -8982,     9889,     3531,   -11763,    -3799,     7373,    -4539,
+    3231,     7054,    -8537,     7616,     6244,    16635,      447,    -2915,
+   13967,      705,    -2669,    -1520,    -1771,   -16188,     5956,     5117,
+    6371,    -9936,    -1448,     2480,     5128,     7550,    -8130,     5236,
+    8213,    -6443,     7707,    -1950,   -13811,     7218,     7031,    -3883,
+      67,     5731,    -2874,    13480,    -3743,     9298,    -3280,     3552,
+   -4425,      -18,    -3785,    -9988,    -5357,     5477,   -11794,     2117,
+    1416,    -9935,     3376,      802,    -5079,    -8243,    12652,       66,
+    3653,    -2368,     6781,   -21895,    -7227,     2487,     7839,     -385,
+    6646,    -7016,    -4658,     5531,    -1705,      834,      129,     3694,
+   -1343,     2238,   -22640,    -6417,   -11139,    11301,    -2945,    -3494,
+   -5626,      185,    -3615,    -2041,    -7972,    -3106,      -60,   -23497,
+   -1566,    17064,     3519,     2518,      304,    -6805,   -10269,     2105,
+    1936,     -426,     -736,    -8122,    -1467,     4238,    -6939,   -13309,
+     360,     7402,    -7970,    12576,     3287,    12194,    -6289,   -16006,
+    9171,     4042,    -9193,     9123,    -2512,     6388,    -4734,    -8739,
+    1028,    -5406,    -1696,     5889,     -666,    -4736,     4971,     3565,
+    9362,    -6292,     3876,    -3652,   -19666,     7523,    -4061,      391,
+  -11773,     7502,    -3763,     4929,    -9478,    13278,     2805,     4496,
+    7814,    16419,    12455,   -14773,     2127,    -2746,     3763,     4847,
+    3698,     6978,     4751,    -6957,    -3581,      -45,     6252,     1513,
+   -4797,    -7925,    11270,    16188,    -2359,    -5269,     9376,   -10777,
+    7262,    20031,    -6515,    -2208,    -5353,     8085,    -1341,    -1303,
+    7333,     5576,     3625,     5763,    -7931,     9833,    -3371,   -10305,
+    6534,   -13539,    -9971,      997,     8464,    -4064,    -1495,     1857,
+   13624,     5458,     9490,   -11086,    -4524,    12022,     -550,     -198,
+     408,    -8455,    -7068,    10289,     9712,    -3366,     9028,    -7621,
+   -5243,     2362,     6909,     4672,    -4933,    -1799,     4709,    -4563,
+     -62,     -566,     1624,    -7010,    14730,   -17791,    -3697,    -2344,
+   -1741,     7099,    -9509,    -6855,    -1989,     3495,    -2289,     2031,
+   12784,      891,    14189,    -3963,    -5683,      421,   -12575,     1724,
+  -12682,    -5970,    -8169,     3143,    -1824,    -5488,    -5130,     8536,
+   12799,      794,     5738,     3459,   -11689,     -258,    -3738,    -3775,
+   -8742,     2333,     8312,    -9383,    10331,    13119,     8398,    10644,
+  -19433,    -6446,   -16277,   -11793,    16284,     9345,    15222,    15834,
+    2009,    -7349,      130,   -14547,      338,    -5998,     3337,    21492,
+    2406,     7703,     -951,    11196,     -564,     3406,     2217,     4806,
+    2374,    -5797,    11839,     8940,   -11874,    18213,     2855,    10492
+};
+
+WebRtc_UWord32 WebRtcSpl_IncreaseSeed(WebRtc_UWord32 *seed)
+{
+    seed[0] = (seed[0] * ((WebRtc_Word32)69069) + 1) & (WEBRTC_SPL_MAX_SEED_USED - 1);
+    return seed[0];
+}
+
+WebRtc_Word16 WebRtcSpl_RandU(WebRtc_UWord32 *seed)
+{
+    return (WebRtc_Word16)(WebRtcSpl_IncreaseSeed(seed) >> 16);
+}
+
+WebRtc_Word16 WebRtcSpl_RandN(WebRtc_UWord32 *seed)
+{
+    return kRandNTable[WebRtcSpl_IncreaseSeed(seed) >> 23];
+}
+
+// Creates an array of uniformly distributed variables
+WebRtc_Word16 WebRtcSpl_RandUArray(WebRtc_Word16* vector,
+                                   WebRtc_Word16 vector_length,
+                                   WebRtc_UWord32* seed)
+{
+    int i;
+    for (i = 0; i < vector_length; i++)
+    {
+        vector[i] = WebRtcSpl_RandU(seed);
+    }
+    return vector_length;
+}
diff --git a/trunk/src/common_audio/signal_processing/refl_coef_to_lpc.c b/trunk/src/common_audio/signal_processing/refl_coef_to_lpc.c
new file mode 100644
index 0000000..d07804d
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/refl_coef_to_lpc.c
@@ -0,0 +1,60 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains the function WebRtcSpl_ReflCoefToLpc().
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "signal_processing_library.h"
+
+void WebRtcSpl_ReflCoefToLpc(G_CONST WebRtc_Word16 *k, int use_order, WebRtc_Word16 *a)
+{
+    WebRtc_Word16 any[WEBRTC_SPL_MAX_LPC_ORDER + 1];
+    WebRtc_Word16 *aptr, *aptr2, *anyptr;
+    G_CONST WebRtc_Word16 *kptr;
+    int m, i;
+
+    kptr = k;
+    *a = 4096; // i.e., (Word16_MAX >> 3)+1.
+    *any = *a;
+    a[1] = WEBRTC_SPL_RSHIFT_W16((*k), 3);
+
+    for (m = 1; m < use_order; m++)
+    {
+        kptr++;
+        aptr = a;
+        aptr++;
+        aptr2 = &a[m];
+        anyptr = any;
+        anyptr++;
+
+        any[m + 1] = WEBRTC_SPL_RSHIFT_W16((*kptr), 3);
+        for (i = 0; i < m; i++)
+        {
+            *anyptr = (*aptr)
+                    + (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT((*aptr2), (*kptr), 15);
+            anyptr++;
+            aptr++;
+            aptr2--;
+        }
+
+        aptr = a;
+        anyptr = any;
+        for (i = 0; i < (m + 2); i++)
+        {
+            *aptr = *anyptr;
+            aptr++;
+            anyptr++;
+        }
+    }
+}
diff --git a/trunk/src/common_audio/signal_processing/resample.c b/trunk/src/common_audio/signal_processing/resample.c
new file mode 100644
index 0000000..19d1778
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/resample.c
@@ -0,0 +1,505 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains the resampling functions for 22 kHz.
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "signal_processing_library.h"
+#include "resample_by_2_internal.h"
+
+// Declaration of internally used functions
+static void WebRtcSpl_32khzTo22khzIntToShort(const WebRtc_Word32 *In, WebRtc_Word16 *Out,
+                                             const WebRtc_Word32 K);
+
+void WebRtcSpl_32khzTo22khzIntToInt(const WebRtc_Word32 *In, WebRtc_Word32 *Out,
+                                    const WebRtc_Word32 K);
+
+// interpolation coefficients
+static const WebRtc_Word16 kCoefficients32To22[5][9] = {
+        {127, -712,  2359, -6333, 23456, 16775, -3695,  945, -154},
+        {-39,  230,  -830,  2785, 32366, -2324,   760, -218,   38},
+        {117, -663,  2222, -6133, 26634, 13070, -3174,  831, -137},
+        {-77,  457, -1677,  5958, 31175, -4136,  1405, -408,   71},
+        { 98, -560,  1900, -5406, 29240,  9423, -2480,  663, -110}
+};
+
+//////////////////////
+// 22 kHz -> 16 kHz //
+//////////////////////
+
+// number of subblocks; options: 1, 2, 4, 5, 10
+#define SUB_BLOCKS_22_16    5
+
+// 22 -> 16 resampler
+void WebRtcSpl_Resample22khzTo16khz(const WebRtc_Word16* in, WebRtc_Word16* out,
+                                    WebRtcSpl_State22khzTo16khz* state, WebRtc_Word32* tmpmem)
+{
+    int k;
+
+    // process two blocks of 10/SUB_BLOCKS_22_16 ms (to reduce temp buffer size)
+    for (k = 0; k < SUB_BLOCKS_22_16; k++)
+    {
+        ///// 22 --> 44 /////
+        // WebRtc_Word16  in[220/SUB_BLOCKS_22_16]
+        // WebRtc_Word32 out[440/SUB_BLOCKS_22_16]
+        /////
+        WebRtcSpl_UpBy2ShortToInt(in, 220 / SUB_BLOCKS_22_16, tmpmem + 16, state->S_22_44);
+
+        ///// 44 --> 32 /////
+        // WebRtc_Word32  in[440/SUB_BLOCKS_22_16]
+        // WebRtc_Word32 out[320/SUB_BLOCKS_22_16]
+        /////
+        // copy state to and from input array
+        tmpmem[8] = state->S_44_32[0];
+        tmpmem[9] = state->S_44_32[1];
+        tmpmem[10] = state->S_44_32[2];
+        tmpmem[11] = state->S_44_32[3];
+        tmpmem[12] = state->S_44_32[4];
+        tmpmem[13] = state->S_44_32[5];
+        tmpmem[14] = state->S_44_32[6];
+        tmpmem[15] = state->S_44_32[7];
+        state->S_44_32[0] = tmpmem[440 / SUB_BLOCKS_22_16 + 8];
+        state->S_44_32[1] = tmpmem[440 / SUB_BLOCKS_22_16 + 9];
+        state->S_44_32[2] = tmpmem[440 / SUB_BLOCKS_22_16 + 10];
+        state->S_44_32[3] = tmpmem[440 / SUB_BLOCKS_22_16 + 11];
+        state->S_44_32[4] = tmpmem[440 / SUB_BLOCKS_22_16 + 12];
+        state->S_44_32[5] = tmpmem[440 / SUB_BLOCKS_22_16 + 13];
+        state->S_44_32[6] = tmpmem[440 / SUB_BLOCKS_22_16 + 14];
+        state->S_44_32[7] = tmpmem[440 / SUB_BLOCKS_22_16 + 15];
+
+        WebRtcSpl_Resample44khzTo32khz(tmpmem + 8, tmpmem, 40 / SUB_BLOCKS_22_16);
+
+        ///// 32 --> 16 /////
+        // WebRtc_Word32  in[320/SUB_BLOCKS_22_16]
+        // WebRtc_Word32 out[160/SUB_BLOCKS_22_16]
+        /////
+        WebRtcSpl_DownBy2IntToShort(tmpmem, 320 / SUB_BLOCKS_22_16, out, state->S_32_16);
+
+        // move input/output pointers 10/SUB_BLOCKS_22_16 ms seconds ahead
+        in += 220 / SUB_BLOCKS_22_16;
+        out += 160 / SUB_BLOCKS_22_16;
+    }
+}
+
+// initialize state of 22 -> 16 resampler
+void WebRtcSpl_ResetResample22khzTo16khz(WebRtcSpl_State22khzTo16khz* state)
+{
+    int k;
+    for (k = 0; k < 8; k++)
+    {
+        state->S_22_44[k] = 0;
+        state->S_44_32[k] = 0;
+        state->S_32_16[k] = 0;
+    }
+}
+
+//////////////////////
+// 16 kHz -> 22 kHz //
+//////////////////////
+
+// number of subblocks; options: 1, 2, 4, 5, 10
+#define SUB_BLOCKS_16_22    4
+
+// 16 -> 22 resampler
+void WebRtcSpl_Resample16khzTo22khz(const WebRtc_Word16* in, WebRtc_Word16* out,
+                                    WebRtcSpl_State16khzTo22khz* state, WebRtc_Word32* tmpmem)
+{
+    int k;
+
+    // process two blocks of 10/SUB_BLOCKS_16_22 ms (to reduce temp buffer size)
+    for (k = 0; k < SUB_BLOCKS_16_22; k++)
+    {
+        ///// 16 --> 32 /////
+        // WebRtc_Word16  in[160/SUB_BLOCKS_16_22]
+        // WebRtc_Word32 out[320/SUB_BLOCKS_16_22]
+        /////
+        WebRtcSpl_UpBy2ShortToInt(in, 160 / SUB_BLOCKS_16_22, tmpmem + 8, state->S_16_32);
+
+        ///// 32 --> 22 /////
+        // WebRtc_Word32  in[320/SUB_BLOCKS_16_22]
+        // WebRtc_Word32 out[220/SUB_BLOCKS_16_22]
+        /////
+        // copy state to and from input array
+        tmpmem[0] = state->S_32_22[0];
+        tmpmem[1] = state->S_32_22[1];
+        tmpmem[2] = state->S_32_22[2];
+        tmpmem[3] = state->S_32_22[3];
+        tmpmem[4] = state->S_32_22[4];
+        tmpmem[5] = state->S_32_22[5];
+        tmpmem[6] = state->S_32_22[6];
+        tmpmem[7] = state->S_32_22[7];
+        state->S_32_22[0] = tmpmem[320 / SUB_BLOCKS_16_22];
+        state->S_32_22[1] = tmpmem[320 / SUB_BLOCKS_16_22 + 1];
+        state->S_32_22[2] = tmpmem[320 / SUB_BLOCKS_16_22 + 2];
+        state->S_32_22[3] = tmpmem[320 / SUB_BLOCKS_16_22 + 3];
+        state->S_32_22[4] = tmpmem[320 / SUB_BLOCKS_16_22 + 4];
+        state->S_32_22[5] = tmpmem[320 / SUB_BLOCKS_16_22 + 5];
+        state->S_32_22[6] = tmpmem[320 / SUB_BLOCKS_16_22 + 6];
+        state->S_32_22[7] = tmpmem[320 / SUB_BLOCKS_16_22 + 7];
+
+        WebRtcSpl_32khzTo22khzIntToShort(tmpmem, out, 20 / SUB_BLOCKS_16_22);
+
+        // move input/output pointers 10/SUB_BLOCKS_16_22 ms seconds ahead
+        in += 160 / SUB_BLOCKS_16_22;
+        out += 220 / SUB_BLOCKS_16_22;
+    }
+}
+
+// initialize state of 16 -> 22 resampler
+void WebRtcSpl_ResetResample16khzTo22khz(WebRtcSpl_State16khzTo22khz* state)
+{
+    int k;
+    for (k = 0; k < 8; k++)
+    {
+        state->S_16_32[k] = 0;
+        state->S_32_22[k] = 0;
+    }
+}
+
+//////////////////////
+// 22 kHz ->  8 kHz //
+//////////////////////
+
+// number of subblocks; options: 1, 2, 5, 10
+#define SUB_BLOCKS_22_8     2
+
+// 22 -> 8 resampler
+void WebRtcSpl_Resample22khzTo8khz(const WebRtc_Word16* in, WebRtc_Word16* out,
+                                   WebRtcSpl_State22khzTo8khz* state, WebRtc_Word32* tmpmem)
+{
+    int k;
+
+    // process two blocks of 10/SUB_BLOCKS_22_8 ms (to reduce temp buffer size)
+    for (k = 0; k < SUB_BLOCKS_22_8; k++)
+    {
+        ///// 22 --> 22 lowpass /////
+        // WebRtc_Word16  in[220/SUB_BLOCKS_22_8]
+        // WebRtc_Word32 out[220/SUB_BLOCKS_22_8]
+        /////
+        WebRtcSpl_LPBy2ShortToInt(in, 220 / SUB_BLOCKS_22_8, tmpmem + 16, state->S_22_22);
+
+        ///// 22 --> 16 /////
+        // WebRtc_Word32  in[220/SUB_BLOCKS_22_8]
+        // WebRtc_Word32 out[160/SUB_BLOCKS_22_8]
+        /////
+        // copy state to and from input array
+        tmpmem[8] = state->S_22_16[0];
+        tmpmem[9] = state->S_22_16[1];
+        tmpmem[10] = state->S_22_16[2];
+        tmpmem[11] = state->S_22_16[3];
+        tmpmem[12] = state->S_22_16[4];
+        tmpmem[13] = state->S_22_16[5];
+        tmpmem[14] = state->S_22_16[6];
+        tmpmem[15] = state->S_22_16[7];
+        state->S_22_16[0] = tmpmem[220 / SUB_BLOCKS_22_8 + 8];
+        state->S_22_16[1] = tmpmem[220 / SUB_BLOCKS_22_8 + 9];
+        state->S_22_16[2] = tmpmem[220 / SUB_BLOCKS_22_8 + 10];
+        state->S_22_16[3] = tmpmem[220 / SUB_BLOCKS_22_8 + 11];
+        state->S_22_16[4] = tmpmem[220 / SUB_BLOCKS_22_8 + 12];
+        state->S_22_16[5] = tmpmem[220 / SUB_BLOCKS_22_8 + 13];
+        state->S_22_16[6] = tmpmem[220 / SUB_BLOCKS_22_8 + 14];
+        state->S_22_16[7] = tmpmem[220 / SUB_BLOCKS_22_8 + 15];
+
+        WebRtcSpl_Resample44khzTo32khz(tmpmem + 8, tmpmem, 20 / SUB_BLOCKS_22_8);
+
+        ///// 16 --> 8 /////
+        // WebRtc_Word32 in[160/SUB_BLOCKS_22_8]
+        // WebRtc_Word32 out[80/SUB_BLOCKS_22_8]
+        /////
+        WebRtcSpl_DownBy2IntToShort(tmpmem, 160 / SUB_BLOCKS_22_8, out, state->S_16_8);
+
+        // move input/output pointers 10/SUB_BLOCKS_22_8 ms seconds ahead
+        in += 220 / SUB_BLOCKS_22_8;
+        out += 80 / SUB_BLOCKS_22_8;
+    }
+}
+
+// initialize state of 22 -> 8 resampler
+void WebRtcSpl_ResetResample22khzTo8khz(WebRtcSpl_State22khzTo8khz* state)
+{
+    int k;
+    for (k = 0; k < 8; k++)
+    {
+        state->S_22_22[k] = 0;
+        state->S_22_22[k + 8] = 0;
+        state->S_22_16[k] = 0;
+        state->S_16_8[k] = 0;
+    }
+}
+
+//////////////////////
+//  8 kHz -> 22 kHz //
+//////////////////////
+
+// number of subblocks; options: 1, 2, 5, 10
+#define SUB_BLOCKS_8_22     2
+
+// 8 -> 22 resampler
+void WebRtcSpl_Resample8khzTo22khz(const WebRtc_Word16* in, WebRtc_Word16* out,
+                                   WebRtcSpl_State8khzTo22khz* state, WebRtc_Word32* tmpmem)
+{
+    int k;
+
+    // process two blocks of 10/SUB_BLOCKS_8_22 ms (to reduce temp buffer size)
+    for (k = 0; k < SUB_BLOCKS_8_22; k++)
+    {
+        ///// 8 --> 16 /////
+        // WebRtc_Word16  in[80/SUB_BLOCKS_8_22]
+        // WebRtc_Word32 out[160/SUB_BLOCKS_8_22]
+        /////
+        WebRtcSpl_UpBy2ShortToInt(in, 80 / SUB_BLOCKS_8_22, tmpmem + 18, state->S_8_16);
+
+        ///// 16 --> 11 /////
+        // WebRtc_Word32  in[160/SUB_BLOCKS_8_22]
+        // WebRtc_Word32 out[110/SUB_BLOCKS_8_22]
+        /////
+        // copy state to and from input array
+        tmpmem[10] = state->S_16_11[0];
+        tmpmem[11] = state->S_16_11[1];
+        tmpmem[12] = state->S_16_11[2];
+        tmpmem[13] = state->S_16_11[3];
+        tmpmem[14] = state->S_16_11[4];
+        tmpmem[15] = state->S_16_11[5];
+        tmpmem[16] = state->S_16_11[6];
+        tmpmem[17] = state->S_16_11[7];
+        state->S_16_11[0] = tmpmem[160 / SUB_BLOCKS_8_22 + 10];
+        state->S_16_11[1] = tmpmem[160 / SUB_BLOCKS_8_22 + 11];
+        state->S_16_11[2] = tmpmem[160 / SUB_BLOCKS_8_22 + 12];
+        state->S_16_11[3] = tmpmem[160 / SUB_BLOCKS_8_22 + 13];
+        state->S_16_11[4] = tmpmem[160 / SUB_BLOCKS_8_22 + 14];
+        state->S_16_11[5] = tmpmem[160 / SUB_BLOCKS_8_22 + 15];
+        state->S_16_11[6] = tmpmem[160 / SUB_BLOCKS_8_22 + 16];
+        state->S_16_11[7] = tmpmem[160 / SUB_BLOCKS_8_22 + 17];
+
+        WebRtcSpl_32khzTo22khzIntToInt(tmpmem + 10, tmpmem, 10 / SUB_BLOCKS_8_22);
+
+        ///// 11 --> 22 /////
+        // WebRtc_Word32  in[110/SUB_BLOCKS_8_22]
+        // WebRtc_Word16 out[220/SUB_BLOCKS_8_22]
+        /////
+        WebRtcSpl_UpBy2IntToShort(tmpmem, 110 / SUB_BLOCKS_8_22, out, state->S_11_22);
+
+        // move input/output pointers 10/SUB_BLOCKS_8_22 ms seconds ahead
+        in += 80 / SUB_BLOCKS_8_22;
+        out += 220 / SUB_BLOCKS_8_22;
+    }
+}
+
+// initialize state of 8 -> 22 resampler
+void WebRtcSpl_ResetResample8khzTo22khz(WebRtcSpl_State8khzTo22khz* state)
+{
+    int k;
+    for (k = 0; k < 8; k++)
+    {
+        state->S_8_16[k] = 0;
+        state->S_16_11[k] = 0;
+        state->S_11_22[k] = 0;
+    }
+}
+
+// compute two inner-products and store them to output array
+static void WebRtcSpl_DotProdIntToInt(const WebRtc_Word32* in1, const WebRtc_Word32* in2,
+                                      const WebRtc_Word16* coef_ptr, WebRtc_Word32* out1,
+                                      WebRtc_Word32* out2)
+{
+    WebRtc_Word32 tmp1 = 16384;
+    WebRtc_Word32 tmp2 = 16384;
+    WebRtc_Word16 coef;
+
+    coef = coef_ptr[0];
+    tmp1 += coef * in1[0];
+    tmp2 += coef * in2[-0];
+
+    coef = coef_ptr[1];
+    tmp1 += coef * in1[1];
+    tmp2 += coef * in2[-1];
+
+    coef = coef_ptr[2];
+    tmp1 += coef * in1[2];
+    tmp2 += coef * in2[-2];
+
+    coef = coef_ptr[3];
+    tmp1 += coef * in1[3];
+    tmp2 += coef * in2[-3];
+
+    coef = coef_ptr[4];
+    tmp1 += coef * in1[4];
+    tmp2 += coef * in2[-4];
+
+    coef = coef_ptr[5];
+    tmp1 += coef * in1[5];
+    tmp2 += coef * in2[-5];
+
+    coef = coef_ptr[6];
+    tmp1 += coef * in1[6];
+    tmp2 += coef * in2[-6];
+
+    coef = coef_ptr[7];
+    tmp1 += coef * in1[7];
+    tmp2 += coef * in2[-7];
+
+    coef = coef_ptr[8];
+    *out1 = tmp1 + coef * in1[8];
+    *out2 = tmp2 + coef * in2[-8];
+}
+
+// compute two inner-products and store them to output array
+static void WebRtcSpl_DotProdIntToShort(const WebRtc_Word32* in1, const WebRtc_Word32* in2,
+                                        const WebRtc_Word16* coef_ptr, WebRtc_Word16* out1,
+                                        WebRtc_Word16* out2)
+{
+    WebRtc_Word32 tmp1 = 16384;
+    WebRtc_Word32 tmp2 = 16384;
+    WebRtc_Word16 coef;
+
+    coef = coef_ptr[0];
+    tmp1 += coef * in1[0];
+    tmp2 += coef * in2[-0];
+
+    coef = coef_ptr[1];
+    tmp1 += coef * in1[1];
+    tmp2 += coef * in2[-1];
+
+    coef = coef_ptr[2];
+    tmp1 += coef * in1[2];
+    tmp2 += coef * in2[-2];
+
+    coef = coef_ptr[3];
+    tmp1 += coef * in1[3];
+    tmp2 += coef * in2[-3];
+
+    coef = coef_ptr[4];
+    tmp1 += coef * in1[4];
+    tmp2 += coef * in2[-4];
+
+    coef = coef_ptr[5];
+    tmp1 += coef * in1[5];
+    tmp2 += coef * in2[-5];
+
+    coef = coef_ptr[6];
+    tmp1 += coef * in1[6];
+    tmp2 += coef * in2[-6];
+
+    coef = coef_ptr[7];
+    tmp1 += coef * in1[7];
+    tmp2 += coef * in2[-7];
+
+    coef = coef_ptr[8];
+    tmp1 += coef * in1[8];
+    tmp2 += coef * in2[-8];
+
+    // scale down, round and saturate
+    tmp1 >>= 15;
+    if (tmp1 > (WebRtc_Word32)0x00007FFF)
+        tmp1 = 0x00007FFF;
+    if (tmp1 < (WebRtc_Word32)0xFFFF8000)
+        tmp1 = 0xFFFF8000;
+    tmp2 >>= 15;
+    if (tmp2 > (WebRtc_Word32)0x00007FFF)
+        tmp2 = 0x00007FFF;
+    if (tmp2 < (WebRtc_Word32)0xFFFF8000)
+        tmp2 = 0xFFFF8000;
+    *out1 = (WebRtc_Word16)tmp1;
+    *out2 = (WebRtc_Word16)tmp2;
+}
+
+//   Resampling ratio: 11/16
+// input:  WebRtc_Word32 (normalized, not saturated) :: size 16 * K
+// output: WebRtc_Word32 (shifted 15 positions to the left, + offset 16384) :: size 11 * K
+//      K: Number of blocks
+
+void WebRtcSpl_32khzTo22khzIntToInt(const WebRtc_Word32* In,
+                                    WebRtc_Word32* Out,
+                                    const WebRtc_Word32 K)
+{
+    /////////////////////////////////////////////////////////////
+    // Filter operation:
+    //
+    // Perform resampling (16 input samples -> 11 output samples);
+    // process in sub blocks of size 16 samples.
+    WebRtc_Word32 m;
+
+    for (m = 0; m < K; m++)
+    {
+        // first output sample
+        Out[0] = ((WebRtc_Word32)In[3] << 15) + (1 << 14);
+
+        // sum and accumulate filter coefficients and input samples
+        WebRtcSpl_DotProdIntToInt(&In[0], &In[22], kCoefficients32To22[0], &Out[1], &Out[10]);
+
+        // sum and accumulate filter coefficients and input samples
+        WebRtcSpl_DotProdIntToInt(&In[2], &In[20], kCoefficients32To22[1], &Out[2], &Out[9]);
+
+        // sum and accumulate filter coefficients and input samples
+        WebRtcSpl_DotProdIntToInt(&In[3], &In[19], kCoefficients32To22[2], &Out[3], &Out[8]);
+
+        // sum and accumulate filter coefficients and input samples
+        WebRtcSpl_DotProdIntToInt(&In[5], &In[17], kCoefficients32To22[3], &Out[4], &Out[7]);
+
+        // sum and accumulate filter coefficients and input samples
+        WebRtcSpl_DotProdIntToInt(&In[6], &In[16], kCoefficients32To22[4], &Out[5], &Out[6]);
+
+        // update pointers
+        In += 16;
+        Out += 11;
+    }
+}
+
+//   Resampling ratio: 11/16
+// input:  WebRtc_Word32 (normalized, not saturated) :: size 16 * K
+// output: WebRtc_Word16 (saturated) :: size 11 * K
+//      K: Number of blocks
+
+void WebRtcSpl_32khzTo22khzIntToShort(const WebRtc_Word32 *In,
+                                      WebRtc_Word16 *Out,
+                                      const WebRtc_Word32 K)
+{
+    /////////////////////////////////////////////////////////////
+    // Filter operation:
+    //
+    // Perform resampling (16 input samples -> 11 output samples);
+    // process in sub blocks of size 16 samples.
+    WebRtc_Word32 tmp;
+    WebRtc_Word32 m;
+
+    for (m = 0; m < K; m++)
+    {
+        // first output sample
+        tmp = In[3];
+        if (tmp > (WebRtc_Word32)0x00007FFF)
+            tmp = 0x00007FFF;
+        if (tmp < (WebRtc_Word32)0xFFFF8000)
+            tmp = 0xFFFF8000;
+        Out[0] = (WebRtc_Word16)tmp;
+
+        // sum and accumulate filter coefficients and input samples
+        WebRtcSpl_DotProdIntToShort(&In[0], &In[22], kCoefficients32To22[0], &Out[1], &Out[10]);
+
+        // sum and accumulate filter coefficients and input samples
+        WebRtcSpl_DotProdIntToShort(&In[2], &In[20], kCoefficients32To22[1], &Out[2], &Out[9]);
+
+        // sum and accumulate filter coefficients and input samples
+        WebRtcSpl_DotProdIntToShort(&In[3], &In[19], kCoefficients32To22[2], &Out[3], &Out[8]);
+
+        // sum and accumulate filter coefficients and input samples
+        WebRtcSpl_DotProdIntToShort(&In[5], &In[17], kCoefficients32To22[3], &Out[4], &Out[7]);
+
+        // sum and accumulate filter coefficients and input samples
+        WebRtcSpl_DotProdIntToShort(&In[6], &In[16], kCoefficients32To22[4], &Out[5], &Out[6]);
+
+        // update pointers
+        In += 16;
+        Out += 11;
+    }
+}
diff --git a/trunk/src/common_audio/signal_processing/resample_48khz.c b/trunk/src/common_audio/signal_processing/resample_48khz.c
new file mode 100644
index 0000000..31cbe6b
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/resample_48khz.c
@@ -0,0 +1,186 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains resampling functions between 48 kHz and nb/wb.
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include <string.h>
+#include "signal_processing_library.h"
+#include "resample_by_2_internal.h"
+
+////////////////////////////
+///// 48 kHz -> 16 kHz /////
+////////////////////////////
+
+// 48 -> 16 resampler
+void WebRtcSpl_Resample48khzTo16khz(const WebRtc_Word16* in, WebRtc_Word16* out,
+                                    WebRtcSpl_State48khzTo16khz* state, WebRtc_Word32* tmpmem)
+{
+    ///// 48 --> 48(LP) /////
+    // WebRtc_Word16  in[480]
+    // WebRtc_Word32 out[480]
+    /////
+    WebRtcSpl_LPBy2ShortToInt(in, 480, tmpmem + 16, state->S_48_48);
+
+    ///// 48 --> 32 /////
+    // WebRtc_Word32  in[480]
+    // WebRtc_Word32 out[320]
+    /////
+    // copy state to and from input array
+    memcpy(tmpmem + 8, state->S_48_32, 8 * sizeof(WebRtc_Word32));
+    memcpy(state->S_48_32, tmpmem + 488, 8 * sizeof(WebRtc_Word32));
+    WebRtcSpl_Resample48khzTo32khz(tmpmem + 8, tmpmem, 160);
+
+    ///// 32 --> 16 /////
+    // WebRtc_Word32  in[320]
+    // WebRtc_Word16 out[160]
+    /////
+    WebRtcSpl_DownBy2IntToShort(tmpmem, 320, out, state->S_32_16);
+}
+
+// initialize state of 48 -> 16 resampler
+void WebRtcSpl_ResetResample48khzTo16khz(WebRtcSpl_State48khzTo16khz* state)
+{
+    memset(state->S_48_48, 0, 16 * sizeof(WebRtc_Word32));
+    memset(state->S_48_32, 0, 8 * sizeof(WebRtc_Word32));
+    memset(state->S_32_16, 0, 8 * sizeof(WebRtc_Word32));
+}
+
+////////////////////////////
+///// 16 kHz -> 48 kHz /////
+////////////////////////////
+
+// 16 -> 48 resampler
+void WebRtcSpl_Resample16khzTo48khz(const WebRtc_Word16* in, WebRtc_Word16* out,
+                                    WebRtcSpl_State16khzTo48khz* state, WebRtc_Word32* tmpmem)
+{
+    ///// 16 --> 32 /////
+    // WebRtc_Word16  in[160]
+    // WebRtc_Word32 out[320]
+    /////
+    WebRtcSpl_UpBy2ShortToInt(in, 160, tmpmem + 16, state->S_16_32);
+
+    ///// 32 --> 24 /////
+    // WebRtc_Word32  in[320]
+    // WebRtc_Word32 out[240]
+    // copy state to and from input array
+    /////
+    memcpy(tmpmem + 8, state->S_32_24, 8 * sizeof(WebRtc_Word32));
+    memcpy(state->S_32_24, tmpmem + 328, 8 * sizeof(WebRtc_Word32));
+    WebRtcSpl_Resample32khzTo24khz(tmpmem + 8, tmpmem, 80);
+
+    ///// 24 --> 48 /////
+    // WebRtc_Word32  in[240]
+    // WebRtc_Word16 out[480]
+    /////
+    WebRtcSpl_UpBy2IntToShort(tmpmem, 240, out, state->S_24_48);
+}
+
+// initialize state of 16 -> 48 resampler
+void WebRtcSpl_ResetResample16khzTo48khz(WebRtcSpl_State16khzTo48khz* state)
+{
+    memset(state->S_16_32, 0, 8 * sizeof(WebRtc_Word32));
+    memset(state->S_32_24, 0, 8 * sizeof(WebRtc_Word32));
+    memset(state->S_24_48, 0, 8 * sizeof(WebRtc_Word32));
+}
+
+////////////////////////////
+///// 48 kHz ->  8 kHz /////
+////////////////////////////
+
+// 48 -> 8 resampler
+void WebRtcSpl_Resample48khzTo8khz(const WebRtc_Word16* in, WebRtc_Word16* out,
+                                   WebRtcSpl_State48khzTo8khz* state, WebRtc_Word32* tmpmem)
+{
+    ///// 48 --> 24 /////
+    // WebRtc_Word16  in[480]
+    // WebRtc_Word32 out[240]
+    /////
+    WebRtcSpl_DownBy2ShortToInt(in, 480, tmpmem + 256, state->S_48_24);
+
+    ///// 24 --> 24(LP) /////
+    // WebRtc_Word32  in[240]
+    // WebRtc_Word32 out[240]
+    /////
+    WebRtcSpl_LPBy2IntToInt(tmpmem + 256, 240, tmpmem + 16, state->S_24_24);
+
+    ///// 24 --> 16 /////
+    // WebRtc_Word32  in[240]
+    // WebRtc_Word32 out[160]
+    /////
+    // copy state to and from input array
+    memcpy(tmpmem + 8, state->S_24_16, 8 * sizeof(WebRtc_Word32));
+    memcpy(state->S_24_16, tmpmem + 248, 8 * sizeof(WebRtc_Word32));
+    WebRtcSpl_Resample48khzTo32khz(tmpmem + 8, tmpmem, 80);
+
+    ///// 16 --> 8 /////
+    // WebRtc_Word32  in[160]
+    // WebRtc_Word16 out[80]
+    /////
+    WebRtcSpl_DownBy2IntToShort(tmpmem, 160, out, state->S_16_8);
+}
+
+// initialize state of 48 -> 8 resampler
+void WebRtcSpl_ResetResample48khzTo8khz(WebRtcSpl_State48khzTo8khz* state)
+{
+    memset(state->S_48_24, 0, 8 * sizeof(WebRtc_Word32));
+    memset(state->S_24_24, 0, 16 * sizeof(WebRtc_Word32));
+    memset(state->S_24_16, 0, 8 * sizeof(WebRtc_Word32));
+    memset(state->S_16_8, 0, 8 * sizeof(WebRtc_Word32));
+}
+
+////////////////////////////
+/////  8 kHz -> 48 kHz /////
+////////////////////////////
+
+// 8 -> 48 resampler
+void WebRtcSpl_Resample8khzTo48khz(const WebRtc_Word16* in, WebRtc_Word16* out,
+                                   WebRtcSpl_State8khzTo48khz* state, WebRtc_Word32* tmpmem)
+{
+    ///// 8 --> 16 /////
+    // WebRtc_Word16  in[80]
+    // WebRtc_Word32 out[160]
+    /////
+    WebRtcSpl_UpBy2ShortToInt(in, 80, tmpmem + 264, state->S_8_16);
+
+    ///// 16 --> 12 /////
+    // WebRtc_Word32  in[160]
+    // WebRtc_Word32 out[120]
+    /////
+    // copy state to and from input array
+    memcpy(tmpmem + 256, state->S_16_12, 8 * sizeof(WebRtc_Word32));
+    memcpy(state->S_16_12, tmpmem + 416, 8 * sizeof(WebRtc_Word32));
+    WebRtcSpl_Resample32khzTo24khz(tmpmem + 256, tmpmem + 240, 40);
+
+    ///// 12 --> 24 /////
+    // WebRtc_Word32  in[120]
+    // WebRtc_Word16 out[240]
+    /////
+    WebRtcSpl_UpBy2IntToInt(tmpmem + 240, 120, tmpmem, state->S_12_24);
+
+    ///// 24 --> 48 /////
+    // WebRtc_Word32  in[240]
+    // WebRtc_Word16 out[480]
+    /////
+    WebRtcSpl_UpBy2IntToShort(tmpmem, 240, out, state->S_24_48);
+}
+
+// initialize state of 8 -> 48 resampler
+void WebRtcSpl_ResetResample8khzTo48khz(WebRtcSpl_State8khzTo48khz* state)
+{
+    memset(state->S_8_16, 0, 8 * sizeof(WebRtc_Word32));
+    memset(state->S_16_12, 0, 8 * sizeof(WebRtc_Word32));
+    memset(state->S_12_24, 0, 8 * sizeof(WebRtc_Word32));
+    memset(state->S_24_48, 0, 8 * sizeof(WebRtc_Word32));
+}
diff --git a/trunk/src/common_audio/signal_processing/resample_by_2.c b/trunk/src/common_audio/signal_processing/resample_by_2.c
new file mode 100644
index 0000000..e239db7
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/resample_by_2.c
@@ -0,0 +1,181 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains the resampling by two functions.
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "signal_processing_library.h"
+
+#ifdef WEBRTC_ARCH_ARM_V7A
+
+// allpass filter coefficients.
+static const WebRtc_UWord32 kResampleAllpass1[3] = {3284, 24441, 49528 << 15};
+static const WebRtc_UWord32 kResampleAllpass2[3] =
+  {12199, 37471 << 15, 60255 << 15};
+
+// Multiply two 32-bit values and accumulate to another input value.
+// Return: state + ((diff * tbl_value) >> 16)
+
+static __inline WebRtc_Word32 MUL_ACCUM_1(WebRtc_Word32 tbl_value,
+                                          WebRtc_Word32 diff,
+                                          WebRtc_Word32 state) {
+  WebRtc_Word32 result;
+  __asm__("smlawb %r0, %r1, %r2, %r3": "=r"(result): "r"(diff),
+                                       "r"(tbl_value), "r"(state));
+  return result;
+}
+
+// Multiply two 32-bit values and accumulate to another input value.
+// Return: Return: state + (((diff << 1) * tbl_value) >> 32)
+//
+// The reason to introduce this function is that, in case we can't use smlawb
+// instruction (in MUL_ACCUM_1) due to input value range, we can still use 
+// smmla to save some cycles.
+
+static __inline WebRtc_Word32 MUL_ACCUM_2(WebRtc_Word32 tbl_value,
+                                          WebRtc_Word32 diff,
+                                          WebRtc_Word32 state) {
+  WebRtc_Word32 result;
+  __asm__("smmla %r0, %r1, %r2, %r3": "=r"(result): "r"(diff << 1),
+                                      "r"(tbl_value), "r"(state));
+  return result;
+}
+
+#else
+
+// allpass filter coefficients.
+static const WebRtc_UWord16 kResampleAllpass1[3] = {3284, 24441, 49528};
+static const WebRtc_UWord16 kResampleAllpass2[3] = {12199, 37471, 60255};
+
+// Multiply a 32-bit value with a 16-bit value and accumulate to another input:
+#define MUL_ACCUM_1(a, b, c) WEBRTC_SPL_SCALEDIFF32(a, b, c)
+#define MUL_ACCUM_2(a, b, c) WEBRTC_SPL_SCALEDIFF32(a, b, c)
+
+#endif  // WEBRTC_ARCH_ARM_V7A
+
+
+// decimator
+void WebRtcSpl_DownsampleBy2(const WebRtc_Word16* in, const WebRtc_Word16 len,
+                             WebRtc_Word16* out, WebRtc_Word32* filtState) {
+  WebRtc_Word32 tmp1, tmp2, diff, in32, out32;
+  WebRtc_Word16 i;
+
+  register WebRtc_Word32 state0 = filtState[0];
+  register WebRtc_Word32 state1 = filtState[1];
+  register WebRtc_Word32 state2 = filtState[2];
+  register WebRtc_Word32 state3 = filtState[3];
+  register WebRtc_Word32 state4 = filtState[4];
+  register WebRtc_Word32 state5 = filtState[5];
+  register WebRtc_Word32 state6 = filtState[6];
+  register WebRtc_Word32 state7 = filtState[7];
+
+  for (i = (len >> 1); i > 0; i--) {
+    // lower allpass filter
+    in32 = (WebRtc_Word32)(*in++) << 10;
+    diff = in32 - state1;
+    tmp1 = MUL_ACCUM_1(kResampleAllpass2[0], diff, state0);
+    state0 = in32;
+    diff = tmp1 - state2;
+    tmp2 = MUL_ACCUM_2(kResampleAllpass2[1], diff, state1);
+    state1 = tmp1;
+    diff = tmp2 - state3;
+    state3 = MUL_ACCUM_2(kResampleAllpass2[2], diff, state2);
+    state2 = tmp2;
+
+    // upper allpass filter
+    in32 = (WebRtc_Word32)(*in++) << 10;
+    diff = in32 - state5;
+    tmp1 = MUL_ACCUM_1(kResampleAllpass1[0], diff, state4);
+    state4 = in32;
+    diff = tmp1 - state6;
+    tmp2 = MUL_ACCUM_1(kResampleAllpass1[1], diff, state5);
+    state5 = tmp1;
+    diff = tmp2 - state7;
+    state7 = MUL_ACCUM_2(kResampleAllpass1[2], diff, state6);
+    state6 = tmp2;
+
+    // add two allpass outputs, divide by two and round
+    out32 = (state3 + state7 + 1024) >> 11;
+
+    // limit amplitude to prevent wrap-around, and write to output array
+    *out++ = WebRtcSpl_SatW32ToW16(out32);
+  }
+
+  filtState[0] = state0;
+  filtState[1] = state1;
+  filtState[2] = state2;
+  filtState[3] = state3;
+  filtState[4] = state4;
+  filtState[5] = state5;
+  filtState[6] = state6;
+  filtState[7] = state7;
+}
+
+
+void WebRtcSpl_UpsampleBy2(const WebRtc_Word16* in, WebRtc_Word16 len,
+                           WebRtc_Word16* out, WebRtc_Word32* filtState) {
+  WebRtc_Word32 tmp1, tmp2, diff, in32, out32;
+  WebRtc_Word16 i;
+
+  register WebRtc_Word32 state0 = filtState[0];
+  register WebRtc_Word32 state1 = filtState[1];
+  register WebRtc_Word32 state2 = filtState[2];
+  register WebRtc_Word32 state3 = filtState[3];
+  register WebRtc_Word32 state4 = filtState[4];
+  register WebRtc_Word32 state5 = filtState[5];
+  register WebRtc_Word32 state6 = filtState[6];
+  register WebRtc_Word32 state7 = filtState[7];
+
+  for (i = len; i > 0; i--) {
+    // lower allpass filter
+    in32 = (WebRtc_Word32)(*in++) << 10;
+    diff = in32 - state1;
+    tmp1 = MUL_ACCUM_1(kResampleAllpass1[0], diff, state0);
+    state0 = in32;
+    diff = tmp1 - state2;
+    tmp2 = MUL_ACCUM_1(kResampleAllpass1[1], diff, state1);
+    state1 = tmp1;
+    diff = tmp2 - state3;
+    state3 = MUL_ACCUM_2(kResampleAllpass1[2], diff, state2);
+    state2 = tmp2;
+
+    // round; limit amplitude to prevent wrap-around; write to output array
+    out32 = (state3 + 512) >> 10;
+    *out++ = WebRtcSpl_SatW32ToW16(out32);
+
+    // upper allpass filter
+    diff = in32 - state5;
+    tmp1 = MUL_ACCUM_1(kResampleAllpass2[0], diff, state4);
+    state4 = in32;
+    diff = tmp1 - state6;
+    tmp2 = MUL_ACCUM_2(kResampleAllpass2[1], diff, state5);
+    state5 = tmp1;
+    diff = tmp2 - state7;
+    state7 = MUL_ACCUM_2(kResampleAllpass2[2], diff, state6);
+    state6 = tmp2;
+
+    // round; limit amplitude to prevent wrap-around; write to output array
+    out32 = (state7 + 512) >> 10;
+    *out++ = WebRtcSpl_SatW32ToW16(out32);
+  }
+
+  filtState[0] = state0;
+  filtState[1] = state1;
+  filtState[2] = state2;
+  filtState[3] = state3;
+  filtState[4] = state4;
+  filtState[5] = state5;
+  filtState[6] = state6;
+  filtState[7] = state7;
+}
diff --git a/trunk/src/common_audio/signal_processing/resample_by_2_internal.c b/trunk/src/common_audio/signal_processing/resample_by_2_internal.c
new file mode 100644
index 0000000..cbd2395
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/resample_by_2_internal.c
@@ -0,0 +1,679 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This header file contains some internal resampling functions.
+ *
+ */
+
+#include "resample_by_2_internal.h"
+
+// allpass filter coefficients.
+static const WebRtc_Word16 kResampleAllpass[2][3] = {
+        {821, 6110, 12382},
+        {3050, 9368, 15063}
+};
+
+//
+//   decimator
+// input:  WebRtc_Word32 (shifted 15 positions to the left, + offset 16384) OVERWRITTEN!
+// output: WebRtc_Word16 (saturated) (of length len/2)
+// state:  filter state array; length = 8
+
+void WebRtcSpl_DownBy2IntToShort(WebRtc_Word32 *in, WebRtc_Word32 len, WebRtc_Word16 *out,
+                                 WebRtc_Word32 *state)
+{
+    WebRtc_Word32 tmp0, tmp1, diff;
+    WebRtc_Word32 i;
+
+    len >>= 1;
+
+    // lower allpass filter (operates on even input samples)
+    for (i = 0; i < len; i++)
+    {
+        tmp0 = in[i << 1];
+        diff = tmp0 - state[1];
+        // scale down and round
+        diff = (diff + (1 << 13)) >> 14;
+        tmp1 = state[0] + diff * kResampleAllpass[1][0];
+        state[0] = tmp0;
+        diff = tmp1 - state[2];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        tmp0 = state[1] + diff * kResampleAllpass[1][1];
+        state[1] = tmp1;
+        diff = tmp0 - state[3];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        state[3] = state[2] + diff * kResampleAllpass[1][2];
+        state[2] = tmp0;
+
+        // divide by two and store temporarily
+        in[i << 1] = (state[3] >> 1);
+    }
+
+    in++;
+
+    // upper allpass filter (operates on odd input samples)
+    for (i = 0; i < len; i++)
+    {
+        tmp0 = in[i << 1];
+        diff = tmp0 - state[5];
+        // scale down and round
+        diff = (diff + (1 << 13)) >> 14;
+        tmp1 = state[4] + diff * kResampleAllpass[0][0];
+        state[4] = tmp0;
+        diff = tmp1 - state[6];
+        // scale down and round
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        tmp0 = state[5] + diff * kResampleAllpass[0][1];
+        state[5] = tmp1;
+        diff = tmp0 - state[7];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        state[7] = state[6] + diff * kResampleAllpass[0][2];
+        state[6] = tmp0;
+
+        // divide by two and store temporarily
+        in[i << 1] = (state[7] >> 1);
+    }
+
+    in--;
+
+    // combine allpass outputs
+    for (i = 0; i < len; i += 2)
+    {
+        // divide by two, add both allpass outputs and round
+        tmp0 = (in[i << 1] + in[(i << 1) + 1]) >> 15;
+        tmp1 = (in[(i << 1) + 2] + in[(i << 1) + 3]) >> 15;
+        if (tmp0 > (WebRtc_Word32)0x00007FFF)
+            tmp0 = 0x00007FFF;
+        if (tmp0 < (WebRtc_Word32)0xFFFF8000)
+            tmp0 = 0xFFFF8000;
+        out[i] = (WebRtc_Word16)tmp0;
+        if (tmp1 > (WebRtc_Word32)0x00007FFF)
+            tmp1 = 0x00007FFF;
+        if (tmp1 < (WebRtc_Word32)0xFFFF8000)
+            tmp1 = 0xFFFF8000;
+        out[i + 1] = (WebRtc_Word16)tmp1;
+    }
+}
+
+//
+//   decimator
+// input:  WebRtc_Word16
+// output: WebRtc_Word32 (shifted 15 positions to the left, + offset 16384) (of length len/2)
+// state:  filter state array; length = 8
+
+void WebRtcSpl_DownBy2ShortToInt(const WebRtc_Word16 *in,
+                                  WebRtc_Word32 len,
+                                  WebRtc_Word32 *out,
+                                  WebRtc_Word32 *state)
+{
+    WebRtc_Word32 tmp0, tmp1, diff;
+    WebRtc_Word32 i;
+
+    len >>= 1;
+
+    // lower allpass filter (operates on even input samples)
+    for (i = 0; i < len; i++)
+    {
+        tmp0 = ((WebRtc_Word32)in[i << 1] << 15) + (1 << 14);
+        diff = tmp0 - state[1];
+        // scale down and round
+        diff = (diff + (1 << 13)) >> 14;
+        tmp1 = state[0] + diff * kResampleAllpass[1][0];
+        state[0] = tmp0;
+        diff = tmp1 - state[2];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        tmp0 = state[1] + diff * kResampleAllpass[1][1];
+        state[1] = tmp1;
+        diff = tmp0 - state[3];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        state[3] = state[2] + diff * kResampleAllpass[1][2];
+        state[2] = tmp0;
+
+        // divide by two and store temporarily
+        out[i] = (state[3] >> 1);
+    }
+
+    in++;
+
+    // upper allpass filter (operates on odd input samples)
+    for (i = 0; i < len; i++)
+    {
+        tmp0 = ((WebRtc_Word32)in[i << 1] << 15) + (1 << 14);
+        diff = tmp0 - state[5];
+        // scale down and round
+        diff = (diff + (1 << 13)) >> 14;
+        tmp1 = state[4] + diff * kResampleAllpass[0][0];
+        state[4] = tmp0;
+        diff = tmp1 - state[6];
+        // scale down and round
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        tmp0 = state[5] + diff * kResampleAllpass[0][1];
+        state[5] = tmp1;
+        diff = tmp0 - state[7];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        state[7] = state[6] + diff * kResampleAllpass[0][2];
+        state[6] = tmp0;
+
+        // divide by two and store temporarily
+        out[i] += (state[7] >> 1);
+    }
+
+    in--;
+}
+
+//
+//   interpolator
+// input:  WebRtc_Word16
+// output: WebRtc_Word32 (normalized, not saturated) (of length len*2)
+// state:  filter state array; length = 8
+void WebRtcSpl_UpBy2ShortToInt(const WebRtc_Word16 *in, WebRtc_Word32 len, WebRtc_Word32 *out,
+                               WebRtc_Word32 *state)
+{
+    WebRtc_Word32 tmp0, tmp1, diff;
+    WebRtc_Word32 i;
+
+    // upper allpass filter (generates odd output samples)
+    for (i = 0; i < len; i++)
+    {
+        tmp0 = ((WebRtc_Word32)in[i] << 15) + (1 << 14);
+        diff = tmp0 - state[5];
+        // scale down and round
+        diff = (diff + (1 << 13)) >> 14;
+        tmp1 = state[4] + diff * kResampleAllpass[0][0];
+        state[4] = tmp0;
+        diff = tmp1 - state[6];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        tmp0 = state[5] + diff * kResampleAllpass[0][1];
+        state[5] = tmp1;
+        diff = tmp0 - state[7];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        state[7] = state[6] + diff * kResampleAllpass[0][2];
+        state[6] = tmp0;
+
+        // scale down, round and store
+        out[i << 1] = state[7] >> 15;
+    }
+
+    out++;
+
+    // lower allpass filter (generates even output samples)
+    for (i = 0; i < len; i++)
+    {
+        tmp0 = ((WebRtc_Word32)in[i] << 15) + (1 << 14);
+        diff = tmp0 - state[1];
+        // scale down and round
+        diff = (diff + (1 << 13)) >> 14;
+        tmp1 = state[0] + diff * kResampleAllpass[1][0];
+        state[0] = tmp0;
+        diff = tmp1 - state[2];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        tmp0 = state[1] + diff * kResampleAllpass[1][1];
+        state[1] = tmp1;
+        diff = tmp0 - state[3];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        state[3] = state[2] + diff * kResampleAllpass[1][2];
+        state[2] = tmp0;
+
+        // scale down, round and store
+        out[i << 1] = state[3] >> 15;
+    }
+}
+
+//
+//   interpolator
+// input:  WebRtc_Word32 (shifted 15 positions to the left, + offset 16384)
+// output: WebRtc_Word32 (shifted 15 positions to the left, + offset 16384) (of length len*2)
+// state:  filter state array; length = 8
+void WebRtcSpl_UpBy2IntToInt(const WebRtc_Word32 *in, WebRtc_Word32 len, WebRtc_Word32 *out,
+                             WebRtc_Word32 *state)
+{
+    WebRtc_Word32 tmp0, tmp1, diff;
+    WebRtc_Word32 i;
+
+    // upper allpass filter (generates odd output samples)
+    for (i = 0; i < len; i++)
+    {
+        tmp0 = in[i];
+        diff = tmp0 - state[5];
+        // scale down and round
+        diff = (diff + (1 << 13)) >> 14;
+        tmp1 = state[4] + diff * kResampleAllpass[0][0];
+        state[4] = tmp0;
+        diff = tmp1 - state[6];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        tmp0 = state[5] + diff * kResampleAllpass[0][1];
+        state[5] = tmp1;
+        diff = tmp0 - state[7];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        state[7] = state[6] + diff * kResampleAllpass[0][2];
+        state[6] = tmp0;
+
+        // scale down, round and store
+        out[i << 1] = state[7];
+    }
+
+    out++;
+
+    // lower allpass filter (generates even output samples)
+    for (i = 0; i < len; i++)
+    {
+        tmp0 = in[i];
+        diff = tmp0 - state[1];
+        // scale down and round
+        diff = (diff + (1 << 13)) >> 14;
+        tmp1 = state[0] + diff * kResampleAllpass[1][0];
+        state[0] = tmp0;
+        diff = tmp1 - state[2];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        tmp0 = state[1] + diff * kResampleAllpass[1][1];
+        state[1] = tmp1;
+        diff = tmp0 - state[3];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        state[3] = state[2] + diff * kResampleAllpass[1][2];
+        state[2] = tmp0;
+
+        // scale down, round and store
+        out[i << 1] = state[3];
+    }
+}
+
+//
+//   interpolator
+// input:  WebRtc_Word32 (shifted 15 positions to the left, + offset 16384)
+// output: WebRtc_Word16 (saturated) (of length len*2)
+// state:  filter state array; length = 8
+void WebRtcSpl_UpBy2IntToShort(const WebRtc_Word32 *in, WebRtc_Word32 len, WebRtc_Word16 *out,
+                               WebRtc_Word32 *state)
+{
+    WebRtc_Word32 tmp0, tmp1, diff;
+    WebRtc_Word32 i;
+
+    // upper allpass filter (generates odd output samples)
+    for (i = 0; i < len; i++)
+    {
+        tmp0 = in[i];
+        diff = tmp0 - state[5];
+        // scale down and round
+        diff = (diff + (1 << 13)) >> 14;
+        tmp1 = state[4] + diff * kResampleAllpass[0][0];
+        state[4] = tmp0;
+        diff = tmp1 - state[6];
+        // scale down and round
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        tmp0 = state[5] + diff * kResampleAllpass[0][1];
+        state[5] = tmp1;
+        diff = tmp0 - state[7];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        state[7] = state[6] + diff * kResampleAllpass[0][2];
+        state[6] = tmp0;
+
+        // scale down, saturate and store
+        tmp1 = state[7] >> 15;
+        if (tmp1 > (WebRtc_Word32)0x00007FFF)
+            tmp1 = 0x00007FFF;
+        if (tmp1 < (WebRtc_Word32)0xFFFF8000)
+            tmp1 = 0xFFFF8000;
+        out[i << 1] = (WebRtc_Word16)tmp1;
+    }
+
+    out++;
+
+    // lower allpass filter (generates even output samples)
+    for (i = 0; i < len; i++)
+    {
+        tmp0 = in[i];
+        diff = tmp0 - state[1];
+        // scale down and round
+        diff = (diff + (1 << 13)) >> 14;
+        tmp1 = state[0] + diff * kResampleAllpass[1][0];
+        state[0] = tmp0;
+        diff = tmp1 - state[2];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        tmp0 = state[1] + diff * kResampleAllpass[1][1];
+        state[1] = tmp1;
+        diff = tmp0 - state[3];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        state[3] = state[2] + diff * kResampleAllpass[1][2];
+        state[2] = tmp0;
+
+        // scale down, saturate and store
+        tmp1 = state[3] >> 15;
+        if (tmp1 > (WebRtc_Word32)0x00007FFF)
+            tmp1 = 0x00007FFF;
+        if (tmp1 < (WebRtc_Word32)0xFFFF8000)
+            tmp1 = 0xFFFF8000;
+        out[i << 1] = (WebRtc_Word16)tmp1;
+    }
+}
+
+//   lowpass filter
+// input:  WebRtc_Word16
+// output: WebRtc_Word32 (normalized, not saturated)
+// state:  filter state array; length = 8
+void WebRtcSpl_LPBy2ShortToInt(const WebRtc_Word16* in, WebRtc_Word32 len, WebRtc_Word32* out,
+                               WebRtc_Word32* state)
+{
+    WebRtc_Word32 tmp0, tmp1, diff;
+    WebRtc_Word32 i;
+
+    len >>= 1;
+
+    // lower allpass filter: odd input -> even output samples
+    in++;
+    // initial state of polyphase delay element
+    tmp0 = state[12];
+    for (i = 0; i < len; i++)
+    {
+        diff = tmp0 - state[1];
+        // scale down and round
+        diff = (diff + (1 << 13)) >> 14;
+        tmp1 = state[0] + diff * kResampleAllpass[1][0];
+        state[0] = tmp0;
+        diff = tmp1 - state[2];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        tmp0 = state[1] + diff * kResampleAllpass[1][1];
+        state[1] = tmp1;
+        diff = tmp0 - state[3];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        state[3] = state[2] + diff * kResampleAllpass[1][2];
+        state[2] = tmp0;
+
+        // scale down, round and store
+        out[i << 1] = state[3] >> 1;
+        tmp0 = ((WebRtc_Word32)in[i << 1] << 15) + (1 << 14);
+    }
+    in--;
+
+    // upper allpass filter: even input -> even output samples
+    for (i = 0; i < len; i++)
+    {
+        tmp0 = ((WebRtc_Word32)in[i << 1] << 15) + (1 << 14);
+        diff = tmp0 - state[5];
+        // scale down and round
+        diff = (diff + (1 << 13)) >> 14;
+        tmp1 = state[4] + diff * kResampleAllpass[0][0];
+        state[4] = tmp0;
+        diff = tmp1 - state[6];
+        // scale down and round
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        tmp0 = state[5] + diff * kResampleAllpass[0][1];
+        state[5] = tmp1;
+        diff = tmp0 - state[7];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        state[7] = state[6] + diff * kResampleAllpass[0][2];
+        state[6] = tmp0;
+
+        // average the two allpass outputs, scale down and store
+        out[i << 1] = (out[i << 1] + (state[7] >> 1)) >> 15;
+    }
+
+    // switch to odd output samples
+    out++;
+
+    // lower allpass filter: even input -> odd output samples
+    for (i = 0; i < len; i++)
+    {
+        tmp0 = ((WebRtc_Word32)in[i << 1] << 15) + (1 << 14);
+        diff = tmp0 - state[9];
+        // scale down and round
+        diff = (diff + (1 << 13)) >> 14;
+        tmp1 = state[8] + diff * kResampleAllpass[1][0];
+        state[8] = tmp0;
+        diff = tmp1 - state[10];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        tmp0 = state[9] + diff * kResampleAllpass[1][1];
+        state[9] = tmp1;
+        diff = tmp0 - state[11];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        state[11] = state[10] + diff * kResampleAllpass[1][2];
+        state[10] = tmp0;
+
+        // scale down, round and store
+        out[i << 1] = state[11] >> 1;
+    }
+
+    // upper allpass filter: odd input -> odd output samples
+    in++;
+    for (i = 0; i < len; i++)
+    {
+        tmp0 = ((WebRtc_Word32)in[i << 1] << 15) + (1 << 14);
+        diff = tmp0 - state[13];
+        // scale down and round
+        diff = (diff + (1 << 13)) >> 14;
+        tmp1 = state[12] + diff * kResampleAllpass[0][0];
+        state[12] = tmp0;
+        diff = tmp1 - state[14];
+        // scale down and round
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        tmp0 = state[13] + diff * kResampleAllpass[0][1];
+        state[13] = tmp1;
+        diff = tmp0 - state[15];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        state[15] = state[14] + diff * kResampleAllpass[0][2];
+        state[14] = tmp0;
+
+        // average the two allpass outputs, scale down and store
+        out[i << 1] = (out[i << 1] + (state[15] >> 1)) >> 15;
+    }
+}
+
+//   lowpass filter
+// input:  WebRtc_Word32 (shifted 15 positions to the left, + offset 16384)
+// output: WebRtc_Word32 (normalized, not saturated)
+// state:  filter state array; length = 8
+void WebRtcSpl_LPBy2IntToInt(const WebRtc_Word32* in, WebRtc_Word32 len, WebRtc_Word32* out,
+                             WebRtc_Word32* state)
+{
+    WebRtc_Word32 tmp0, tmp1, diff;
+    WebRtc_Word32 i;
+
+    len >>= 1;
+
+    // lower allpass filter: odd input -> even output samples
+    in++;
+    // initial state of polyphase delay element
+    tmp0 = state[12];
+    for (i = 0; i < len; i++)
+    {
+        diff = tmp0 - state[1];
+        // scale down and round
+        diff = (diff + (1 << 13)) >> 14;
+        tmp1 = state[0] + diff * kResampleAllpass[1][0];
+        state[0] = tmp0;
+        diff = tmp1 - state[2];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        tmp0 = state[1] + diff * kResampleAllpass[1][1];
+        state[1] = tmp1;
+        diff = tmp0 - state[3];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        state[3] = state[2] + diff * kResampleAllpass[1][2];
+        state[2] = tmp0;
+
+        // scale down, round and store
+        out[i << 1] = state[3] >> 1;
+        tmp0 = in[i << 1];
+    }
+    in--;
+
+    // upper allpass filter: even input -> even output samples
+    for (i = 0; i < len; i++)
+    {
+        tmp0 = in[i << 1];
+        diff = tmp0 - state[5];
+        // scale down and round
+        diff = (diff + (1 << 13)) >> 14;
+        tmp1 = state[4] + diff * kResampleAllpass[0][0];
+        state[4] = tmp0;
+        diff = tmp1 - state[6];
+        // scale down and round
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        tmp0 = state[5] + diff * kResampleAllpass[0][1];
+        state[5] = tmp1;
+        diff = tmp0 - state[7];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        state[7] = state[6] + diff * kResampleAllpass[0][2];
+        state[6] = tmp0;
+
+        // average the two allpass outputs, scale down and store
+        out[i << 1] = (out[i << 1] + (state[7] >> 1)) >> 15;
+    }
+
+    // switch to odd output samples
+    out++;
+
+    // lower allpass filter: even input -> odd output samples
+    for (i = 0; i < len; i++)
+    {
+        tmp0 = in[i << 1];
+        diff = tmp0 - state[9];
+        // scale down and round
+        diff = (diff + (1 << 13)) >> 14;
+        tmp1 = state[8] + diff * kResampleAllpass[1][0];
+        state[8] = tmp0;
+        diff = tmp1 - state[10];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        tmp0 = state[9] + diff * kResampleAllpass[1][1];
+        state[9] = tmp1;
+        diff = tmp0 - state[11];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        state[11] = state[10] + diff * kResampleAllpass[1][2];
+        state[10] = tmp0;
+
+        // scale down, round and store
+        out[i << 1] = state[11] >> 1;
+    }
+
+    // upper allpass filter: odd input -> odd output samples
+    in++;
+    for (i = 0; i < len; i++)
+    {
+        tmp0 = in[i << 1];
+        diff = tmp0 - state[13];
+        // scale down and round
+        diff = (diff + (1 << 13)) >> 14;
+        tmp1 = state[12] + diff * kResampleAllpass[0][0];
+        state[12] = tmp0;
+        diff = tmp1 - state[14];
+        // scale down and round
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        tmp0 = state[13] + diff * kResampleAllpass[0][1];
+        state[13] = tmp1;
+        diff = tmp0 - state[15];
+        // scale down and truncate
+        diff = diff >> 14;
+        if (diff < 0)
+            diff += 1;
+        state[15] = state[14] + diff * kResampleAllpass[0][2];
+        state[14] = tmp0;
+
+        // average the two allpass outputs, scale down and store
+        out[i << 1] = (out[i << 1] + (state[15] >> 1)) >> 15;
+    }
+}
diff --git a/trunk/src/common_audio/signal_processing/resample_by_2_internal.h b/trunk/src/common_audio/signal_processing/resample_by_2_internal.h
new file mode 100644
index 0000000..b6ac9f0
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/resample_by_2_internal.h
@@ -0,0 +1,47 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This header file contains some internal resampling functions.
+ *
+ */
+
+#ifndef WEBRTC_SPL_RESAMPLE_BY_2_INTERNAL_H_
+#define WEBRTC_SPL_RESAMPLE_BY_2_INTERNAL_H_
+
+#include "typedefs.h"
+
+/*******************************************************************
+ * resample_by_2_fast.c
+ * Functions for internal use in the other resample functions
+ ******************************************************************/
+void WebRtcSpl_DownBy2IntToShort(WebRtc_Word32 *in, WebRtc_Word32 len, WebRtc_Word16 *out,
+                                 WebRtc_Word32 *state);
+
+void WebRtcSpl_DownBy2ShortToInt(const WebRtc_Word16 *in, WebRtc_Word32 len,
+                                 WebRtc_Word32 *out, WebRtc_Word32 *state);
+
+void WebRtcSpl_UpBy2ShortToInt(const WebRtc_Word16 *in, WebRtc_Word32 len,
+                               WebRtc_Word32 *out, WebRtc_Word32 *state);
+
+void WebRtcSpl_UpBy2IntToInt(const WebRtc_Word32 *in, WebRtc_Word32 len, WebRtc_Word32 *out,
+                             WebRtc_Word32 *state);
+
+void WebRtcSpl_UpBy2IntToShort(const WebRtc_Word32 *in, WebRtc_Word32 len,
+                               WebRtc_Word16 *out, WebRtc_Word32 *state);
+
+void WebRtcSpl_LPBy2ShortToInt(const WebRtc_Word16* in, WebRtc_Word32 len,
+                               WebRtc_Word32* out, WebRtc_Word32* state);
+
+void WebRtcSpl_LPBy2IntToInt(const WebRtc_Word32* in, WebRtc_Word32 len, WebRtc_Word32* out,
+                             WebRtc_Word32* state);
+
+#endif // WEBRTC_SPL_RESAMPLE_BY_2_INTERNAL_H_
diff --git a/trunk/src/common_audio/signal_processing/resample_fractional.c b/trunk/src/common_audio/signal_processing/resample_fractional.c
new file mode 100644
index 0000000..51003d4
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/resample_fractional.c
@@ -0,0 +1,242 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains the resampling functions between 48, 44, 32 and 24 kHz.
+ * The description headers can be found in signal_processing_library.h
+ *
+ */
+
+#include "signal_processing_library.h"
+
+// interpolation coefficients
+static const WebRtc_Word16 kCoefficients48To32[2][8] = {
+        {778, -2050, 1087, 23285, 12903, -3783, 441, 222},
+        {222, 441, -3783, 12903, 23285, 1087, -2050, 778}
+};
+
+static const WebRtc_Word16 kCoefficients32To24[3][8] = {
+        {767, -2362, 2434, 24406, 10620, -3838, 721, 90},
+        {386, -381, -2646, 19062, 19062, -2646, -381, 386},
+        {90, 721, -3838, 10620, 24406, 2434, -2362, 767}
+};
+
+static const WebRtc_Word16 kCoefficients44To32[4][9] = {
+        {117, -669, 2245, -6183, 26267, 13529, -3245, 845, -138},
+        {-101, 612, -2283, 8532, 29790, -5138, 1789, -524, 91},
+        {50, -292, 1016, -3064, 32010, 3933, -1147, 315, -53},
+        {-156, 974, -3863, 18603, 21691, -6246, 2353, -712, 126}
+};
+
+//   Resampling ratio: 2/3
+// input:  WebRtc_Word32 (normalized, not saturated) :: size 3 * K
+// output: WebRtc_Word32 (shifted 15 positions to the left, + offset 16384) :: size 2 * K
+//      K: number of blocks
+
+void WebRtcSpl_Resample48khzTo32khz(const WebRtc_Word32 *In, WebRtc_Word32 *Out,
+                                    const WebRtc_Word32 K)
+{
+    /////////////////////////////////////////////////////////////
+    // Filter operation:
+    //
+    // Perform resampling (3 input samples -> 2 output samples);
+    // process in sub blocks of size 3 samples.
+    WebRtc_Word32 tmp;
+    WebRtc_Word32 m;
+
+    for (m = 0; m < K; m++)
+    {
+        tmp = 1 << 14;
+        tmp += kCoefficients48To32[0][0] * In[0];
+        tmp += kCoefficients48To32[0][1] * In[1];
+        tmp += kCoefficients48To32[0][2] * In[2];
+        tmp += kCoefficients48To32[0][3] * In[3];
+        tmp += kCoefficients48To32[0][4] * In[4];
+        tmp += kCoefficients48To32[0][5] * In[5];
+        tmp += kCoefficients48To32[0][6] * In[6];
+        tmp += kCoefficients48To32[0][7] * In[7];
+        Out[0] = tmp;
+
+        tmp = 1 << 14;
+        tmp += kCoefficients48To32[1][0] * In[1];
+        tmp += kCoefficients48To32[1][1] * In[2];
+        tmp += kCoefficients48To32[1][2] * In[3];
+        tmp += kCoefficients48To32[1][3] * In[4];
+        tmp += kCoefficients48To32[1][4] * In[5];
+        tmp += kCoefficients48To32[1][5] * In[6];
+        tmp += kCoefficients48To32[1][6] * In[7];
+        tmp += kCoefficients48To32[1][7] * In[8];
+        Out[1] = tmp;
+
+        // update pointers
+        In += 3;
+        Out += 2;
+    }
+}
+
+//   Resampling ratio: 3/4
+// input:  WebRtc_Word32 (normalized, not saturated) :: size 4 * K
+// output: WebRtc_Word32 (shifted 15 positions to the left, + offset 16384) :: size 3 * K
+//      K: number of blocks
+
+void WebRtcSpl_Resample32khzTo24khz(const WebRtc_Word32 *In, WebRtc_Word32 *Out,
+                                    const WebRtc_Word32 K)
+{
+    /////////////////////////////////////////////////////////////
+    // Filter operation:
+    //
+    // Perform resampling (4 input samples -> 3 output samples);
+    // process in sub blocks of size 4 samples.
+    WebRtc_Word32 m;
+    WebRtc_Word32 tmp;
+
+    for (m = 0; m < K; m++)
+    {
+        tmp = 1 << 14;
+        tmp += kCoefficients32To24[0][0] * In[0];
+        tmp += kCoefficients32To24[0][1] * In[1];
+        tmp += kCoefficients32To24[0][2] * In[2];
+        tmp += kCoefficients32To24[0][3] * In[3];
+        tmp += kCoefficients32To24[0][4] * In[4];
+        tmp += kCoefficients32To24[0][5] * In[5];
+        tmp += kCoefficients32To24[0][6] * In[6];
+        tmp += kCoefficients32To24[0][7] * In[7];
+        Out[0] = tmp;
+
+        tmp = 1 << 14;
+        tmp += kCoefficients32To24[1][0] * In[1];
+        tmp += kCoefficients32To24[1][1] * In[2];
+        tmp += kCoefficients32To24[1][2] * In[3];
+        tmp += kCoefficients32To24[1][3] * In[4];
+        tmp += kCoefficients32To24[1][4] * In[5];
+        tmp += kCoefficients32To24[1][5] * In[6];
+        tmp += kCoefficients32To24[1][6] * In[7];
+        tmp += kCoefficients32To24[1][7] * In[8];
+        Out[1] = tmp;
+
+        tmp = 1 << 14;
+        tmp += kCoefficients32To24[2][0] * In[2];
+        tmp += kCoefficients32To24[2][1] * In[3];
+        tmp += kCoefficients32To24[2][2] * In[4];
+        tmp += kCoefficients32To24[2][3] * In[5];
+        tmp += kCoefficients32To24[2][4] * In[6];
+        tmp += kCoefficients32To24[2][5] * In[7];
+        tmp += kCoefficients32To24[2][6] * In[8];
+        tmp += kCoefficients32To24[2][7] * In[9];
+        Out[2] = tmp;
+
+        // update pointers
+        In += 4;
+        Out += 3;
+    }
+}
+
+//
+// fractional resampling filters
+//   Fout = 11/16 * Fin
+//   Fout =  8/11 * Fin
+//
+
+// compute two inner-products and store them to output array
+static void WebRtcSpl_ResampDotProduct(const WebRtc_Word32 *in1, const WebRtc_Word32 *in2,
+                               const WebRtc_Word16 *coef_ptr, WebRtc_Word32 *out1,
+                               WebRtc_Word32 *out2)
+{
+    WebRtc_Word32 tmp1 = 16384;
+    WebRtc_Word32 tmp2 = 16384;
+    WebRtc_Word16 coef;
+
+    coef = coef_ptr[0];
+    tmp1 += coef * in1[0];
+    tmp2 += coef * in2[-0];
+
+    coef = coef_ptr[1];
+    tmp1 += coef * in1[1];
+    tmp2 += coef * in2[-1];
+
+    coef = coef_ptr[2];
+    tmp1 += coef * in1[2];
+    tmp2 += coef * in2[-2];
+
+    coef = coef_ptr[3];
+    tmp1 += coef * in1[3];
+    tmp2 += coef * in2[-3];
+
+    coef = coef_ptr[4];
+    tmp1 += coef * in1[4];
+    tmp2 += coef * in2[-4];
+
+    coef = coef_ptr[5];
+    tmp1 += coef * in1[5];
+    tmp2 += coef * in2[-5];
+
+    coef = coef_ptr[6];
+    tmp1 += coef * in1[6];
+    tmp2 += coef * in2[-6];
+
+    coef = coef_ptr[7];
+    tmp1 += coef * in1[7];
+    tmp2 += coef * in2[-7];
+
+    coef = coef_ptr[8];
+    *out1 = tmp1 + coef * in1[8];
+    *out2 = tmp2 + coef * in2[-8];
+}
+
+//   Resampling ratio: 8/11
+// input:  WebRtc_Word32 (normalized, not saturated) :: size 11 * K
+// output: WebRtc_Word32 (shifted 15 positions to the left, + offset 16384) :: size  8 * K
+//      K: number of blocks
+
+void WebRtcSpl_Resample44khzTo32khz(const WebRtc_Word32 *In, WebRtc_Word32 *Out,
+                                    const WebRtc_Word32 K)
+{
+    /////////////////////////////////////////////////////////////
+    // Filter operation:
+    //
+    // Perform resampling (11 input samples -> 8 output samples);
+    // process in sub blocks of size 11 samples.
+    WebRtc_Word32 tmp;
+    WebRtc_Word32 m;
+
+    for (m = 0; m < K; m++)
+    {
+        tmp = 1 << 14;
+
+        // first output sample
+        Out[0] = ((WebRtc_Word32)In[3] << 15) + tmp;
+
+        // sum and accumulate filter coefficients and input samples
+        tmp += kCoefficients44To32[3][0] * In[5];
+        tmp += kCoefficients44To32[3][1] * In[6];
+        tmp += kCoefficients44To32[3][2] * In[7];
+        tmp += kCoefficients44To32[3][3] * In[8];
+        tmp += kCoefficients44To32[3][4] * In[9];
+        tmp += kCoefficients44To32[3][5] * In[10];
+        tmp += kCoefficients44To32[3][6] * In[11];
+        tmp += kCoefficients44To32[3][7] * In[12];
+        tmp += kCoefficients44To32[3][8] * In[13];
+        Out[4] = tmp;
+
+        // sum and accumulate filter coefficients and input samples
+        WebRtcSpl_ResampDotProduct(&In[0], &In[17], kCoefficients44To32[0], &Out[1], &Out[7]);
+
+        // sum and accumulate filter coefficients and input samples
+        WebRtcSpl_ResampDotProduct(&In[2], &In[15], kCoefficients44To32[1], &Out[2], &Out[6]);
+
+        // sum and accumulate filter coefficients and input samples
+        WebRtcSpl_ResampDotProduct(&In[3], &In[14], kCoefficients44To32[2], &Out[3], &Out[5]);
+
+        // update pointers
+        In += 11;
+        Out += 8;
+    }
+}
diff --git a/trunk/src/common_audio/signal_processing/signal_processing.gypi b/trunk/src/common_audio/signal_processing/signal_processing.gypi
new file mode 100644
index 0000000..c67bf7c
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/signal_processing.gypi
@@ -0,0 +1,85 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+  'targets': [
+    {
+      'target_name': 'signal_processing',
+      'type': '<(library)',
+      'include_dirs': [
+        'include',
+      ],
+      'direct_dependent_settings': {
+        'include_dirs': [
+          'include',
+        ],
+      },
+      'sources': [
+        'include/signal_processing_library.h',
+        'include/spl_inl.h',
+        'auto_corr_to_refl_coef.c',
+        'auto_correlation.c',
+        'complex_fft.c',
+        'complex_bit_reverse.c',
+        'copy_set_operations.c',
+        'cross_correlation.c',
+        'division_operations.c',
+        'dot_product_with_scale.c',
+        'downsample_fast.c',
+        'energy.c',
+        'filter_ar.c',
+        'filter_ar_fast_q12.c',
+        'filter_ma_fast_q12.c',
+        'get_hanning_window.c',
+        'get_scaling_square.c',
+        'ilbc_specific_functions.c',
+        'levinson_durbin.c',
+        'lpc_to_refl_coef.c',
+        'min_max_operations.c',
+        'randomization_functions.c',
+        'refl_coef_to_lpc.c',
+        'resample.c',
+        'resample_48khz.c',
+        'resample_by_2.c',
+        'resample_by_2_internal.c',
+        'resample_by_2_internal.h',
+        'resample_fractional.c',
+        'spl_sqrt.c',
+        'spl_sqrt_floor.c',
+        'spl_version.c',
+        'splitting_filter.c',
+        'sqrt_of_one_minus_x_squared.c',
+        'vector_scaling_operations.c',
+      ],
+    }, # spl
+  ], # targets
+  'conditions': [
+    ['build_with_chromium==0', {
+      'targets': [
+        {
+          'target_name': 'signal_processing_unittests',
+          'type': 'executable',
+          'dependencies': [
+            'signal_processing',
+            '<(webrtc_root)/../test/test.gyp:test_support_main',
+            '<(webrtc_root)/../testing/gtest.gyp:gtest',
+          ],
+          'sources': [
+            'signal_processing_unittest.cc',
+          ],
+        }, # spl_unittests
+      ], # targets
+    }], # build_with_chromium
+  ], # conditions
+}
+
+# Local Variables:
+# tab-width:2
+# indent-tabs-mode:nil
+# End:
+# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/common_audio/signal_processing/signal_processing_unittest.cc b/trunk/src/common_audio/signal_processing/signal_processing_unittest.cc
new file mode 100644
index 0000000..f6d1123
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/signal_processing_unittest.cc
@@ -0,0 +1,454 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "signal_processing_library.h"
+#include "gtest/gtest.h"
+
+class SplTest : public testing::Test {
+ protected:
+  virtual ~SplTest() {
+  }
+  void SetUp() {
+  }
+  void TearDown() {
+  }
+};
+
+TEST_F(SplTest, MacroTest) {
+    // Macros with inputs.
+    int A = 10;
+    int B = 21;
+    int a = -3;
+    int b = WEBRTC_SPL_WORD32_MAX;
+    int nr = 2;
+    int d_ptr2 = 0;
+
+    EXPECT_EQ(10, WEBRTC_SPL_MIN(A, B));
+    EXPECT_EQ(21, WEBRTC_SPL_MAX(A, B));
+
+    EXPECT_EQ(3, WEBRTC_SPL_ABS_W16(a));
+    EXPECT_EQ(3, WEBRTC_SPL_ABS_W32(a));
+    EXPECT_EQ(0, WEBRTC_SPL_GET_BYTE(&B, nr));
+    WEBRTC_SPL_SET_BYTE(&d_ptr2, 1, nr);
+    EXPECT_EQ(65536, d_ptr2);
+
+    EXPECT_EQ(-63, WEBRTC_SPL_MUL(a, B));
+    EXPECT_EQ(-2147483645, WEBRTC_SPL_MUL(a, b));
+    EXPECT_EQ(2147483651u, WEBRTC_SPL_UMUL(a, b));
+    b = WEBRTC_SPL_WORD16_MAX >> 1;
+    EXPECT_EQ(65535u, WEBRTC_SPL_UMUL_RSFT16(a, b));
+    EXPECT_EQ(1073627139u, WEBRTC_SPL_UMUL_16_16(a, b));
+    EXPECT_EQ(16382u, WEBRTC_SPL_UMUL_16_16_RSFT16(a, b));
+    EXPECT_EQ(4294918147u, WEBRTC_SPL_UMUL_32_16(a, b));
+    EXPECT_EQ(65535u, WEBRTC_SPL_UMUL_32_16_RSFT16(a, b));
+    EXPECT_EQ(-49149, WEBRTC_SPL_MUL_16_U16(a, b));
+
+    a = b;
+    b = -3;
+    EXPECT_EQ(-5461, WEBRTC_SPL_DIV(a, b));
+    EXPECT_EQ(0u, WEBRTC_SPL_UDIV(a, b));
+
+    EXPECT_EQ(-1, WEBRTC_SPL_MUL_16_32_RSFT16(a, b));
+    EXPECT_EQ(-1, WEBRTC_SPL_MUL_16_32_RSFT15(a, b));
+    EXPECT_EQ(-3, WEBRTC_SPL_MUL_16_32_RSFT14(a, b));
+    EXPECT_EQ(-24, WEBRTC_SPL_MUL_16_32_RSFT11(a, b));
+
+    int a32 = WEBRTC_SPL_WORD32_MAX;
+    int a32a = (WEBRTC_SPL_WORD32_MAX >> 16);
+    int a32b = (WEBRTC_SPL_WORD32_MAX & 0x0000ffff);
+    EXPECT_EQ(5, WEBRTC_SPL_MUL_32_32_RSFT32(a32a, a32b, A));
+    EXPECT_EQ(5, WEBRTC_SPL_MUL_32_32_RSFT32BI(a32, A));
+
+    EXPECT_EQ(-49149, WEBRTC_SPL_MUL_16_16(a, b));
+    EXPECT_EQ(-12288, WEBRTC_SPL_MUL_16_16_RSFT(a, b, 2));
+
+    EXPECT_EQ(-12287, WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(a, b, 2));
+    EXPECT_EQ(-1, WEBRTC_SPL_MUL_16_16_RSFT_WITH_FIXROUND(a, b));
+
+    EXPECT_EQ(16380, WEBRTC_SPL_ADD_SAT_W32(a, b));
+    EXPECT_EQ(21, WEBRTC_SPL_SAT(a, A, B));
+    EXPECT_EQ(21, WEBRTC_SPL_SAT(a, B, A));
+    EXPECT_EQ(-49149, WEBRTC_SPL_MUL_32_16(a, b));
+
+    EXPECT_EQ(16386, WEBRTC_SPL_SUB_SAT_W32(a, b));
+    EXPECT_EQ(16380, WEBRTC_SPL_ADD_SAT_W16(a, b));
+    EXPECT_EQ(16386, WEBRTC_SPL_SUB_SAT_W16(a, b));
+
+    EXPECT_TRUE(WEBRTC_SPL_IS_NEG(b));
+
+    // Shifting with negative numbers allowed
+    int shift_amount = 1;  // Workaround compiler warning using variable here.
+    // Positive means left shift
+    EXPECT_EQ(32766, WEBRTC_SPL_SHIFT_W16(a, shift_amount));
+    EXPECT_EQ(32766, WEBRTC_SPL_SHIFT_W32(a, shift_amount));
+
+    // Shifting with negative numbers not allowed
+    // We cannot do casting here due to signed/unsigned problem
+    EXPECT_EQ(8191, WEBRTC_SPL_RSHIFT_W16(a, 1));
+    EXPECT_EQ(32766, WEBRTC_SPL_LSHIFT_W16(a, 1));
+    EXPECT_EQ(8191, WEBRTC_SPL_RSHIFT_W32(a, 1));
+    EXPECT_EQ(32766, WEBRTC_SPL_LSHIFT_W32(a, 1));
+
+    EXPECT_EQ(8191, WEBRTC_SPL_RSHIFT_U16(a, 1));
+    EXPECT_EQ(32766, WEBRTC_SPL_LSHIFT_U16(a, 1));
+    EXPECT_EQ(8191u, WEBRTC_SPL_RSHIFT_U32(a, 1));
+    EXPECT_EQ(32766u, WEBRTC_SPL_LSHIFT_U32(a, 1));
+
+    EXPECT_EQ(1470, WEBRTC_SPL_RAND(A));
+}
+
+TEST_F(SplTest, InlineTest) {
+    WebRtc_Word16 a = 121;
+    WebRtc_Word16 b = -17;
+    WebRtc_Word32 A = 111121;
+    WebRtc_Word32 B = -1711;
+    char bVersion[8];
+
+    EXPECT_EQ(104, WebRtcSpl_AddSatW16(a, b));
+    EXPECT_EQ(138, WebRtcSpl_SubSatW16(a, b));
+
+    EXPECT_EQ(109410, WebRtcSpl_AddSatW32(A, B));
+    EXPECT_EQ(112832, WebRtcSpl_SubSatW32(A, B));
+
+    EXPECT_EQ(17, WebRtcSpl_GetSizeInBits(A));
+    EXPECT_EQ(14, WebRtcSpl_NormW32(A));
+    EXPECT_EQ(4, WebRtcSpl_NormW16(B));
+    EXPECT_EQ(15, WebRtcSpl_NormU32(A));
+
+    EXPECT_EQ(0, WebRtcSpl_get_version(bVersion, 8));
+}
+
+TEST_F(SplTest, MathOperationsTest) {
+    int A = 117;
+    WebRtc_Word32 num = 117;
+    WebRtc_Word32 den = -5;
+    WebRtc_UWord16 denU = 5;
+    EXPECT_EQ(10, WebRtcSpl_Sqrt(A));
+    EXPECT_EQ(10, WebRtcSpl_SqrtFloor(A));
+
+
+    EXPECT_EQ(-91772805, WebRtcSpl_DivResultInQ31(den, num));
+    EXPECT_EQ(-23, WebRtcSpl_DivW32W16ResW16(num, (WebRtc_Word16)den));
+    EXPECT_EQ(-23, WebRtcSpl_DivW32W16(num, (WebRtc_Word16)den));
+    EXPECT_EQ(23u, WebRtcSpl_DivU32U16(num, denU));
+    EXPECT_EQ(0, WebRtcSpl_DivW32HiLow(128, 0, 256));
+}
+
+TEST_F(SplTest, BasicArrayOperationsTest) {
+    const int kVectorSize = 4;
+    int B[] = {4, 12, 133, 1100};
+    WebRtc_UWord8 b8[kVectorSize];
+    WebRtc_Word16 b16[kVectorSize];
+    WebRtc_Word32 b32[kVectorSize];
+
+    WebRtc_UWord8 bTmp8[kVectorSize];
+    WebRtc_Word16 bTmp16[kVectorSize];
+    WebRtc_Word32 bTmp32[kVectorSize];
+
+    WebRtcSpl_MemSetW16(b16, 3, kVectorSize);
+    for (int kk = 0; kk < kVectorSize; ++kk) {
+        EXPECT_EQ(3, b16[kk]);
+    }
+    EXPECT_EQ(kVectorSize, WebRtcSpl_ZerosArrayW16(b16, kVectorSize));
+    for (int kk = 0; kk < kVectorSize; ++kk) {
+        EXPECT_EQ(0, b16[kk]);
+    }
+    EXPECT_EQ(kVectorSize, WebRtcSpl_OnesArrayW16(b16, kVectorSize));
+    for (int kk = 0; kk < kVectorSize; ++kk) {
+        EXPECT_EQ(1, b16[kk]);
+    }
+    WebRtcSpl_MemSetW32(b32, 3, kVectorSize);
+    for (int kk = 0; kk < kVectorSize; ++kk) {
+        EXPECT_EQ(3, b32[kk]);
+    }
+    EXPECT_EQ(kVectorSize, WebRtcSpl_ZerosArrayW32(b32, kVectorSize));
+    for (int kk = 0; kk < kVectorSize; ++kk) {
+        EXPECT_EQ(0, b32[kk]);
+    }
+    EXPECT_EQ(kVectorSize, WebRtcSpl_OnesArrayW32(b32, kVectorSize));
+    for (int kk = 0; kk < kVectorSize; ++kk) {
+        EXPECT_EQ(1, b32[kk]);
+    }
+    for (int kk = 0; kk < kVectorSize; ++kk) {
+        bTmp8[kk] = (WebRtc_Word8)kk;
+        bTmp16[kk] = (WebRtc_Word16)kk;
+        bTmp32[kk] = (WebRtc_Word32)kk;
+    }
+    WEBRTC_SPL_MEMCPY_W8(b8, bTmp8, kVectorSize);
+    for (int kk = 0; kk < kVectorSize; ++kk) {
+        EXPECT_EQ(b8[kk], bTmp8[kk]);
+    }
+    WEBRTC_SPL_MEMCPY_W16(b16, bTmp16, kVectorSize);
+    for (int kk = 0; kk < kVectorSize; ++kk) {
+        EXPECT_EQ(b16[kk], bTmp16[kk]);
+    }
+//    WEBRTC_SPL_MEMCPY_W32(b32, bTmp32, kVectorSize);
+//    for (int kk = 0; kk < kVectorSize; ++kk) {
+//        EXPECT_EQ(b32[kk], bTmp32[kk]);
+//    }
+    EXPECT_EQ(2, WebRtcSpl_CopyFromEndW16(b16, kVectorSize, 2, bTmp16));
+    for (int kk = 0; kk < 2; ++kk) {
+        EXPECT_EQ(kk+2, bTmp16[kk]);
+    }
+
+    for (int kk = 0; kk < kVectorSize; ++kk) {
+        b32[kk] = B[kk];
+        b16[kk] = (WebRtc_Word16)B[kk];
+    }
+    WebRtcSpl_VectorBitShiftW32ToW16(bTmp16, kVectorSize, b32, 1);
+    for (int kk = 0; kk < kVectorSize; ++kk) {
+        EXPECT_EQ((B[kk]>>1), bTmp16[kk]);
+    }
+    WebRtcSpl_VectorBitShiftW16(bTmp16, kVectorSize, b16, 1);
+    for (int kk = 0; kk < kVectorSize; ++kk) {
+        EXPECT_EQ((B[kk]>>1), bTmp16[kk]);
+    }
+    WebRtcSpl_VectorBitShiftW32(bTmp32, kVectorSize, b32, 1);
+    for (int kk = 0; kk < kVectorSize; ++kk) {
+        EXPECT_EQ((B[kk]>>1), bTmp32[kk]);
+    }
+
+    WebRtcSpl_MemCpyReversedOrder(&bTmp16[3], b16, kVectorSize);
+    for (int kk = 0; kk < kVectorSize; ++kk) {
+        EXPECT_EQ(b16[3-kk], bTmp16[kk]);
+    }
+}
+
+TEST_F(SplTest, MinMaxOperationsTest) {
+    const int kVectorSize = 4;
+    int B[] = {4, 12, 133, -1100};
+    WebRtc_Word16 b16[kVectorSize];
+    WebRtc_Word32 b32[kVectorSize];
+
+    for (int kk = 0; kk < kVectorSize; ++kk) {
+        b16[kk] = B[kk];
+        b32[kk] = B[kk];
+    }
+
+    EXPECT_EQ(1100, WebRtcSpl_MaxAbsValueW16(b16, kVectorSize));
+    EXPECT_EQ(1100, WebRtcSpl_MaxAbsValueW32(b32, kVectorSize));
+    EXPECT_EQ(133, WebRtcSpl_MaxValueW16(b16, kVectorSize));
+    EXPECT_EQ(133, WebRtcSpl_MaxValueW32(b32, kVectorSize));
+    EXPECT_EQ(3, WebRtcSpl_MaxAbsIndexW16(b16, kVectorSize));
+    EXPECT_EQ(2, WebRtcSpl_MaxIndexW16(b16, kVectorSize));
+    EXPECT_EQ(2, WebRtcSpl_MaxIndexW32(b32, kVectorSize));
+
+    EXPECT_EQ(-1100, WebRtcSpl_MinValueW16(b16, kVectorSize));
+    EXPECT_EQ(-1100, WebRtcSpl_MinValueW32(b32, kVectorSize));
+    EXPECT_EQ(3, WebRtcSpl_MinIndexW16(b16, kVectorSize));
+    EXPECT_EQ(3, WebRtcSpl_MinIndexW32(b32, kVectorSize));
+
+    EXPECT_EQ(0, WebRtcSpl_GetScalingSquare(b16, kVectorSize, 1));
+}
+
+TEST_F(SplTest, VectorOperationsTest) {
+    const int kVectorSize = 4;
+    int B[] = {4, 12, 133, 1100};
+    WebRtc_Word16 a16[kVectorSize];
+    WebRtc_Word16 b16[kVectorSize];
+    WebRtc_Word32 b32[kVectorSize];
+    WebRtc_Word16 bTmp16[kVectorSize];
+
+    for (int kk = 0; kk < kVectorSize; ++kk) {
+        a16[kk] = B[kk];
+        b16[kk] = B[kk];
+    }
+
+    WebRtcSpl_AffineTransformVector(bTmp16, b16, 3, 7, 2, kVectorSize);
+    for (int kk = 0; kk < kVectorSize; ++kk) {
+        EXPECT_EQ((B[kk]*3+7)>>2, bTmp16[kk]);
+    }
+    WebRtcSpl_ScaleAndAddVectorsWithRound(b16, 3, b16, 2, 2, bTmp16, kVectorSize);
+    for (int kk = 0; kk < kVectorSize; ++kk) {
+        EXPECT_EQ((B[kk]*3+B[kk]*2+2)>>2, bTmp16[kk]);
+    }
+
+    WebRtcSpl_AddAffineVectorToVector(bTmp16, b16, 3, 7, 2, kVectorSize);
+    for (int kk = 0; kk < kVectorSize; ++kk) {
+        EXPECT_EQ(((B[kk]*3+B[kk]*2+2)>>2)+((b16[kk]*3+7)>>2), bTmp16[kk]);
+    }
+
+    WebRtcSpl_CrossCorrelation(b32, b16, bTmp16, kVectorSize, 2, 2, 0);
+    for (int kk = 0; kk < 2; ++kk) {
+        EXPECT_EQ(614236, b32[kk]);
+    }
+//    EXPECT_EQ(, WebRtcSpl_DotProduct(b16, bTmp16, 4));
+    EXPECT_EQ(306962, WebRtcSpl_DotProductWithScale(b16, b16, kVectorSize, 2));
+
+    WebRtcSpl_ScaleVector(b16, bTmp16, 13, kVectorSize, 2);
+    for (int kk = 0; kk < kVectorSize; ++kk) {
+        EXPECT_EQ((b16[kk]*13)>>2, bTmp16[kk]);
+    }
+    WebRtcSpl_ScaleVectorWithSat(b16, bTmp16, 13, kVectorSize, 2);
+    for (int kk = 0; kk < kVectorSize; ++kk) {
+        EXPECT_EQ((b16[kk]*13)>>2, bTmp16[kk]);
+    }
+    WebRtcSpl_ScaleAndAddVectors(a16, 13, 2, b16, 7, 2, bTmp16, kVectorSize);
+    for (int kk = 0; kk < kVectorSize; ++kk) {
+        EXPECT_EQ(((a16[kk]*13)>>2)+((b16[kk]*7)>>2), bTmp16[kk]);
+    }
+
+    WebRtcSpl_AddVectorsAndShift(bTmp16, a16, b16, kVectorSize, 2);
+    for (int kk = 0; kk < kVectorSize; ++kk) {
+        EXPECT_EQ(B[kk] >> 1, bTmp16[kk]);
+    }
+    WebRtcSpl_ReverseOrderMultArrayElements(bTmp16, a16, &b16[3], kVectorSize, 2);
+    for (int kk = 0; kk < kVectorSize; ++kk) {
+        EXPECT_EQ((a16[kk]*b16[3-kk])>>2, bTmp16[kk]);
+    }
+    WebRtcSpl_ElementwiseVectorMult(bTmp16, a16, b16, kVectorSize, 6);
+    for (int kk = 0; kk < kVectorSize; ++kk) {
+        EXPECT_EQ((a16[kk]*b16[kk])>>6, bTmp16[kk]);
+    }
+
+    WebRtcSpl_SqrtOfOneMinusXSquared(b16, kVectorSize, bTmp16);
+    for (int kk = 0; kk < kVectorSize - 1; ++kk) {
+        EXPECT_EQ(32767, bTmp16[kk]);
+    }
+    EXPECT_EQ(32749, bTmp16[kVectorSize - 1]);
+}
+
+TEST_F(SplTest, EstimatorsTest) {
+    const int kVectorSize = 4;
+    int B[] = {4, 12, 133, 1100};
+    WebRtc_Word16 b16[kVectorSize];
+    WebRtc_Word32 b32[kVectorSize];
+    WebRtc_Word16 bTmp16[kVectorSize];
+
+    for (int kk = 0; kk < kVectorSize; ++kk) {
+        b16[kk] = B[kk];
+        b32[kk] = B[kk];
+    }
+
+    EXPECT_EQ(0, WebRtcSpl_LevinsonDurbin(b32, b16, bTmp16, 2));
+}
+
+TEST_F(SplTest, FilterTest) {
+    const int kVectorSize = 4;
+    const int kFilterOrder = 3;
+    WebRtc_Word16 A[] = {1, 2, 33, 100};
+    WebRtc_Word16 A5[] = {1, 2, 33, 100, -5};
+    WebRtc_Word16 B[] = {4, 12, 133, 110};
+    WebRtc_Word16 data_in[kVectorSize];
+    WebRtc_Word16 data_out[kVectorSize];
+    WebRtc_Word16 bTmp16Low[kVectorSize];
+    WebRtc_Word16 bState[kVectorSize];
+    WebRtc_Word16 bStateLow[kVectorSize];
+
+    WebRtcSpl_ZerosArrayW16(bState, kVectorSize);
+    WebRtcSpl_ZerosArrayW16(bStateLow, kVectorSize);
+
+    for (int kk = 0; kk < kVectorSize; ++kk) {
+        data_in[kk] = A[kk];
+        data_out[kk] = 0;
+    }
+
+    // MA filters.
+    // Note that the input data has |kFilterOrder| states before the actual
+    // data (one sample).
+    WebRtcSpl_FilterMAFastQ12(&data_in[kFilterOrder], data_out, B,
+                              kFilterOrder + 1, 1);
+    EXPECT_EQ(0, data_out[0]);
+    // AR filters.
+    // Note that the output data has |kFilterOrder| states before the actual
+    // data (one sample).
+    WebRtcSpl_FilterARFastQ12(data_in, &data_out[kFilterOrder], A,
+                              kFilterOrder + 1, 1);
+    EXPECT_EQ(0, data_out[kFilterOrder]);
+
+    EXPECT_EQ(kVectorSize, WebRtcSpl_FilterAR(A5,
+                                              5,
+                                              data_in,
+                                              kVectorSize,
+                                              bState,
+                                              kVectorSize,
+                                              bStateLow,
+                                              kVectorSize,
+                                              data_out,
+                                              bTmp16Low,
+                                              kVectorSize));
+}
+
+TEST_F(SplTest, RandTest) {
+    const int kVectorSize = 4;
+    WebRtc_Word16 BU[] = {3653, 12446, 8525, 30691};
+    WebRtc_Word16 b16[kVectorSize];
+    WebRtc_UWord32 bSeed = 100000;
+
+    EXPECT_EQ(464449057u, WebRtcSpl_IncreaseSeed(&bSeed));
+    EXPECT_EQ(31565, WebRtcSpl_RandU(&bSeed));
+    EXPECT_EQ(-9786, WebRtcSpl_RandN(&bSeed));
+    EXPECT_EQ(kVectorSize, WebRtcSpl_RandUArray(b16, kVectorSize, &bSeed));
+    for (int kk = 0; kk < kVectorSize; ++kk) {
+        EXPECT_EQ(BU[kk], b16[kk]);
+    }
+}
+
+TEST_F(SplTest, SignalProcessingTest) {
+    const int kVectorSize = 4;
+    int A[] = {1, 2, 33, 100};
+    const WebRtc_Word16 kHanning[4] = { 2399, 8192, 13985, 16384 };
+    WebRtc_Word16 b16[kVectorSize];
+
+    WebRtc_Word16 bTmp16[kVectorSize];
+    WebRtc_Word32 bTmp32[kVectorSize];
+
+    int bScale = 0;
+
+    for (int kk = 0; kk < kVectorSize; ++kk) {
+        b16[kk] = A[kk];
+    }
+
+    EXPECT_EQ(2, WebRtcSpl_AutoCorrelation(b16, kVectorSize, 1, bTmp32, &bScale));
+    // TODO(bjornv): Activate the Reflection Coefficient tests when refactoring.
+//    WebRtcSpl_ReflCoefToLpc(b16, kVectorSize, bTmp16);
+////    for (int kk = 0; kk < kVectorSize; ++kk) {
+////        EXPECT_EQ(aTmp16[kk], bTmp16[kk]);
+////    }
+//    WebRtcSpl_LpcToReflCoef(bTmp16, kVectorSize, b16);
+////    for (int kk = 0; kk < kVectorSize; ++kk) {
+////        EXPECT_EQ(a16[kk], b16[kk]);
+////    }
+//    WebRtcSpl_AutoCorrToReflCoef(b32, kVectorSize, bTmp16);
+////    for (int kk = 0; kk < kVectorSize; ++kk) {
+////        EXPECT_EQ(aTmp16[kk], bTmp16[kk]);
+////    }
+
+    WebRtcSpl_GetHanningWindow(bTmp16, kVectorSize);
+    for (int kk = 0; kk < kVectorSize; ++kk) {
+        EXPECT_EQ(kHanning[kk], bTmp16[kk]);
+    }
+
+    for (int kk = 0; kk < kVectorSize; ++kk) {
+        b16[kk] = A[kk];
+    }
+    EXPECT_EQ(11094 , WebRtcSpl_Energy(b16, kVectorSize, &bScale));
+    EXPECT_EQ(0, bScale);
+}
+
+TEST_F(SplTest, FFTTest) {
+    WebRtc_Word16 B[] = {1, 2, 33, 100,
+            2, 3, 34, 101,
+            3, 4, 35, 102,
+            4, 5, 36, 103};
+
+    EXPECT_EQ(0, WebRtcSpl_ComplexFFT(B, 3, 1));
+//    for (int kk = 0; kk < 16; ++kk) {
+//        EXPECT_EQ(A[kk], B[kk]);
+//    }
+    EXPECT_EQ(0, WebRtcSpl_ComplexIFFT(B, 3, 1));
+//    for (int kk = 0; kk < 16; ++kk) {
+//        EXPECT_EQ(A[kk], B[kk]);
+//    }
+    WebRtcSpl_ComplexBitReverse(B, 3);
+    for (int kk = 0; kk < 16; ++kk) {
+        //EXPECT_EQ(A[kk], B[kk]);
+    }
+}
diff --git a/trunk/src/common_audio/signal_processing/spl_sqrt.c b/trunk/src/common_audio/signal_processing/spl_sqrt.c
new file mode 100644
index 0000000..cfe2cd3
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/spl_sqrt.c
@@ -0,0 +1,184 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains the function WebRtcSpl_Sqrt().
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "signal_processing_library.h"
+
+WebRtc_Word32 WebRtcSpl_SqrtLocal(WebRtc_Word32 in);
+
+WebRtc_Word32 WebRtcSpl_SqrtLocal(WebRtc_Word32 in)
+{
+
+    WebRtc_Word16 x_half, t16;
+    WebRtc_Word32 A, B, x2;
+
+    /* The following block performs:
+     y=in/2
+     x=y-2^30
+     x_half=x/2^31
+     t = 1 + (x_half) - 0.5*((x_half)^2) + 0.5*((x_half)^3) - 0.625*((x_half)^4)
+         + 0.875*((x_half)^5)
+     */
+
+    B = in;
+
+    B = WEBRTC_SPL_RSHIFT_W32(B, 1); // B = in/2
+    B = B - ((WebRtc_Word32)0x40000000); // B = in/2 - 1/2
+    x_half = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(B, 16);// x_half = x/2 = (in-1)/2
+    B = B + ((WebRtc_Word32)0x40000000); // B = 1 + x/2
+    B = B + ((WebRtc_Word32)0x40000000); // Add 0.5 twice (since 1.0 does not exist in Q31)
+
+    x2 = ((WebRtc_Word32)x_half) * ((WebRtc_Word32)x_half) * 2; // A = (x/2)^2
+    A = -x2; // A = -(x/2)^2
+    B = B + (A >> 1); // B = 1 + x/2 - 0.5*(x/2)^2
+
+    A = WEBRTC_SPL_RSHIFT_W32(A, 16);
+    A = A * A * 2; // A = (x/2)^4
+    t16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(A, 16);
+    B = B + WEBRTC_SPL_MUL_16_16(-20480, t16) * 2; // B = B - 0.625*A
+    // After this, B = 1 + x/2 - 0.5*(x/2)^2 - 0.625*(x/2)^4
+
+    t16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(A, 16);
+    A = WEBRTC_SPL_MUL_16_16(x_half, t16) * 2; // A = (x/2)^5
+    t16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(A, 16);
+    B = B + WEBRTC_SPL_MUL_16_16(28672, t16) * 2; // B = B + 0.875*A
+    // After this, B = 1 + x/2 - 0.5*(x/2)^2 - 0.625*(x/2)^4 + 0.875*(x/2)^5
+
+    t16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(x2, 16);
+    A = WEBRTC_SPL_MUL_16_16(x_half, t16) * 2; // A = x/2^3
+
+    B = B + (A >> 1); // B = B + 0.5*A
+    // After this, B = 1 + x/2 - 0.5*(x/2)^2 + 0.5*(x/2)^3 - 0.625*(x/2)^4 + 0.875*(x/2)^5
+
+    B = B + ((WebRtc_Word32)32768); // Round off bit
+
+    return B;
+}
+
+WebRtc_Word32 WebRtcSpl_Sqrt(WebRtc_Word32 value)
+{
+    /*
+     Algorithm:
+
+     Six term Taylor Series is used here to compute the square root of a number
+     y^0.5 = (1+x)^0.5 where x = y-1
+     = 1+(x/2)-0.5*((x/2)^2+0.5*((x/2)^3-0.625*((x/2)^4+0.875*((x/2)^5)
+     0.5 <= x < 1
+
+     Example of how the algorithm works, with ut=sqrt(in), and
+     with in=73632 and ut=271 (even shift value case):
+
+     in=73632
+     y= in/131072
+     x=y-1
+     t = 1 + (x/2) - 0.5*((x/2)^2) + 0.5*((x/2)^3) - 0.625*((x/2)^4) + 0.875*((x/2)^5)
+     ut=t*(1/sqrt(2))*512
+
+     or:
+
+     in=73632
+     in2=73632*2^14
+     y= in2/2^31
+     x=y-1
+     t = 1 + (x/2) - 0.5*((x/2)^2) + 0.5*((x/2)^3) - 0.625*((x/2)^4) + 0.875*((x/2)^5)
+     ut=t*(1/sqrt(2))
+     ut2=ut*2^9
+
+     which gives:
+
+     in  = 73632
+     in2 = 1206386688
+     y   = 0.56176757812500
+     x   = -0.43823242187500
+     t   = 0.74973506527313
+     ut  = 0.53014274874797
+     ut2 = 2.714330873589594e+002
+
+     or:
+
+     in=73632
+     in2=73632*2^14
+     y=in2/2
+     x=y-2^30
+     x_half=x/2^31
+     t = 1 + (x_half) - 0.5*((x_half)^2) + 0.5*((x_half)^3) - 0.625*((x_half)^4)
+         + 0.875*((x_half)^5)
+     ut=t*(1/sqrt(2))
+     ut2=ut*2^9
+
+     which gives:
+
+     in  = 73632
+     in2 = 1206386688
+     y   = 603193344
+     x   = -470548480
+     x_half =  -0.21911621093750
+     t   = 0.74973506527313
+     ut  = 0.53014274874797
+     ut2 = 2.714330873589594e+002
+
+     */
+
+    WebRtc_Word16 x_norm, nshift, t16, sh;
+    WebRtc_Word32 A;
+
+    WebRtc_Word16 k_sqrt_2 = 23170; // 1/sqrt2 (==5a82)
+
+    A = value;
+
+    if (A == 0)
+        return (WebRtc_Word32)0; // sqrt(0) = 0
+
+    sh = WebRtcSpl_NormW32(A); // # shifts to normalize A
+    A = WEBRTC_SPL_LSHIFT_W32(A, sh); // Normalize A
+    if (A < (WEBRTC_SPL_WORD32_MAX - 32767))
+    {
+        A = A + ((WebRtc_Word32)32768); // Round off bit
+    } else
+    {
+        A = WEBRTC_SPL_WORD32_MAX;
+    }
+
+    x_norm = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(A, 16); // x_norm = AH
+
+    nshift = WEBRTC_SPL_RSHIFT_W16(sh, 1); // nshift = sh>>1
+    nshift = -nshift; // Negate the power for later de-normalization
+
+    A = (WebRtc_Word32)WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)x_norm, 16);
+    A = WEBRTC_SPL_ABS_W32(A); // A = abs(x_norm<<16)
+    A = WebRtcSpl_SqrtLocal(A); // A = sqrt(A)
+
+    if ((-2 * nshift) == sh)
+    { // Even shift value case
+
+        t16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(A, 16); // t16 = AH
+
+        A = WEBRTC_SPL_MUL_16_16(k_sqrt_2, t16) * 2; // A = 1/sqrt(2)*t16
+        A = A + ((WebRtc_Word32)32768); // Round off
+        A = A & ((WebRtc_Word32)0x7fff0000); // Round off
+
+        A = WEBRTC_SPL_RSHIFT_W32(A, 15); // A = A>>16
+
+    } else
+    {
+        A = WEBRTC_SPL_RSHIFT_W32(A, 16); // A = A>>16
+    }
+
+    A = A & ((WebRtc_Word32)0x0000ffff);
+    A = (WebRtc_Word32)WEBRTC_SPL_SHIFT_W32(A, nshift); // De-normalize the result
+
+    return A;
+}
diff --git a/trunk/src/common_audio/signal_processing/spl_sqrt_floor.c b/trunk/src/common_audio/signal_processing/spl_sqrt_floor.c
new file mode 100644
index 0000000..f0e8ae2
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/spl_sqrt_floor.c
@@ -0,0 +1,54 @@
+/*
+ * Written by Wilco Dijkstra, 1996. Refer to file LICENSE under
+ * trunk/third_party_mods/sqrt_floor.
+ *
+ * Minor modifications in code style for WebRTC, 2012.
+ */
+
+#include "signal_processing_library.h"
+
+/*
+ * Algorithm:
+ * Successive approximation of the equation (root + delta) ^ 2 = N
+ * until delta < 1. If delta < 1 we have the integer part of SQRT (N).
+ * Use delta = 2^i for i = 15 .. 0.
+ *
+ * Output precision is 16 bits. Note for large input values (close to
+ * 0x7FFFFFFF), bit 15 (the highest bit of the low 16-bit half word)
+ * contains the MSB information (a non-sign value). Do with caution
+ * if you need to cast the output to int16_t type.
+ *
+ * If the input value is negative, it returns 0.
+ */
+
+#define WEBRTC_SPL_SQRT_ITER(N)                 \
+  try1 = root + (1 << (N));                     \
+  if (value >= try1 << (N))                     \
+  {                                             \
+    value -= try1 << (N);                       \
+    root |= 2 << (N);                           \
+  }
+
+int32_t WebRtcSpl_SqrtFloor(int32_t value)
+{
+  int32_t root = 0, try1;
+
+  WEBRTC_SPL_SQRT_ITER (15);
+  WEBRTC_SPL_SQRT_ITER (14);
+  WEBRTC_SPL_SQRT_ITER (13);
+  WEBRTC_SPL_SQRT_ITER (12);
+  WEBRTC_SPL_SQRT_ITER (11);
+  WEBRTC_SPL_SQRT_ITER (10);
+  WEBRTC_SPL_SQRT_ITER ( 9);
+  WEBRTC_SPL_SQRT_ITER ( 8);
+  WEBRTC_SPL_SQRT_ITER ( 7);
+  WEBRTC_SPL_SQRT_ITER ( 6);
+  WEBRTC_SPL_SQRT_ITER ( 5);
+  WEBRTC_SPL_SQRT_ITER ( 4);
+  WEBRTC_SPL_SQRT_ITER ( 3);
+  WEBRTC_SPL_SQRT_ITER ( 2);
+  WEBRTC_SPL_SQRT_ITER ( 1);
+  WEBRTC_SPL_SQRT_ITER ( 0);
+
+  return root >> 1;
+}
diff --git a/trunk/src/common_audio/signal_processing/spl_sqrt_floor.s b/trunk/src/common_audio/signal_processing/spl_sqrt_floor.s
new file mode 100644
index 0000000..cfd9ed0
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/spl_sqrt_floor.s
@@ -0,0 +1,88 @@
+@ Written by Wilco Dijkstra, 1996. Refer to file LICENSE under
+@ trunk/third_party_mods/sqrt_floor.
+@
+@ Minor modifications in code style for WebRTC, 2012.
+@ Output is bit-exact with the reference C code in spl_sqrt_floor.c.
+
+@ Input :             r0 32 bit unsigned integer
+@ Output:             r0 = INT (SQRT (r0)), precision is 16 bits
+@ Registers touched:  r1, r2
+
+.global WebRtcSpl_SqrtFloor
+
+.align  2
+WebRtcSpl_SqrtFloor:
+.fnstart
+  mov    r1, #3 << 30
+  mov    r2, #1 << 30
+
+  @ unroll for i = 0 .. 15
+
+  cmp    r0, r2, ror #2 * 0
+  subhs  r0, r0, r2, ror #2 * 0
+  adc    r2, r1, r2, lsl #1
+
+  cmp    r0, r2, ror #2 * 1
+  subhs  r0, r0, r2, ror #2 * 1
+  adc    r2, r1, r2, lsl #1
+
+  cmp    r0, r2, ror #2 * 2
+  subhs  r0, r0, r2, ror #2 * 2
+  adc    r2, r1, r2, lsl #1
+
+  cmp    r0, r2, ror #2 * 3
+  subhs  r0, r0, r2, ror #2 * 3
+  adc    r2, r1, r2, lsl #1
+
+  cmp    r0, r2, ror #2 * 4
+  subhs  r0, r0, r2, ror #2 * 4
+  adc    r2, r1, r2, lsl #1
+
+  cmp    r0, r2, ror #2 * 5
+  subhs  r0, r0, r2, ror #2 * 5
+  adc    r2, r1, r2, lsl #1
+
+  cmp    r0, r2, ror #2 * 6
+  subhs  r0, r0, r2, ror #2 * 6
+  adc    r2, r1, r2, lsl #1
+
+  cmp    r0, r2, ror #2 * 7
+  subhs  r0, r0, r2, ror #2 * 7
+  adc    r2, r1, r2, lsl #1
+
+  cmp    r0, r2, ror #2 * 8
+  subhs  r0, r0, r2, ror #2 * 8
+  adc    r2, r1, r2, lsl #1
+
+  cmp    r0, r2, ror #2 * 9
+  subhs  r0, r0, r2, ror #2 * 9
+  adc    r2, r1, r2, lsl #1
+
+  cmp    r0, r2, ror #2 * 10
+  subhs  r0, r0, r2, ror #2 * 10
+  adc    r2, r1, r2, lsl #1
+
+  cmp    r0, r2, ror #2 * 11
+  subhs  r0, r0, r2, ror #2 * 11
+  adc    r2, r1, r2, lsl #1
+
+  cmp    r0, r2, ror #2 * 12
+  subhs  r0, r0, r2, ror #2 * 12
+  adc    r2, r1, r2, lsl #1
+
+  cmp    r0, r2, ror #2 * 13
+  subhs  r0, r0, r2, ror #2 * 13
+  adc    r2, r1, r2, lsl #1
+
+  cmp    r0, r2, ror #2 * 14
+  subhs  r0, r0, r2, ror #2 * 14
+  adc    r2, r1, r2, lsl #1
+
+  cmp    r0, r2, ror #2 * 15
+  subhs  r0, r0, r2, ror #2 * 15
+  adc    r2, r1, r2, lsl #1
+
+  bic    r0, r2, #3 << 30  @ for rounding add: cmp r0, r2  adc r2, #1
+  bx lr
+
+.fnend
diff --git a/trunk/src/common_audio/signal_processing/spl_version.c b/trunk/src/common_audio/signal_processing/spl_version.c
new file mode 100644
index 0000000..936925e
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/spl_version.c
@@ -0,0 +1,25 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains the function WebRtcSpl_get_version().
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include <string.h>
+#include "signal_processing_library.h"
+
+WebRtc_Word16 WebRtcSpl_get_version(char* version, WebRtc_Word16 length_in_bytes)
+{
+    strncpy(version, "1.2.0", length_in_bytes);
+    return 0;
+}
diff --git a/trunk/src/common_audio/signal_processing/splitting_filter.c b/trunk/src/common_audio/signal_processing/splitting_filter.c
new file mode 100644
index 0000000..f1acf67
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/splitting_filter.c
@@ -0,0 +1,198 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * This file contains the splitting filter functions.
+ *
+ */
+
+#include "signal_processing_library.h"
+
+// Number of samples in a low/high-band frame.
+enum
+{
+    kBandFrameLength = 160
+};
+
+// QMF filter coefficients in Q16.
+static const WebRtc_UWord16 WebRtcSpl_kAllPassFilter1[3] = {6418, 36982, 57261};
+static const WebRtc_UWord16 WebRtcSpl_kAllPassFilter2[3] = {21333, 49062, 63010};
+
+///////////////////////////////////////////////////////////////////////////////////////////////
+// WebRtcSpl_AllPassQMF(...)
+//
+// Allpass filter used by the analysis and synthesis parts of the QMF filter.
+//
+// Input:
+//    - in_data             : Input data sequence (Q10)
+//    - data_length         : Length of data sequence (>2)
+//    - filter_coefficients : Filter coefficients (length 3, Q16)
+//
+// Input & Output:
+//    - filter_state        : Filter state (length 6, Q10).
+//
+// Output:
+//    - out_data            : Output data sequence (Q10), length equal to
+//                            |data_length|
+//
+
+void WebRtcSpl_AllPassQMF(WebRtc_Word32* in_data, const WebRtc_Word16 data_length,
+                          WebRtc_Word32* out_data, const WebRtc_UWord16* filter_coefficients,
+                          WebRtc_Word32* filter_state)
+{
+    // The procedure is to filter the input with three first order all pass filters
+    // (cascade operations).
+    //
+    //         a_3 + q^-1    a_2 + q^-1    a_1 + q^-1
+    // y[n] =  -----------   -----------   -----------   x[n]
+    //         1 + a_3q^-1   1 + a_2q^-1   1 + a_1q^-1
+    //
+    // The input vector |filter_coefficients| includes these three filter coefficients.
+    // The filter state contains the in_data state, in_data[-1], followed by
+    // the out_data state, out_data[-1]. This is repeated for each cascade.
+    // The first cascade filter will filter the |in_data| and store the output in
+    // |out_data|. The second will the take the |out_data| as input and make an
+    // intermediate storage in |in_data|, to save memory. The third, and final, cascade
+    // filter operation takes the |in_data| (which is the output from the previous cascade
+    // filter) and store the output in |out_data|.
+    // Note that the input vector values are changed during the process.
+    WebRtc_Word16 k;
+    WebRtc_Word32 diff;
+    // First all-pass cascade; filter from in_data to out_data.
+
+    // Let y_i[n] indicate the output of cascade filter i (with filter coefficient a_i) at
+    // vector position n. Then the final output will be y[n] = y_3[n]
+
+    // First loop, use the states stored in memory.
+    // "diff" should be safe from wrap around since max values are 2^25
+    diff = WEBRTC_SPL_SUB_SAT_W32(in_data[0], filter_state[1]); // = (x[0] - y_1[-1])
+    // y_1[0] =  x[-1] + a_1 * (x[0] - y_1[-1])
+    out_data[0] = WEBRTC_SPL_SCALEDIFF32(filter_coefficients[0], diff, filter_state[0]);
+
+    // For the remaining loops, use previous values.
+    for (k = 1; k < data_length; k++)
+    {
+        diff = WEBRTC_SPL_SUB_SAT_W32(in_data[k], out_data[k - 1]); // = (x[n] - y_1[n-1])
+        // y_1[n] =  x[n-1] + a_1 * (x[n] - y_1[n-1])
+        out_data[k] = WEBRTC_SPL_SCALEDIFF32(filter_coefficients[0], diff, in_data[k - 1]);
+    }
+
+    // Update states.
+    filter_state[0] = in_data[data_length - 1]; // x[N-1], becomes x[-1] next time
+    filter_state[1] = out_data[data_length - 1]; // y_1[N-1], becomes y_1[-1] next time
+
+    // Second all-pass cascade; filter from out_data to in_data.
+    diff = WEBRTC_SPL_SUB_SAT_W32(out_data[0], filter_state[3]); // = (y_1[0] - y_2[-1])
+    // y_2[0] =  y_1[-1] + a_2 * (y_1[0] - y_2[-1])
+    in_data[0] = WEBRTC_SPL_SCALEDIFF32(filter_coefficients[1], diff, filter_state[2]);
+    for (k = 1; k < data_length; k++)
+    {
+        diff = WEBRTC_SPL_SUB_SAT_W32(out_data[k], in_data[k - 1]); // =(y_1[n] - y_2[n-1])
+        // y_2[0] =  y_1[-1] + a_2 * (y_1[0] - y_2[-1])
+        in_data[k] = WEBRTC_SPL_SCALEDIFF32(filter_coefficients[1], diff, out_data[k-1]);
+    }
+
+    filter_state[2] = out_data[data_length - 1]; // y_1[N-1], becomes y_1[-1] next time
+    filter_state[3] = in_data[data_length - 1]; // y_2[N-1], becomes y_2[-1] next time
+
+    // Third all-pass cascade; filter from in_data to out_data.
+    diff = WEBRTC_SPL_SUB_SAT_W32(in_data[0], filter_state[5]); // = (y_2[0] - y[-1])
+    // y[0] =  y_2[-1] + a_3 * (y_2[0] - y[-1])
+    out_data[0] = WEBRTC_SPL_SCALEDIFF32(filter_coefficients[2], diff, filter_state[4]);
+    for (k = 1; k < data_length; k++)
+    {
+        diff = WEBRTC_SPL_SUB_SAT_W32(in_data[k], out_data[k - 1]); // = (y_2[n] - y[n-1])
+        // y[n] =  y_2[n-1] + a_3 * (y_2[n] - y[n-1])
+        out_data[k] = WEBRTC_SPL_SCALEDIFF32(filter_coefficients[2], diff, in_data[k-1]);
+    }
+    filter_state[4] = in_data[data_length - 1]; // y_2[N-1], becomes y_2[-1] next time
+    filter_state[5] = out_data[data_length - 1]; // y[N-1], becomes y[-1] next time
+}
+
+void WebRtcSpl_AnalysisQMF(const WebRtc_Word16* in_data, WebRtc_Word16* low_band,
+                           WebRtc_Word16* high_band, WebRtc_Word32* filter_state1,
+                           WebRtc_Word32* filter_state2)
+{
+    WebRtc_Word16 i;
+    WebRtc_Word16 k;
+    WebRtc_Word32 tmp;
+    WebRtc_Word32 half_in1[kBandFrameLength];
+    WebRtc_Word32 half_in2[kBandFrameLength];
+    WebRtc_Word32 filter1[kBandFrameLength];
+    WebRtc_Word32 filter2[kBandFrameLength];
+
+    // Split even and odd samples. Also shift them to Q10.
+    for (i = 0, k = 0; i < kBandFrameLength; i++, k += 2)
+    {
+        half_in2[i] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)in_data[k], 10);
+        half_in1[i] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)in_data[k + 1], 10);
+    }
+
+    // All pass filter even and odd samples, independently.
+    WebRtcSpl_AllPassQMF(half_in1, kBandFrameLength, filter1, WebRtcSpl_kAllPassFilter1,
+                         filter_state1);
+    WebRtcSpl_AllPassQMF(half_in2, kBandFrameLength, filter2, WebRtcSpl_kAllPassFilter2,
+                         filter_state2);
+
+    // Take the sum and difference of filtered version of odd and even
+    // branches to get upper & lower band.
+    for (i = 0; i < kBandFrameLength; i++)
+    {
+        tmp = filter1[i] + filter2[i] + 1024;
+        tmp = WEBRTC_SPL_RSHIFT_W32(tmp, 11);
+        low_band[i] = WebRtcSpl_SatW32ToW16(tmp);
+
+        tmp = filter1[i] - filter2[i] + 1024;
+        tmp = WEBRTC_SPL_RSHIFT_W32(tmp, 11);
+        high_band[i] = WebRtcSpl_SatW32ToW16(tmp);
+    }
+}
+
+void WebRtcSpl_SynthesisQMF(const WebRtc_Word16* low_band, const WebRtc_Word16* high_band,
+                            WebRtc_Word16* out_data, WebRtc_Word32* filter_state1,
+                            WebRtc_Word32* filter_state2)
+{
+    WebRtc_Word32 tmp;
+    WebRtc_Word32 half_in1[kBandFrameLength];
+    WebRtc_Word32 half_in2[kBandFrameLength];
+    WebRtc_Word32 filter1[kBandFrameLength];
+    WebRtc_Word32 filter2[kBandFrameLength];
+    WebRtc_Word16 i;
+    WebRtc_Word16 k;
+
+    // Obtain the sum and difference channels out of upper and lower-band channels.
+    // Also shift to Q10 domain.
+    for (i = 0; i < kBandFrameLength; i++)
+    {
+        tmp = (WebRtc_Word32)low_band[i] + (WebRtc_Word32)high_band[i];
+        half_in1[i] = WEBRTC_SPL_LSHIFT_W32(tmp, 10);
+        tmp = (WebRtc_Word32)low_band[i] - (WebRtc_Word32)high_band[i];
+        half_in2[i] = WEBRTC_SPL_LSHIFT_W32(tmp, 10);
+    }
+
+    // all-pass filter the sum and difference channels
+    WebRtcSpl_AllPassQMF(half_in1, kBandFrameLength, filter1, WebRtcSpl_kAllPassFilter2,
+                         filter_state1);
+    WebRtcSpl_AllPassQMF(half_in2, kBandFrameLength, filter2, WebRtcSpl_kAllPassFilter1,
+                         filter_state2);
+
+    // The filtered signals are even and odd samples of the output. Combine
+    // them. The signals are Q10 should shift them back to Q0 and take care of
+    // saturation.
+    for (i = 0, k = 0; i < kBandFrameLength; i++)
+    {
+        tmp = WEBRTC_SPL_RSHIFT_W32(filter2[i] + 512, 10);
+        out_data[k++] = WebRtcSpl_SatW32ToW16(tmp);
+
+        tmp = WEBRTC_SPL_RSHIFT_W32(filter1[i] + 512, 10);
+        out_data[k++] = WebRtcSpl_SatW32ToW16(tmp);
+    }
+
+}
diff --git a/trunk/src/common_audio/signal_processing/sqrt_of_one_minus_x_squared.c b/trunk/src/common_audio/signal_processing/sqrt_of_one_minus_x_squared.c
new file mode 100644
index 0000000..9fb2c73
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/sqrt_of_one_minus_x_squared.c
@@ -0,0 +1,35 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains the function WebRtcSpl_SqrtOfOneMinusXSquared().
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "signal_processing_library.h"
+
+void WebRtcSpl_SqrtOfOneMinusXSquared(WebRtc_Word16 *xQ15, int vector_length,
+                                      WebRtc_Word16 *yQ15)
+{
+    WebRtc_Word32 sq;
+    int m;
+    WebRtc_Word16 tmp;
+
+    for (m = 0; m < vector_length; m++)
+    {
+        tmp = xQ15[m];
+        sq = WEBRTC_SPL_MUL_16_16(tmp, tmp); // x^2 in Q30
+        sq = 1073741823 - sq; // 1-x^2, where 1 ~= 0.99999999906 is 1073741823 in Q30
+        sq = WebRtcSpl_Sqrt(sq); // sqrt(1-x^2) in Q15
+        yQ15[m] = (WebRtc_Word16)sq;
+    }
+}
diff --git a/trunk/src/common_audio/signal_processing/vector_scaling_operations.c b/trunk/src/common_audio/signal_processing/vector_scaling_operations.c
new file mode 100644
index 0000000..20d239c
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/vector_scaling_operations.c
@@ -0,0 +1,151 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains implementations of the functions
+ * WebRtcSpl_VectorBitShiftW16()
+ * WebRtcSpl_VectorBitShiftW32()
+ * WebRtcSpl_VectorBitShiftW32ToW16()
+ * WebRtcSpl_ScaleVector()
+ * WebRtcSpl_ScaleVectorWithSat()
+ * WebRtcSpl_ScaleAndAddVectors()
+ *
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "signal_processing_library.h"
+
+void WebRtcSpl_VectorBitShiftW16(WebRtc_Word16 *res,
+                             WebRtc_Word16 length,
+                             G_CONST WebRtc_Word16 *in,
+                             WebRtc_Word16 right_shifts)
+{
+    int i;
+
+    if (right_shifts > 0)
+    {
+        for (i = length; i > 0; i--)
+        {
+            (*res++) = ((*in++) >> right_shifts);
+        }
+    } else
+    {
+        for (i = length; i > 0; i--)
+        {
+            (*res++) = ((*in++) << (-right_shifts));
+        }
+    }
+}
+
+void WebRtcSpl_VectorBitShiftW32(WebRtc_Word32 *out_vector,
+                             WebRtc_Word16 vector_length,
+                             G_CONST WebRtc_Word32 *in_vector,
+                             WebRtc_Word16 right_shifts)
+{
+    int i;
+
+    if (right_shifts > 0)
+    {
+        for (i = vector_length; i > 0; i--)
+        {
+            (*out_vector++) = ((*in_vector++) >> right_shifts);
+        }
+    } else
+    {
+        for (i = vector_length; i > 0; i--)
+        {
+            (*out_vector++) = ((*in_vector++) << (-right_shifts));
+        }
+    }
+}
+
+void WebRtcSpl_VectorBitShiftW32ToW16(WebRtc_Word16 *res,
+                                  WebRtc_Word16 length,
+                                  G_CONST WebRtc_Word32 *in,
+                                  WebRtc_Word16 right_shifts)
+{
+    int i;
+
+    if (right_shifts >= 0)
+    {
+        for (i = length; i > 0; i--)
+        {
+            (*res++) = (WebRtc_Word16)((*in++) >> right_shifts);
+        }
+    } else
+    {
+        WebRtc_Word16 left_shifts = -right_shifts;
+        for (i = length; i > 0; i--)
+        {
+            (*res++) = (WebRtc_Word16)((*in++) << left_shifts);
+        }
+    }
+}
+
+void WebRtcSpl_ScaleVector(G_CONST WebRtc_Word16 *in_vector, WebRtc_Word16 *out_vector,
+                           WebRtc_Word16 gain, WebRtc_Word16 in_vector_length,
+                           WebRtc_Word16 right_shifts)
+{
+    // Performs vector operation: out_vector = (gain*in_vector)>>right_shifts
+    int i;
+    G_CONST WebRtc_Word16 *inptr;
+    WebRtc_Word16 *outptr;
+
+    inptr = in_vector;
+    outptr = out_vector;
+
+    for (i = 0; i < in_vector_length; i++)
+    {
+        (*outptr++) = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(*inptr++, gain, right_shifts);
+    }
+}
+
+void WebRtcSpl_ScaleVectorWithSat(G_CONST WebRtc_Word16 *in_vector, WebRtc_Word16 *out_vector,
+                                 WebRtc_Word16 gain, WebRtc_Word16 in_vector_length,
+                                 WebRtc_Word16 right_shifts)
+{
+    // Performs vector operation: out_vector = (gain*in_vector)>>right_shifts
+    int i;
+    WebRtc_Word32 tmpW32;
+    G_CONST WebRtc_Word16 *inptr;
+    WebRtc_Word16 *outptr;
+
+    inptr = in_vector;
+    outptr = out_vector;
+
+    for (i = 0; i < in_vector_length; i++)
+    {
+        tmpW32 = WEBRTC_SPL_MUL_16_16_RSFT(*inptr++, gain, right_shifts);
+        (*outptr++) = WebRtcSpl_SatW32ToW16(tmpW32);
+    }
+}
+
+void WebRtcSpl_ScaleAndAddVectors(G_CONST WebRtc_Word16 *in1, WebRtc_Word16 gain1, int shift1,
+                                  G_CONST WebRtc_Word16 *in2, WebRtc_Word16 gain2, int shift2,
+                                  WebRtc_Word16 *out, int vector_length)
+{
+    // Performs vector operation: out = (gain1*in1)>>shift1 + (gain2*in2)>>shift2
+    int i;
+    G_CONST WebRtc_Word16 *in1ptr;
+    G_CONST WebRtc_Word16 *in2ptr;
+    WebRtc_Word16 *outptr;
+
+    in1ptr = in1;
+    in2ptr = in2;
+    outptr = out;
+
+    for (i = 0; i < vector_length; i++)
+    {
+        (*outptr++) = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(gain1, *in1ptr++, shift1)
+                + (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(gain2, *in2ptr++, shift2);
+    }
+}
diff --git a/trunk/src/common_audio/signal_processing/webrtc_fft_t_1024_8.c b/trunk/src/common_audio/signal_processing/webrtc_fft_t_1024_8.c
new file mode 100644
index 0000000..b587380
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/webrtc_fft_t_1024_8.c
@@ -0,0 +1,704 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains the Q14 radix-8 tables used in ARM9e optimizations.
+ *
+ */
+
+extern const int s_Q14S_8;
+const int s_Q14S_8 = 1024;
+extern const unsigned short t_Q14S_8[2032];
+const unsigned short t_Q14S_8[2032] = {
+  0x4000,0x0000 ,0x4000,0x0000 ,0x4000,0x0000 ,
+  0x22a3,0x187e ,0x3249,0x0c7c ,0x11a8,0x238e ,
+  0x0000,0x2d41 ,0x22a3,0x187e ,0xdd5d,0x3b21 ,
+  0xdd5d,0x3b21 ,0x11a8,0x238e ,0xb4be,0x3ec5 ,
+  0xc000,0x4000 ,0x0000,0x2d41 ,0xa57e,0x2d41 ,
+  0xac61,0x3b21 ,0xee58,0x3537 ,0xb4be,0x0c7c ,
+  0xa57e,0x2d41 ,0xdd5d,0x3b21 ,0xdd5d,0xe782 ,
+  0xac61,0x187e ,0xcdb7,0x3ec5 ,0x11a8,0xcac9 ,
+  0x4000,0x0000 ,0x4000,0x0000 ,0x4000,0x0000 ,
+  0x396b,0x0646 ,0x3cc8,0x0324 ,0x35eb,0x0964 ,
+  0x3249,0x0c7c ,0x396b,0x0646 ,0x2aaa,0x1294 ,
+  0x2aaa,0x1294 ,0x35eb,0x0964 ,0x1e7e,0x1b5d ,
+  0x22a3,0x187e ,0x3249,0x0c7c ,0x11a8,0x238e ,
+  0x1a46,0x1e2b ,0x2e88,0x0f8d ,0x0471,0x2afb ,
+  0x11a8,0x238e ,0x2aaa,0x1294 ,0xf721,0x3179 ,
+  0x08df,0x289a ,0x26b3,0x1590 ,0xea02,0x36e5 ,
+  0x0000,0x2d41 ,0x22a3,0x187e ,0xdd5d,0x3b21 ,
+  0xf721,0x3179 ,0x1e7e,0x1b5d ,0xd178,0x3e15 ,
+  0xee58,0x3537 ,0x1a46,0x1e2b ,0xc695,0x3fb1 ,
+  0xe5ba,0x3871 ,0x15fe,0x20e7 ,0xbcf0,0x3fec ,
+  0xdd5d,0x3b21 ,0x11a8,0x238e ,0xb4be,0x3ec5 ,
+  0xd556,0x3d3f ,0x0d48,0x2620 ,0xae2e,0x3c42 ,
+  0xcdb7,0x3ec5 ,0x08df,0x289a ,0xa963,0x3871 ,
+  0xc695,0x3fb1 ,0x0471,0x2afb ,0xa678,0x3368 ,
+  0xc000,0x4000 ,0x0000,0x2d41 ,0xa57e,0x2d41 ,
+  0xba09,0x3fb1 ,0xfb8f,0x2f6c ,0xa678,0x2620 ,
+  0xb4be,0x3ec5 ,0xf721,0x3179 ,0xa963,0x1e2b ,
+  0xb02d,0x3d3f ,0xf2b8,0x3368 ,0xae2e,0x1590 ,
+  0xac61,0x3b21 ,0xee58,0x3537 ,0xb4be,0x0c7c ,
+  0xa963,0x3871 ,0xea02,0x36e5 ,0xbcf0,0x0324 ,
+  0xa73b,0x3537 ,0xe5ba,0x3871 ,0xc695,0xf9ba ,
+  0xa5ed,0x3179 ,0xe182,0x39db ,0xd178,0xf073 ,
+  0xa57e,0x2d41 ,0xdd5d,0x3b21 ,0xdd5d,0xe782 ,
+  0xa5ed,0x289a ,0xd94d,0x3c42 ,0xea02,0xdf19 ,
+  0xa73b,0x238e ,0xd556,0x3d3f ,0xf721,0xd766 ,
+  0xa963,0x1e2b ,0xd178,0x3e15 ,0x0471,0xd094 ,
+  0xac61,0x187e ,0xcdb7,0x3ec5 ,0x11a8,0xcac9 ,
+  0xb02d,0x1294 ,0xca15,0x3f4f ,0x1e7e,0xc625 ,
+  0xb4be,0x0c7c ,0xc695,0x3fb1 ,0x2aaa,0xc2c1 ,
+  0xba09,0x0646 ,0xc338,0x3fec ,0x35eb,0xc0b1 ,
+  0x4000,0x0000 ,0x4000,0x0000 ,0x4000,0x0000 ,
+  0x3e69,0x0192 ,0x3f36,0x00c9 ,0x3d9a,0x025b ,
+  0x3cc8,0x0324 ,0x3e69,0x0192 ,0x3b1e,0x04b5 ,
+  0x3b1e,0x04b5 ,0x3d9a,0x025b ,0x388e,0x070e ,
+  0x396b,0x0646 ,0x3cc8,0x0324 ,0x35eb,0x0964 ,
+  0x37af,0x07d6 ,0x3bf4,0x03ed ,0x3334,0x0bb7 ,
+  0x35eb,0x0964 ,0x3b1e,0x04b5 ,0x306c,0x0e06 ,
+  0x341e,0x0af1 ,0x3a46,0x057e ,0x2d93,0x1050 ,
+  0x3249,0x0c7c ,0x396b,0x0646 ,0x2aaa,0x1294 ,
+  0x306c,0x0e06 ,0x388e,0x070e ,0x27b3,0x14d2 ,
+  0x2e88,0x0f8d ,0x37af,0x07d6 ,0x24ae,0x1709 ,
+  0x2c9d,0x1112 ,0x36ce,0x089d ,0x219c,0x1937 ,
+  0x2aaa,0x1294 ,0x35eb,0x0964 ,0x1e7e,0x1b5d ,
+  0x28b2,0x1413 ,0x3505,0x0a2b ,0x1b56,0x1d79 ,
+  0x26b3,0x1590 ,0x341e,0x0af1 ,0x1824,0x1f8c ,
+  0x24ae,0x1709 ,0x3334,0x0bb7 ,0x14ea,0x2193 ,
+  0x22a3,0x187e ,0x3249,0x0c7c ,0x11a8,0x238e ,
+  0x2093,0x19ef ,0x315b,0x0d41 ,0x0e61,0x257e ,
+  0x1e7e,0x1b5d ,0x306c,0x0e06 ,0x0b14,0x2760 ,
+  0x1c64,0x1cc6 ,0x2f7b,0x0eca ,0x07c4,0x2935 ,
+  0x1a46,0x1e2b ,0x2e88,0x0f8d ,0x0471,0x2afb ,
+  0x1824,0x1f8c ,0x2d93,0x1050 ,0x011c,0x2cb2 ,
+  0x15fe,0x20e7 ,0x2c9d,0x1112 ,0xfdc7,0x2e5a ,
+  0x13d5,0x223d ,0x2ba4,0x11d3 ,0xfa73,0x2ff2 ,
+  0x11a8,0x238e ,0x2aaa,0x1294 ,0xf721,0x3179 ,
+  0x0f79,0x24da ,0x29af,0x1354 ,0xf3d2,0x32ef ,
+  0x0d48,0x2620 ,0x28b2,0x1413 ,0xf087,0x3453 ,
+  0x0b14,0x2760 ,0x27b3,0x14d2 ,0xed41,0x35a5 ,
+  0x08df,0x289a ,0x26b3,0x1590 ,0xea02,0x36e5 ,
+  0x06a9,0x29ce ,0x25b1,0x164c ,0xe6cb,0x3812 ,
+  0x0471,0x2afb ,0x24ae,0x1709 ,0xe39c,0x392b ,
+  0x0239,0x2c21 ,0x23a9,0x17c4 ,0xe077,0x3a30 ,
+  0x0000,0x2d41 ,0x22a3,0x187e ,0xdd5d,0x3b21 ,
+  0xfdc7,0x2e5a ,0x219c,0x1937 ,0xda4f,0x3bfd ,
+  0xfb8f,0x2f6c ,0x2093,0x19ef ,0xd74e,0x3cc5 ,
+  0xf957,0x3076 ,0x1f89,0x1aa7 ,0xd45c,0x3d78 ,
+  0xf721,0x3179 ,0x1e7e,0x1b5d ,0xd178,0x3e15 ,
+  0xf4ec,0x3274 ,0x1d72,0x1c12 ,0xcea5,0x3e9d ,
+  0xf2b8,0x3368 ,0x1c64,0x1cc6 ,0xcbe2,0x3f0f ,
+  0xf087,0x3453 ,0x1b56,0x1d79 ,0xc932,0x3f6b ,
+  0xee58,0x3537 ,0x1a46,0x1e2b ,0xc695,0x3fb1 ,
+  0xec2b,0x3612 ,0x1935,0x1edc ,0xc40c,0x3fe1 ,
+  0xea02,0x36e5 ,0x1824,0x1f8c ,0xc197,0x3ffb ,
+  0xe7dc,0x37b0 ,0x1711,0x203a ,0xbf38,0x3fff ,
+  0xe5ba,0x3871 ,0x15fe,0x20e7 ,0xbcf0,0x3fec ,
+  0xe39c,0x392b ,0x14ea,0x2193 ,0xbabf,0x3fc4 ,
+  0xe182,0x39db ,0x13d5,0x223d ,0xb8a6,0x3f85 ,
+  0xdf6d,0x3a82 ,0x12bf,0x22e7 ,0xb6a5,0x3f30 ,
+  0xdd5d,0x3b21 ,0x11a8,0x238e ,0xb4be,0x3ec5 ,
+  0xdb52,0x3bb6 ,0x1091,0x2435 ,0xb2f2,0x3e45 ,
+  0xd94d,0x3c42 ,0x0f79,0x24da ,0xb140,0x3daf ,
+  0xd74e,0x3cc5 ,0x0e61,0x257e ,0xafa9,0x3d03 ,
+  0xd556,0x3d3f ,0x0d48,0x2620 ,0xae2e,0x3c42 ,
+  0xd363,0x3daf ,0x0c2e,0x26c1 ,0xacd0,0x3b6d ,
+  0xd178,0x3e15 ,0x0b14,0x2760 ,0xab8e,0x3a82 ,
+  0xcf94,0x3e72 ,0x09fa,0x27fe ,0xaa6a,0x3984 ,
+  0xcdb7,0x3ec5 ,0x08df,0x289a ,0xa963,0x3871 ,
+  0xcbe2,0x3f0f ,0x07c4,0x2935 ,0xa87b,0x374b ,
+  0xca15,0x3f4f ,0x06a9,0x29ce ,0xa7b1,0x3612 ,
+  0xc851,0x3f85 ,0x058d,0x2a65 ,0xa705,0x34c6 ,
+  0xc695,0x3fb1 ,0x0471,0x2afb ,0xa678,0x3368 ,
+  0xc4e2,0x3fd4 ,0x0355,0x2b8f ,0xa60b,0x31f8 ,
+  0xc338,0x3fec ,0x0239,0x2c21 ,0xa5bc,0x3076 ,
+  0xc197,0x3ffb ,0x011c,0x2cb2 ,0xa58d,0x2ee4 ,
+  0xc000,0x4000 ,0x0000,0x2d41 ,0xa57e,0x2d41 ,
+  0xbe73,0x3ffb ,0xfee4,0x2dcf ,0xa58d,0x2b8f ,
+  0xbcf0,0x3fec ,0xfdc7,0x2e5a ,0xa5bc,0x29ce ,
+  0xbb77,0x3fd4 ,0xfcab,0x2ee4 ,0xa60b,0x27fe ,
+  0xba09,0x3fb1 ,0xfb8f,0x2f6c ,0xa678,0x2620 ,
+  0xb8a6,0x3f85 ,0xfa73,0x2ff2 ,0xa705,0x2435 ,
+  0xb74d,0x3f4f ,0xf957,0x3076 ,0xa7b1,0x223d ,
+  0xb600,0x3f0f ,0xf83c,0x30f9 ,0xa87b,0x203a ,
+  0xb4be,0x3ec5 ,0xf721,0x3179 ,0xa963,0x1e2b ,
+  0xb388,0x3e72 ,0xf606,0x31f8 ,0xaa6a,0x1c12 ,
+  0xb25e,0x3e15 ,0xf4ec,0x3274 ,0xab8e,0x19ef ,
+  0xb140,0x3daf ,0xf3d2,0x32ef ,0xacd0,0x17c4 ,
+  0xb02d,0x3d3f ,0xf2b8,0x3368 ,0xae2e,0x1590 ,
+  0xaf28,0x3cc5 ,0xf19f,0x33df ,0xafa9,0x1354 ,
+  0xae2e,0x3c42 ,0xf087,0x3453 ,0xb140,0x1112 ,
+  0xad41,0x3bb6 ,0xef6f,0x34c6 ,0xb2f2,0x0eca ,
+  0xac61,0x3b21 ,0xee58,0x3537 ,0xb4be,0x0c7c ,
+  0xab8e,0x3a82 ,0xed41,0x35a5 ,0xb6a5,0x0a2b ,
+  0xaac8,0x39db ,0xec2b,0x3612 ,0xb8a6,0x07d6 ,
+  0xaa0f,0x392b ,0xeb16,0x367d ,0xbabf,0x057e ,
+  0xa963,0x3871 ,0xea02,0x36e5 ,0xbcf0,0x0324 ,
+  0xa8c5,0x37b0 ,0xe8ef,0x374b ,0xbf38,0x00c9 ,
+  0xa834,0x36e5 ,0xe7dc,0x37b0 ,0xc197,0xfe6e ,
+  0xa7b1,0x3612 ,0xe6cb,0x3812 ,0xc40c,0xfc13 ,
+  0xa73b,0x3537 ,0xe5ba,0x3871 ,0xc695,0xf9ba ,
+  0xa6d3,0x3453 ,0xe4aa,0x38cf ,0xc932,0xf763 ,
+  0xa678,0x3368 ,0xe39c,0x392b ,0xcbe2,0xf50f ,
+  0xa62c,0x3274 ,0xe28e,0x3984 ,0xcea5,0xf2bf ,
+  0xa5ed,0x3179 ,0xe182,0x39db ,0xd178,0xf073 ,
+  0xa5bc,0x3076 ,0xe077,0x3a30 ,0xd45c,0xee2d ,
+  0xa599,0x2f6c ,0xdf6d,0x3a82 ,0xd74e,0xebed ,
+  0xa585,0x2e5a ,0xde64,0x3ad3 ,0xda4f,0xe9b4 ,
+  0xa57e,0x2d41 ,0xdd5d,0x3b21 ,0xdd5d,0xe782 ,
+  0xa585,0x2c21 ,0xdc57,0x3b6d ,0xe077,0xe559 ,
+  0xa599,0x2afb ,0xdb52,0x3bb6 ,0xe39c,0xe33a ,
+  0xa5bc,0x29ce ,0xda4f,0x3bfd ,0xe6cb,0xe124 ,
+  0xa5ed,0x289a ,0xd94d,0x3c42 ,0xea02,0xdf19 ,
+  0xa62c,0x2760 ,0xd84d,0x3c85 ,0xed41,0xdd19 ,
+  0xa678,0x2620 ,0xd74e,0x3cc5 ,0xf087,0xdb26 ,
+  0xa6d3,0x24da ,0xd651,0x3d03 ,0xf3d2,0xd93f ,
+  0xa73b,0x238e ,0xd556,0x3d3f ,0xf721,0xd766 ,
+  0xa7b1,0x223d ,0xd45c,0x3d78 ,0xfa73,0xd59b ,
+  0xa834,0x20e7 ,0xd363,0x3daf ,0xfdc7,0xd3df ,
+  0xa8c5,0x1f8c ,0xd26d,0x3de3 ,0x011c,0xd231 ,
+  0xa963,0x1e2b ,0xd178,0x3e15 ,0x0471,0xd094 ,
+  0xaa0f,0x1cc6 ,0xd085,0x3e45 ,0x07c4,0xcf07 ,
+  0xaac8,0x1b5d ,0xcf94,0x3e72 ,0x0b14,0xcd8c ,
+  0xab8e,0x19ef ,0xcea5,0x3e9d ,0x0e61,0xcc21 ,
+  0xac61,0x187e ,0xcdb7,0x3ec5 ,0x11a8,0xcac9 ,
+  0xad41,0x1709 ,0xcccc,0x3eeb ,0x14ea,0xc983 ,
+  0xae2e,0x1590 ,0xcbe2,0x3f0f ,0x1824,0xc850 ,
+  0xaf28,0x1413 ,0xcafb,0x3f30 ,0x1b56,0xc731 ,
+  0xb02d,0x1294 ,0xca15,0x3f4f ,0x1e7e,0xc625 ,
+  0xb140,0x1112 ,0xc932,0x3f6b ,0x219c,0xc52d ,
+  0xb25e,0x0f8d ,0xc851,0x3f85 ,0x24ae,0xc44a ,
+  0xb388,0x0e06 ,0xc772,0x3f9c ,0x27b3,0xc37b ,
+  0xb4be,0x0c7c ,0xc695,0x3fb1 ,0x2aaa,0xc2c1 ,
+  0xb600,0x0af1 ,0xc5ba,0x3fc4 ,0x2d93,0xc21d ,
+  0xb74d,0x0964 ,0xc4e2,0x3fd4 ,0x306c,0xc18e ,
+  0xb8a6,0x07d6 ,0xc40c,0x3fe1 ,0x3334,0xc115 ,
+  0xba09,0x0646 ,0xc338,0x3fec ,0x35eb,0xc0b1 ,
+  0xbb77,0x04b5 ,0xc266,0x3ff5 ,0x388e,0xc064 ,
+  0xbcf0,0x0324 ,0xc197,0x3ffb ,0x3b1e,0xc02c ,
+  0xbe73,0x0192 ,0xc0ca,0x3fff ,0x3d9a,0xc00b ,
+  0x4000,0x0000 ,0x3f9b,0x0065 ,0x3f36,0x00c9 ,
+  0x3ed0,0x012e ,0x3e69,0x0192 ,0x3e02,0x01f7 ,
+  0x3d9a,0x025b ,0x3d31,0x02c0 ,0x3cc8,0x0324 ,
+  0x3c5f,0x0388 ,0x3bf4,0x03ed ,0x3b8a,0x0451 ,
+  0x3b1e,0x04b5 ,0x3ab2,0x051a ,0x3a46,0x057e ,
+  0x39d9,0x05e2 ,0x396b,0x0646 ,0x38fd,0x06aa ,
+  0x388e,0x070e ,0x381f,0x0772 ,0x37af,0x07d6 ,
+  0x373f,0x0839 ,0x36ce,0x089d ,0x365d,0x0901 ,
+  0x35eb,0x0964 ,0x3578,0x09c7 ,0x3505,0x0a2b ,
+  0x3492,0x0a8e ,0x341e,0x0af1 ,0x33a9,0x0b54 ,
+  0x3334,0x0bb7 ,0x32bf,0x0c1a ,0x3249,0x0c7c ,
+  0x31d2,0x0cdf ,0x315b,0x0d41 ,0x30e4,0x0da4 ,
+  0x306c,0x0e06 ,0x2ff4,0x0e68 ,0x2f7b,0x0eca ,
+  0x2f02,0x0f2b ,0x2e88,0x0f8d ,0x2e0e,0x0fee ,
+  0x2d93,0x1050 ,0x2d18,0x10b1 ,0x2c9d,0x1112 ,
+  0x2c21,0x1173 ,0x2ba4,0x11d3 ,0x2b28,0x1234 ,
+  0x2aaa,0x1294 ,0x2a2d,0x12f4 ,0x29af,0x1354 ,
+  0x2931,0x13b4 ,0x28b2,0x1413 ,0x2833,0x1473 ,
+  0x27b3,0x14d2 ,0x2733,0x1531 ,0x26b3,0x1590 ,
+  0x2632,0x15ee ,0x25b1,0x164c ,0x252f,0x16ab ,
+  0x24ae,0x1709 ,0x242b,0x1766 ,0x23a9,0x17c4 ,
+  0x2326,0x1821 ,0x22a3,0x187e ,0x221f,0x18db ,
+  0x219c,0x1937 ,0x2117,0x1993 ,0x2093,0x19ef ,
+  0x200e,0x1a4b ,0x1f89,0x1aa7 ,0x1f04,0x1b02 ,
+  0x1e7e,0x1b5d ,0x1df8,0x1bb8 ,0x1d72,0x1c12 ,
+  0x1ceb,0x1c6c ,0x1c64,0x1cc6 ,0x1bdd,0x1d20 ,
+  0x1b56,0x1d79 ,0x1ace,0x1dd3 ,0x1a46,0x1e2b ,
+  0x19be,0x1e84 ,0x1935,0x1edc ,0x18ad,0x1f34 ,
+  0x1824,0x1f8c ,0x179b,0x1fe3 ,0x1711,0x203a ,
+  0x1688,0x2091 ,0x15fe,0x20e7 ,0x1574,0x213d ,
+  0x14ea,0x2193 ,0x145f,0x21e8 ,0x13d5,0x223d ,
+  0x134a,0x2292 ,0x12bf,0x22e7 ,0x1234,0x233b ,
+  0x11a8,0x238e ,0x111d,0x23e2 ,0x1091,0x2435 ,
+  0x1005,0x2488 ,0x0f79,0x24da ,0x0eed,0x252c ,
+  0x0e61,0x257e ,0x0dd4,0x25cf ,0x0d48,0x2620 ,
+  0x0cbb,0x2671 ,0x0c2e,0x26c1 ,0x0ba1,0x2711 ,
+  0x0b14,0x2760 ,0x0a87,0x27af ,0x09fa,0x27fe ,
+  0x096d,0x284c ,0x08df,0x289a ,0x0852,0x28e7 ,
+  0x07c4,0x2935 ,0x0736,0x2981 ,0x06a9,0x29ce ,
+  0x061b,0x2a1a ,0x058d,0x2a65 ,0x04ff,0x2ab0 ,
+  0x0471,0x2afb ,0x03e3,0x2b45 ,0x0355,0x2b8f ,
+  0x02c7,0x2bd8 ,0x0239,0x2c21 ,0x01aa,0x2c6a ,
+  0x011c,0x2cb2 ,0x008e,0x2cfa ,0x0000,0x2d41 ,
+  0xff72,0x2d88 ,0xfee4,0x2dcf ,0xfe56,0x2e15 ,
+  0xfdc7,0x2e5a ,0xfd39,0x2e9f ,0xfcab,0x2ee4 ,
+  0xfc1d,0x2f28 ,0xfb8f,0x2f6c ,0xfb01,0x2faf ,
+  0xfa73,0x2ff2 ,0xf9e5,0x3034 ,0xf957,0x3076 ,
+  0xf8ca,0x30b8 ,0xf83c,0x30f9 ,0xf7ae,0x3139 ,
+  0xf721,0x3179 ,0xf693,0x31b9 ,0xf606,0x31f8 ,
+  0xf579,0x3236 ,0xf4ec,0x3274 ,0xf45f,0x32b2 ,
+  0xf3d2,0x32ef ,0xf345,0x332c ,0xf2b8,0x3368 ,
+  0xf22c,0x33a3 ,0xf19f,0x33df ,0xf113,0x3419 ,
+  0xf087,0x3453 ,0xeffb,0x348d ,0xef6f,0x34c6 ,
+  0xeee3,0x34ff ,0xee58,0x3537 ,0xedcc,0x356e ,
+  0xed41,0x35a5 ,0xecb6,0x35dc ,0xec2b,0x3612 ,
+  0xeba1,0x3648 ,0xeb16,0x367d ,0xea8c,0x36b1 ,
+  0xea02,0x36e5 ,0xe978,0x3718 ,0xe8ef,0x374b ,
+  0xe865,0x377e ,0xe7dc,0x37b0 ,0xe753,0x37e1 ,
+  0xe6cb,0x3812 ,0xe642,0x3842 ,0xe5ba,0x3871 ,
+  0xe532,0x38a1 ,0xe4aa,0x38cf ,0xe423,0x38fd ,
+  0xe39c,0x392b ,0xe315,0x3958 ,0xe28e,0x3984 ,
+  0xe208,0x39b0 ,0xe182,0x39db ,0xe0fc,0x3a06 ,
+  0xe077,0x3a30 ,0xdff2,0x3a59 ,0xdf6d,0x3a82 ,
+  0xdee9,0x3aab ,0xde64,0x3ad3 ,0xdde1,0x3afa ,
+  0xdd5d,0x3b21 ,0xdcda,0x3b47 ,0xdc57,0x3b6d ,
+  0xdbd5,0x3b92 ,0xdb52,0x3bb6 ,0xdad1,0x3bda ,
+  0xda4f,0x3bfd ,0xd9ce,0x3c20 ,0xd94d,0x3c42 ,
+  0xd8cd,0x3c64 ,0xd84d,0x3c85 ,0xd7cd,0x3ca5 ,
+  0xd74e,0x3cc5 ,0xd6cf,0x3ce4 ,0xd651,0x3d03 ,
+  0xd5d3,0x3d21 ,0xd556,0x3d3f ,0xd4d8,0x3d5b ,
+  0xd45c,0x3d78 ,0xd3df,0x3d93 ,0xd363,0x3daf ,
+  0xd2e8,0x3dc9 ,0xd26d,0x3de3 ,0xd1f2,0x3dfc ,
+  0xd178,0x3e15 ,0xd0fe,0x3e2d ,0xd085,0x3e45 ,
+  0xd00c,0x3e5c ,0xcf94,0x3e72 ,0xcf1c,0x3e88 ,
+  0xcea5,0x3e9d ,0xce2e,0x3eb1 ,0xcdb7,0x3ec5 ,
+  0xcd41,0x3ed8 ,0xcccc,0x3eeb ,0xcc57,0x3efd ,
+  0xcbe2,0x3f0f ,0xcb6e,0x3f20 ,0xcafb,0x3f30 ,
+  0xca88,0x3f40 ,0xca15,0x3f4f ,0xc9a3,0x3f5d ,
+  0xc932,0x3f6b ,0xc8c1,0x3f78 ,0xc851,0x3f85 ,
+  0xc7e1,0x3f91 ,0xc772,0x3f9c ,0xc703,0x3fa7 ,
+  0xc695,0x3fb1 ,0xc627,0x3fbb ,0xc5ba,0x3fc4 ,
+  0xc54e,0x3fcc ,0xc4e2,0x3fd4 ,0xc476,0x3fdb ,
+  0xc40c,0x3fe1 ,0xc3a1,0x3fe7 ,0xc338,0x3fec ,
+  0xc2cf,0x3ff1 ,0xc266,0x3ff5 ,0xc1fe,0x3ff8 ,
+  0xc197,0x3ffb ,0xc130,0x3ffd ,0xc0ca,0x3fff ,
+  0xc065,0x4000 ,0xc000,0x4000 ,0xbf9c,0x4000 ,
+  0xbf38,0x3fff ,0xbed5,0x3ffd ,0xbe73,0x3ffb ,
+  0xbe11,0x3ff8 ,0xbdb0,0x3ff5 ,0xbd50,0x3ff1 ,
+  0xbcf0,0x3fec ,0xbc91,0x3fe7 ,0xbc32,0x3fe1 ,
+  0xbbd4,0x3fdb ,0xbb77,0x3fd4 ,0xbb1b,0x3fcc ,
+  0xbabf,0x3fc4 ,0xba64,0x3fbb ,0xba09,0x3fb1 ,
+  0xb9af,0x3fa7 ,0xb956,0x3f9c ,0xb8fd,0x3f91 ,
+  0xb8a6,0x3f85 ,0xb84f,0x3f78 ,0xb7f8,0x3f6b ,
+  0xb7a2,0x3f5d ,0xb74d,0x3f4f ,0xb6f9,0x3f40 ,
+  0xb6a5,0x3f30 ,0xb652,0x3f20 ,0xb600,0x3f0f ,
+  0xb5af,0x3efd ,0xb55e,0x3eeb ,0xb50e,0x3ed8 ,
+  0xb4be,0x3ec5 ,0xb470,0x3eb1 ,0xb422,0x3e9d ,
+  0xb3d5,0x3e88 ,0xb388,0x3e72 ,0xb33d,0x3e5c ,
+  0xb2f2,0x3e45 ,0xb2a7,0x3e2d ,0xb25e,0x3e15 ,
+  0xb215,0x3dfc ,0xb1cd,0x3de3 ,0xb186,0x3dc9 ,
+  0xb140,0x3daf ,0xb0fa,0x3d93 ,0xb0b5,0x3d78 ,
+  0xb071,0x3d5b ,0xb02d,0x3d3f ,0xafeb,0x3d21 ,
+  0xafa9,0x3d03 ,0xaf68,0x3ce4 ,0xaf28,0x3cc5 ,
+  0xaee8,0x3ca5 ,0xaea9,0x3c85 ,0xae6b,0x3c64 ,
+  0xae2e,0x3c42 ,0xadf2,0x3c20 ,0xadb6,0x3bfd ,
+  0xad7b,0x3bda ,0xad41,0x3bb6 ,0xad08,0x3b92 ,
+  0xacd0,0x3b6d ,0xac98,0x3b47 ,0xac61,0x3b21 ,
+  0xac2b,0x3afa ,0xabf6,0x3ad3 ,0xabc2,0x3aab ,
+  0xab8e,0x3a82 ,0xab5b,0x3a59 ,0xab29,0x3a30 ,
+  0xaaf8,0x3a06 ,0xaac8,0x39db ,0xaa98,0x39b0 ,
+  0xaa6a,0x3984 ,0xaa3c,0x3958 ,0xaa0f,0x392b ,
+  0xa9e3,0x38fd ,0xa9b7,0x38cf ,0xa98d,0x38a1 ,
+  0xa963,0x3871 ,0xa93a,0x3842 ,0xa912,0x3812 ,
+  0xa8eb,0x37e1 ,0xa8c5,0x37b0 ,0xa89f,0x377e ,
+  0xa87b,0x374b ,0xa857,0x3718 ,0xa834,0x36e5 ,
+  0xa812,0x36b1 ,0xa7f1,0x367d ,0xa7d0,0x3648 ,
+  0xa7b1,0x3612 ,0xa792,0x35dc ,0xa774,0x35a5 ,
+  0xa757,0x356e ,0xa73b,0x3537 ,0xa71f,0x34ff ,
+  0xa705,0x34c6 ,0xa6eb,0x348d ,0xa6d3,0x3453 ,
+  0xa6bb,0x3419 ,0xa6a4,0x33df ,0xa68e,0x33a3 ,
+  0xa678,0x3368 ,0xa664,0x332c ,0xa650,0x32ef ,
+  0xa63e,0x32b2 ,0xa62c,0x3274 ,0xa61b,0x3236 ,
+  0xa60b,0x31f8 ,0xa5fb,0x31b9 ,0xa5ed,0x3179 ,
+  0xa5e0,0x3139 ,0xa5d3,0x30f9 ,0xa5c7,0x30b8 ,
+  0xa5bc,0x3076 ,0xa5b2,0x3034 ,0xa5a9,0x2ff2 ,
+  0xa5a1,0x2faf ,0xa599,0x2f6c ,0xa593,0x2f28 ,
+  0xa58d,0x2ee4 ,0xa588,0x2e9f ,0xa585,0x2e5a ,
+  0xa581,0x2e15 ,0xa57f,0x2dcf ,0xa57e,0x2d88 ,
+  0xa57e,0x2d41 ,0xa57e,0x2cfa ,0xa57f,0x2cb2 ,
+  0xa581,0x2c6a ,0xa585,0x2c21 ,0xa588,0x2bd8 ,
+  0xa58d,0x2b8f ,0xa593,0x2b45 ,0xa599,0x2afb ,
+  0xa5a1,0x2ab0 ,0xa5a9,0x2a65 ,0xa5b2,0x2a1a ,
+  0xa5bc,0x29ce ,0xa5c7,0x2981 ,0xa5d3,0x2935 ,
+  0xa5e0,0x28e7 ,0xa5ed,0x289a ,0xa5fb,0x284c ,
+  0xa60b,0x27fe ,0xa61b,0x27af ,0xa62c,0x2760 ,
+  0xa63e,0x2711 ,0xa650,0x26c1 ,0xa664,0x2671 ,
+  0xa678,0x2620 ,0xa68e,0x25cf ,0xa6a4,0x257e ,
+  0xa6bb,0x252c ,0xa6d3,0x24da ,0xa6eb,0x2488 ,
+  0xa705,0x2435 ,0xa71f,0x23e2 ,0xa73b,0x238e ,
+  0xa757,0x233b ,0xa774,0x22e7 ,0xa792,0x2292 ,
+  0xa7b1,0x223d ,0xa7d0,0x21e8 ,0xa7f1,0x2193 ,
+  0xa812,0x213d ,0xa834,0x20e7 ,0xa857,0x2091 ,
+  0xa87b,0x203a ,0xa89f,0x1fe3 ,0xa8c5,0x1f8c ,
+  0xa8eb,0x1f34 ,0xa912,0x1edc ,0xa93a,0x1e84 ,
+  0xa963,0x1e2b ,0xa98d,0x1dd3 ,0xa9b7,0x1d79 ,
+  0xa9e3,0x1d20 ,0xaa0f,0x1cc6 ,0xaa3c,0x1c6c ,
+  0xaa6a,0x1c12 ,0xaa98,0x1bb8 ,0xaac8,0x1b5d ,
+  0xaaf8,0x1b02 ,0xab29,0x1aa7 ,0xab5b,0x1a4b ,
+  0xab8e,0x19ef ,0xabc2,0x1993 ,0xabf6,0x1937 ,
+  0xac2b,0x18db ,0xac61,0x187e ,0xac98,0x1821 ,
+  0xacd0,0x17c4 ,0xad08,0x1766 ,0xad41,0x1709 ,
+  0xad7b,0x16ab ,0xadb6,0x164c ,0xadf2,0x15ee ,
+  0xae2e,0x1590 ,0xae6b,0x1531 ,0xaea9,0x14d2 ,
+  0xaee8,0x1473 ,0xaf28,0x1413 ,0xaf68,0x13b4 ,
+  0xafa9,0x1354 ,0xafeb,0x12f4 ,0xb02d,0x1294 ,
+  0xb071,0x1234 ,0xb0b5,0x11d3 ,0xb0fa,0x1173 ,
+  0xb140,0x1112 ,0xb186,0x10b1 ,0xb1cd,0x1050 ,
+  0xb215,0x0fee ,0xb25e,0x0f8d ,0xb2a7,0x0f2b ,
+  0xb2f2,0x0eca ,0xb33d,0x0e68 ,0xb388,0x0e06 ,
+  0xb3d5,0x0da4 ,0xb422,0x0d41 ,0xb470,0x0cdf ,
+  0xb4be,0x0c7c ,0xb50e,0x0c1a ,0xb55e,0x0bb7 ,
+  0xb5af,0x0b54 ,0xb600,0x0af1 ,0xb652,0x0a8e ,
+  0xb6a5,0x0a2b ,0xb6f9,0x09c7 ,0xb74d,0x0964 ,
+  0xb7a2,0x0901 ,0xb7f8,0x089d ,0xb84f,0x0839 ,
+  0xb8a6,0x07d6 ,0xb8fd,0x0772 ,0xb956,0x070e ,
+  0xb9af,0x06aa ,0xba09,0x0646 ,0xba64,0x05e2 ,
+  0xbabf,0x057e ,0xbb1b,0x051a ,0xbb77,0x04b5 ,
+  0xbbd4,0x0451 ,0xbc32,0x03ed ,0xbc91,0x0388 ,
+  0xbcf0,0x0324 ,0xbd50,0x02c0 ,0xbdb0,0x025b ,
+  0xbe11,0x01f7 ,0xbe73,0x0192 ,0xbed5,0x012e ,
+  0xbf38,0x00c9 ,0xbf9c,0x0065 };
+
+
+extern const int s_Q14R_8;
+const int s_Q14R_8 = 1024;
+extern const unsigned short t_Q14R_8[2032];
+const unsigned short t_Q14R_8[2032] = {
+  0x4000,0x0000 ,0x4000,0x0000 ,0x4000,0x0000 ,
+  0x3b21,0x187e ,0x3ec5,0x0c7c ,0x3537,0x238e ,
+  0x2d41,0x2d41 ,0x3b21,0x187e ,0x187e,0x3b21 ,
+  0x187e,0x3b21 ,0x3537,0x238e ,0xf384,0x3ec5 ,
+  0x0000,0x4000 ,0x2d41,0x2d41 ,0xd2bf,0x2d41 ,
+  0xe782,0x3b21 ,0x238e,0x3537 ,0xc13b,0x0c7c ,
+  0xd2bf,0x2d41 ,0x187e,0x3b21 ,0xc4df,0xe782 ,
+  0xc4df,0x187e ,0x0c7c,0x3ec5 ,0xdc72,0xcac9 ,
+  0x4000,0x0000 ,0x4000,0x0000 ,0x4000,0x0000 ,
+  0x3fb1,0x0646 ,0x3fec,0x0324 ,0x3f4f,0x0964 ,
+  0x3ec5,0x0c7c ,0x3fb1,0x0646 ,0x3d3f,0x1294 ,
+  0x3d3f,0x1294 ,0x3f4f,0x0964 ,0x39db,0x1b5d ,
+  0x3b21,0x187e ,0x3ec5,0x0c7c ,0x3537,0x238e ,
+  0x3871,0x1e2b ,0x3e15,0x0f8d ,0x2f6c,0x2afb ,
+  0x3537,0x238e ,0x3d3f,0x1294 ,0x289a,0x3179 ,
+  0x3179,0x289a ,0x3c42,0x1590 ,0x20e7,0x36e5 ,
+  0x2d41,0x2d41 ,0x3b21,0x187e ,0x187e,0x3b21 ,
+  0x289a,0x3179 ,0x39db,0x1b5d ,0x0f8d,0x3e15 ,
+  0x238e,0x3537 ,0x3871,0x1e2b ,0x0646,0x3fb1 ,
+  0x1e2b,0x3871 ,0x36e5,0x20e7 ,0xfcdc,0x3fec ,
+  0x187e,0x3b21 ,0x3537,0x238e ,0xf384,0x3ec5 ,
+  0x1294,0x3d3f ,0x3368,0x2620 ,0xea70,0x3c42 ,
+  0x0c7c,0x3ec5 ,0x3179,0x289a ,0xe1d5,0x3871 ,
+  0x0646,0x3fb1 ,0x2f6c,0x2afb ,0xd9e0,0x3368 ,
+  0x0000,0x4000 ,0x2d41,0x2d41 ,0xd2bf,0x2d41 ,
+  0xf9ba,0x3fb1 ,0x2afb,0x2f6c ,0xcc98,0x2620 ,
+  0xf384,0x3ec5 ,0x289a,0x3179 ,0xc78f,0x1e2b ,
+  0xed6c,0x3d3f ,0x2620,0x3368 ,0xc3be,0x1590 ,
+  0xe782,0x3b21 ,0x238e,0x3537 ,0xc13b,0x0c7c ,
+  0xe1d5,0x3871 ,0x20e7,0x36e5 ,0xc014,0x0324 ,
+  0xdc72,0x3537 ,0x1e2b,0x3871 ,0xc04f,0xf9ba ,
+  0xd766,0x3179 ,0x1b5d,0x39db ,0xc1eb,0xf073 ,
+  0xd2bf,0x2d41 ,0x187e,0x3b21 ,0xc4df,0xe782 ,
+  0xce87,0x289a ,0x1590,0x3c42 ,0xc91b,0xdf19 ,
+  0xcac9,0x238e ,0x1294,0x3d3f ,0xce87,0xd766 ,
+  0xc78f,0x1e2b ,0x0f8d,0x3e15 ,0xd505,0xd094 ,
+  0xc4df,0x187e ,0x0c7c,0x3ec5 ,0xdc72,0xcac9 ,
+  0xc2c1,0x1294 ,0x0964,0x3f4f ,0xe4a3,0xc625 ,
+  0xc13b,0x0c7c ,0x0646,0x3fb1 ,0xed6c,0xc2c1 ,
+  0xc04f,0x0646 ,0x0324,0x3fec ,0xf69c,0xc0b1 ,
+  0x4000,0x0000 ,0x4000,0x0000 ,0x4000,0x0000 ,
+  0x3ffb,0x0192 ,0x3fff,0x00c9 ,0x3ff5,0x025b ,
+  0x3fec,0x0324 ,0x3ffb,0x0192 ,0x3fd4,0x04b5 ,
+  0x3fd4,0x04b5 ,0x3ff5,0x025b ,0x3f9c,0x070e ,
+  0x3fb1,0x0646 ,0x3fec,0x0324 ,0x3f4f,0x0964 ,
+  0x3f85,0x07d6 ,0x3fe1,0x03ed ,0x3eeb,0x0bb7 ,
+  0x3f4f,0x0964 ,0x3fd4,0x04b5 ,0x3e72,0x0e06 ,
+  0x3f0f,0x0af1 ,0x3fc4,0x057e ,0x3de3,0x1050 ,
+  0x3ec5,0x0c7c ,0x3fb1,0x0646 ,0x3d3f,0x1294 ,
+  0x3e72,0x0e06 ,0x3f9c,0x070e ,0x3c85,0x14d2 ,
+  0x3e15,0x0f8d ,0x3f85,0x07d6 ,0x3bb6,0x1709 ,
+  0x3daf,0x1112 ,0x3f6b,0x089d ,0x3ad3,0x1937 ,
+  0x3d3f,0x1294 ,0x3f4f,0x0964 ,0x39db,0x1b5d ,
+  0x3cc5,0x1413 ,0x3f30,0x0a2b ,0x38cf,0x1d79 ,
+  0x3c42,0x1590 ,0x3f0f,0x0af1 ,0x37b0,0x1f8c ,
+  0x3bb6,0x1709 ,0x3eeb,0x0bb7 ,0x367d,0x2193 ,
+  0x3b21,0x187e ,0x3ec5,0x0c7c ,0x3537,0x238e ,
+  0x3a82,0x19ef ,0x3e9d,0x0d41 ,0x33df,0x257e ,
+  0x39db,0x1b5d ,0x3e72,0x0e06 ,0x3274,0x2760 ,
+  0x392b,0x1cc6 ,0x3e45,0x0eca ,0x30f9,0x2935 ,
+  0x3871,0x1e2b ,0x3e15,0x0f8d ,0x2f6c,0x2afb ,
+  0x37b0,0x1f8c ,0x3de3,0x1050 ,0x2dcf,0x2cb2 ,
+  0x36e5,0x20e7 ,0x3daf,0x1112 ,0x2c21,0x2e5a ,
+  0x3612,0x223d ,0x3d78,0x11d3 ,0x2a65,0x2ff2 ,
+  0x3537,0x238e ,0x3d3f,0x1294 ,0x289a,0x3179 ,
+  0x3453,0x24da ,0x3d03,0x1354 ,0x26c1,0x32ef ,
+  0x3368,0x2620 ,0x3cc5,0x1413 ,0x24da,0x3453 ,
+  0x3274,0x2760 ,0x3c85,0x14d2 ,0x22e7,0x35a5 ,
+  0x3179,0x289a ,0x3c42,0x1590 ,0x20e7,0x36e5 ,
+  0x3076,0x29ce ,0x3bfd,0x164c ,0x1edc,0x3812 ,
+  0x2f6c,0x2afb ,0x3bb6,0x1709 ,0x1cc6,0x392b ,
+  0x2e5a,0x2c21 ,0x3b6d,0x17c4 ,0x1aa7,0x3a30 ,
+  0x2d41,0x2d41 ,0x3b21,0x187e ,0x187e,0x3b21 ,
+  0x2c21,0x2e5a ,0x3ad3,0x1937 ,0x164c,0x3bfd ,
+  0x2afb,0x2f6c ,0x3a82,0x19ef ,0x1413,0x3cc5 ,
+  0x29ce,0x3076 ,0x3a30,0x1aa7 ,0x11d3,0x3d78 ,
+  0x289a,0x3179 ,0x39db,0x1b5d ,0x0f8d,0x3e15 ,
+  0x2760,0x3274 ,0x3984,0x1c12 ,0x0d41,0x3e9d ,
+  0x2620,0x3368 ,0x392b,0x1cc6 ,0x0af1,0x3f0f ,
+  0x24da,0x3453 ,0x38cf,0x1d79 ,0x089d,0x3f6b ,
+  0x238e,0x3537 ,0x3871,0x1e2b ,0x0646,0x3fb1 ,
+  0x223d,0x3612 ,0x3812,0x1edc ,0x03ed,0x3fe1 ,
+  0x20e7,0x36e5 ,0x37b0,0x1f8c ,0x0192,0x3ffb ,
+  0x1f8c,0x37b0 ,0x374b,0x203a ,0xff37,0x3fff ,
+  0x1e2b,0x3871 ,0x36e5,0x20e7 ,0xfcdc,0x3fec ,
+  0x1cc6,0x392b ,0x367d,0x2193 ,0xfa82,0x3fc4 ,
+  0x1b5d,0x39db ,0x3612,0x223d ,0xf82a,0x3f85 ,
+  0x19ef,0x3a82 ,0x35a5,0x22e7 ,0xf5d5,0x3f30 ,
+  0x187e,0x3b21 ,0x3537,0x238e ,0xf384,0x3ec5 ,
+  0x1709,0x3bb6 ,0x34c6,0x2435 ,0xf136,0x3e45 ,
+  0x1590,0x3c42 ,0x3453,0x24da ,0xeeee,0x3daf ,
+  0x1413,0x3cc5 ,0x33df,0x257e ,0xecac,0x3d03 ,
+  0x1294,0x3d3f ,0x3368,0x2620 ,0xea70,0x3c42 ,
+  0x1112,0x3daf ,0x32ef,0x26c1 ,0xe83c,0x3b6d ,
+  0x0f8d,0x3e15 ,0x3274,0x2760 ,0xe611,0x3a82 ,
+  0x0e06,0x3e72 ,0x31f8,0x27fe ,0xe3ee,0x3984 ,
+  0x0c7c,0x3ec5 ,0x3179,0x289a ,0xe1d5,0x3871 ,
+  0x0af1,0x3f0f ,0x30f9,0x2935 ,0xdfc6,0x374b ,
+  0x0964,0x3f4f ,0x3076,0x29ce ,0xddc3,0x3612 ,
+  0x07d6,0x3f85 ,0x2ff2,0x2a65 ,0xdbcb,0x34c6 ,
+  0x0646,0x3fb1 ,0x2f6c,0x2afb ,0xd9e0,0x3368 ,
+  0x04b5,0x3fd4 ,0x2ee4,0x2b8f ,0xd802,0x31f8 ,
+  0x0324,0x3fec ,0x2e5a,0x2c21 ,0xd632,0x3076 ,
+  0x0192,0x3ffb ,0x2dcf,0x2cb2 ,0xd471,0x2ee4 ,
+  0x0000,0x4000 ,0x2d41,0x2d41 ,0xd2bf,0x2d41 ,
+  0xfe6e,0x3ffb ,0x2cb2,0x2dcf ,0xd11c,0x2b8f ,
+  0xfcdc,0x3fec ,0x2c21,0x2e5a ,0xcf8a,0x29ce ,
+  0xfb4b,0x3fd4 ,0x2b8f,0x2ee4 ,0xce08,0x27fe ,
+  0xf9ba,0x3fb1 ,0x2afb,0x2f6c ,0xcc98,0x2620 ,
+  0xf82a,0x3f85 ,0x2a65,0x2ff2 ,0xcb3a,0x2435 ,
+  0xf69c,0x3f4f ,0x29ce,0x3076 ,0xc9ee,0x223d ,
+  0xf50f,0x3f0f ,0x2935,0x30f9 ,0xc8b5,0x203a ,
+  0xf384,0x3ec5 ,0x289a,0x3179 ,0xc78f,0x1e2b ,
+  0xf1fa,0x3e72 ,0x27fe,0x31f8 ,0xc67c,0x1c12 ,
+  0xf073,0x3e15 ,0x2760,0x3274 ,0xc57e,0x19ef ,
+  0xeeee,0x3daf ,0x26c1,0x32ef ,0xc493,0x17c4 ,
+  0xed6c,0x3d3f ,0x2620,0x3368 ,0xc3be,0x1590 ,
+  0xebed,0x3cc5 ,0x257e,0x33df ,0xc2fd,0x1354 ,
+  0xea70,0x3c42 ,0x24da,0x3453 ,0xc251,0x1112 ,
+  0xe8f7,0x3bb6 ,0x2435,0x34c6 ,0xc1bb,0x0eca ,
+  0xe782,0x3b21 ,0x238e,0x3537 ,0xc13b,0x0c7c ,
+  0xe611,0x3a82 ,0x22e7,0x35a5 ,0xc0d0,0x0a2b ,
+  0xe4a3,0x39db ,0x223d,0x3612 ,0xc07b,0x07d6 ,
+  0xe33a,0x392b ,0x2193,0x367d ,0xc03c,0x057e ,
+  0xe1d5,0x3871 ,0x20e7,0x36e5 ,0xc014,0x0324 ,
+  0xe074,0x37b0 ,0x203a,0x374b ,0xc001,0x00c9 ,
+  0xdf19,0x36e5 ,0x1f8c,0x37b0 ,0xc005,0xfe6e ,
+  0xddc3,0x3612 ,0x1edc,0x3812 ,0xc01f,0xfc13 ,
+  0xdc72,0x3537 ,0x1e2b,0x3871 ,0xc04f,0xf9ba ,
+  0xdb26,0x3453 ,0x1d79,0x38cf ,0xc095,0xf763 ,
+  0xd9e0,0x3368 ,0x1cc6,0x392b ,0xc0f1,0xf50f ,
+  0xd8a0,0x3274 ,0x1c12,0x3984 ,0xc163,0xf2bf ,
+  0xd766,0x3179 ,0x1b5d,0x39db ,0xc1eb,0xf073 ,
+  0xd632,0x3076 ,0x1aa7,0x3a30 ,0xc288,0xee2d ,
+  0xd505,0x2f6c ,0x19ef,0x3a82 ,0xc33b,0xebed ,
+  0xd3df,0x2e5a ,0x1937,0x3ad3 ,0xc403,0xe9b4 ,
+  0xd2bf,0x2d41 ,0x187e,0x3b21 ,0xc4df,0xe782 ,
+  0xd1a6,0x2c21 ,0x17c4,0x3b6d ,0xc5d0,0xe559 ,
+  0xd094,0x2afb ,0x1709,0x3bb6 ,0xc6d5,0xe33a ,
+  0xcf8a,0x29ce ,0x164c,0x3bfd ,0xc7ee,0xe124 ,
+  0xce87,0x289a ,0x1590,0x3c42 ,0xc91b,0xdf19 ,
+  0xcd8c,0x2760 ,0x14d2,0x3c85 ,0xca5b,0xdd19 ,
+  0xcc98,0x2620 ,0x1413,0x3cc5 ,0xcbad,0xdb26 ,
+  0xcbad,0x24da ,0x1354,0x3d03 ,0xcd11,0xd93f ,
+  0xcac9,0x238e ,0x1294,0x3d3f ,0xce87,0xd766 ,
+  0xc9ee,0x223d ,0x11d3,0x3d78 ,0xd00e,0xd59b ,
+  0xc91b,0x20e7 ,0x1112,0x3daf ,0xd1a6,0xd3df ,
+  0xc850,0x1f8c ,0x1050,0x3de3 ,0xd34e,0xd231 ,
+  0xc78f,0x1e2b ,0x0f8d,0x3e15 ,0xd505,0xd094 ,
+  0xc6d5,0x1cc6 ,0x0eca,0x3e45 ,0xd6cb,0xcf07 ,
+  0xc625,0x1b5d ,0x0e06,0x3e72 ,0xd8a0,0xcd8c ,
+  0xc57e,0x19ef ,0x0d41,0x3e9d ,0xda82,0xcc21 ,
+  0xc4df,0x187e ,0x0c7c,0x3ec5 ,0xdc72,0xcac9 ,
+  0xc44a,0x1709 ,0x0bb7,0x3eeb ,0xde6d,0xc983 ,
+  0xc3be,0x1590 ,0x0af1,0x3f0f ,0xe074,0xc850 ,
+  0xc33b,0x1413 ,0x0a2b,0x3f30 ,0xe287,0xc731 ,
+  0xc2c1,0x1294 ,0x0964,0x3f4f ,0xe4a3,0xc625 ,
+  0xc251,0x1112 ,0x089d,0x3f6b ,0xe6c9,0xc52d ,
+  0xc1eb,0x0f8d ,0x07d6,0x3f85 ,0xe8f7,0xc44a ,
+  0xc18e,0x0e06 ,0x070e,0x3f9c ,0xeb2e,0xc37b ,
+  0xc13b,0x0c7c ,0x0646,0x3fb1 ,0xed6c,0xc2c1 ,
+  0xc0f1,0x0af1 ,0x057e,0x3fc4 ,0xefb0,0xc21d ,
+  0xc0b1,0x0964 ,0x04b5,0x3fd4 ,0xf1fa,0xc18e ,
+  0xc07b,0x07d6 ,0x03ed,0x3fe1 ,0xf449,0xc115 ,
+  0xc04f,0x0646 ,0x0324,0x3fec ,0xf69c,0xc0b1 ,
+  0xc02c,0x04b5 ,0x025b,0x3ff5 ,0xf8f2,0xc064 ,
+  0xc014,0x0324 ,0x0192,0x3ffb ,0xfb4b,0xc02c ,
+  0xc005,0x0192 ,0x00c9,0x3fff ,0xfda5,0xc00b ,
+  0x4000,0x0000 ,0x4000,0x0065 ,0x3fff,0x00c9 ,
+  0x3ffd,0x012e ,0x3ffb,0x0192 ,0x3ff8,0x01f7 ,
+  0x3ff5,0x025b ,0x3ff1,0x02c0 ,0x3fec,0x0324 ,
+  0x3fe7,0x0388 ,0x3fe1,0x03ed ,0x3fdb,0x0451 ,
+  0x3fd4,0x04b5 ,0x3fcc,0x051a ,0x3fc4,0x057e ,
+  0x3fbb,0x05e2 ,0x3fb1,0x0646 ,0x3fa7,0x06aa ,
+  0x3f9c,0x070e ,0x3f91,0x0772 ,0x3f85,0x07d6 ,
+  0x3f78,0x0839 ,0x3f6b,0x089d ,0x3f5d,0x0901 ,
+  0x3f4f,0x0964 ,0x3f40,0x09c7 ,0x3f30,0x0a2b ,
+  0x3f20,0x0a8e ,0x3f0f,0x0af1 ,0x3efd,0x0b54 ,
+  0x3eeb,0x0bb7 ,0x3ed8,0x0c1a ,0x3ec5,0x0c7c ,
+  0x3eb1,0x0cdf ,0x3e9d,0x0d41 ,0x3e88,0x0da4 ,
+  0x3e72,0x0e06 ,0x3e5c,0x0e68 ,0x3e45,0x0eca ,
+  0x3e2d,0x0f2b ,0x3e15,0x0f8d ,0x3dfc,0x0fee ,
+  0x3de3,0x1050 ,0x3dc9,0x10b1 ,0x3daf,0x1112 ,
+  0x3d93,0x1173 ,0x3d78,0x11d3 ,0x3d5b,0x1234 ,
+  0x3d3f,0x1294 ,0x3d21,0x12f4 ,0x3d03,0x1354 ,
+  0x3ce4,0x13b4 ,0x3cc5,0x1413 ,0x3ca5,0x1473 ,
+  0x3c85,0x14d2 ,0x3c64,0x1531 ,0x3c42,0x1590 ,
+  0x3c20,0x15ee ,0x3bfd,0x164c ,0x3bda,0x16ab ,
+  0x3bb6,0x1709 ,0x3b92,0x1766 ,0x3b6d,0x17c4 ,
+  0x3b47,0x1821 ,0x3b21,0x187e ,0x3afa,0x18db ,
+  0x3ad3,0x1937 ,0x3aab,0x1993 ,0x3a82,0x19ef ,
+  0x3a59,0x1a4b ,0x3a30,0x1aa7 ,0x3a06,0x1b02 ,
+  0x39db,0x1b5d ,0x39b0,0x1bb8 ,0x3984,0x1c12 ,
+  0x3958,0x1c6c ,0x392b,0x1cc6 ,0x38fd,0x1d20 ,
+  0x38cf,0x1d79 ,0x38a1,0x1dd3 ,0x3871,0x1e2b ,
+  0x3842,0x1e84 ,0x3812,0x1edc ,0x37e1,0x1f34 ,
+  0x37b0,0x1f8c ,0x377e,0x1fe3 ,0x374b,0x203a ,
+  0x3718,0x2091 ,0x36e5,0x20e7 ,0x36b1,0x213d ,
+  0x367d,0x2193 ,0x3648,0x21e8 ,0x3612,0x223d ,
+  0x35dc,0x2292 ,0x35a5,0x22e7 ,0x356e,0x233b ,
+  0x3537,0x238e ,0x34ff,0x23e2 ,0x34c6,0x2435 ,
+  0x348d,0x2488 ,0x3453,0x24da ,0x3419,0x252c ,
+  0x33df,0x257e ,0x33a3,0x25cf ,0x3368,0x2620 ,
+  0x332c,0x2671 ,0x32ef,0x26c1 ,0x32b2,0x2711 ,
+  0x3274,0x2760 ,0x3236,0x27af ,0x31f8,0x27fe ,
+  0x31b9,0x284c ,0x3179,0x289a ,0x3139,0x28e7 ,
+  0x30f9,0x2935 ,0x30b8,0x2981 ,0x3076,0x29ce ,
+  0x3034,0x2a1a ,0x2ff2,0x2a65 ,0x2faf,0x2ab0 ,
+  0x2f6c,0x2afb ,0x2f28,0x2b45 ,0x2ee4,0x2b8f ,
+  0x2e9f,0x2bd8 ,0x2e5a,0x2c21 ,0x2e15,0x2c6a ,
+  0x2dcf,0x2cb2 ,0x2d88,0x2cfa ,0x2d41,0x2d41 ,
+  0x2cfa,0x2d88 ,0x2cb2,0x2dcf ,0x2c6a,0x2e15 ,
+  0x2c21,0x2e5a ,0x2bd8,0x2e9f ,0x2b8f,0x2ee4 ,
+  0x2b45,0x2f28 ,0x2afb,0x2f6c ,0x2ab0,0x2faf ,
+  0x2a65,0x2ff2 ,0x2a1a,0x3034 ,0x29ce,0x3076 ,
+  0x2981,0x30b8 ,0x2935,0x30f9 ,0x28e7,0x3139 ,
+  0x289a,0x3179 ,0x284c,0x31b9 ,0x27fe,0x31f8 ,
+  0x27af,0x3236 ,0x2760,0x3274 ,0x2711,0x32b2 ,
+  0x26c1,0x32ef ,0x2671,0x332c ,0x2620,0x3368 ,
+  0x25cf,0x33a3 ,0x257e,0x33df ,0x252c,0x3419 ,
+  0x24da,0x3453 ,0x2488,0x348d ,0x2435,0x34c6 ,
+  0x23e2,0x34ff ,0x238e,0x3537 ,0x233b,0x356e ,
+  0x22e7,0x35a5 ,0x2292,0x35dc ,0x223d,0x3612 ,
+  0x21e8,0x3648 ,0x2193,0x367d ,0x213d,0x36b1 ,
+  0x20e7,0x36e5 ,0x2091,0x3718 ,0x203a,0x374b ,
+  0x1fe3,0x377e ,0x1f8c,0x37b0 ,0x1f34,0x37e1 ,
+  0x1edc,0x3812 ,0x1e84,0x3842 ,0x1e2b,0x3871 ,
+  0x1dd3,0x38a1 ,0x1d79,0x38cf ,0x1d20,0x38fd ,
+  0x1cc6,0x392b ,0x1c6c,0x3958 ,0x1c12,0x3984 ,
+  0x1bb8,0x39b0 ,0x1b5d,0x39db ,0x1b02,0x3a06 ,
+  0x1aa7,0x3a30 ,0x1a4b,0x3a59 ,0x19ef,0x3a82 ,
+  0x1993,0x3aab ,0x1937,0x3ad3 ,0x18db,0x3afa ,
+  0x187e,0x3b21 ,0x1821,0x3b47 ,0x17c4,0x3b6d ,
+  0x1766,0x3b92 ,0x1709,0x3bb6 ,0x16ab,0x3bda ,
+  0x164c,0x3bfd ,0x15ee,0x3c20 ,0x1590,0x3c42 ,
+  0x1531,0x3c64 ,0x14d2,0x3c85 ,0x1473,0x3ca5 ,
+  0x1413,0x3cc5 ,0x13b4,0x3ce4 ,0x1354,0x3d03 ,
+  0x12f4,0x3d21 ,0x1294,0x3d3f ,0x1234,0x3d5b ,
+  0x11d3,0x3d78 ,0x1173,0x3d93 ,0x1112,0x3daf ,
+  0x10b1,0x3dc9 ,0x1050,0x3de3 ,0x0fee,0x3dfc ,
+  0x0f8d,0x3e15 ,0x0f2b,0x3e2d ,0x0eca,0x3e45 ,
+  0x0e68,0x3e5c ,0x0e06,0x3e72 ,0x0da4,0x3e88 ,
+  0x0d41,0x3e9d ,0x0cdf,0x3eb1 ,0x0c7c,0x3ec5 ,
+  0x0c1a,0x3ed8 ,0x0bb7,0x3eeb ,0x0b54,0x3efd ,
+  0x0af1,0x3f0f ,0x0a8e,0x3f20 ,0x0a2b,0x3f30 ,
+  0x09c7,0x3f40 ,0x0964,0x3f4f ,0x0901,0x3f5d ,
+  0x089d,0x3f6b ,0x0839,0x3f78 ,0x07d6,0x3f85 ,
+  0x0772,0x3f91 ,0x070e,0x3f9c ,0x06aa,0x3fa7 ,
+  0x0646,0x3fb1 ,0x05e2,0x3fbb ,0x057e,0x3fc4 ,
+  0x051a,0x3fcc ,0x04b5,0x3fd4 ,0x0451,0x3fdb ,
+  0x03ed,0x3fe1 ,0x0388,0x3fe7 ,0x0324,0x3fec ,
+  0x02c0,0x3ff1 ,0x025b,0x3ff5 ,0x01f7,0x3ff8 ,
+  0x0192,0x3ffb ,0x012e,0x3ffd ,0x00c9,0x3fff ,
+  0x0065,0x4000 ,0x0000,0x4000 ,0xff9b,0x4000 ,
+  0xff37,0x3fff ,0xfed2,0x3ffd ,0xfe6e,0x3ffb ,
+  0xfe09,0x3ff8 ,0xfda5,0x3ff5 ,0xfd40,0x3ff1 ,
+  0xfcdc,0x3fec ,0xfc78,0x3fe7 ,0xfc13,0x3fe1 ,
+  0xfbaf,0x3fdb ,0xfb4b,0x3fd4 ,0xfae6,0x3fcc ,
+  0xfa82,0x3fc4 ,0xfa1e,0x3fbb ,0xf9ba,0x3fb1 ,
+  0xf956,0x3fa7 ,0xf8f2,0x3f9c ,0xf88e,0x3f91 ,
+  0xf82a,0x3f85 ,0xf7c7,0x3f78 ,0xf763,0x3f6b ,
+  0xf6ff,0x3f5d ,0xf69c,0x3f4f ,0xf639,0x3f40 ,
+  0xf5d5,0x3f30 ,0xf572,0x3f20 ,0xf50f,0x3f0f ,
+  0xf4ac,0x3efd ,0xf449,0x3eeb ,0xf3e6,0x3ed8 ,
+  0xf384,0x3ec5 ,0xf321,0x3eb1 ,0xf2bf,0x3e9d ,
+  0xf25c,0x3e88 ,0xf1fa,0x3e72 ,0xf198,0x3e5c ,
+  0xf136,0x3e45 ,0xf0d5,0x3e2d ,0xf073,0x3e15 ,
+  0xf012,0x3dfc ,0xefb0,0x3de3 ,0xef4f,0x3dc9 ,
+  0xeeee,0x3daf ,0xee8d,0x3d93 ,0xee2d,0x3d78 ,
+  0xedcc,0x3d5b ,0xed6c,0x3d3f ,0xed0c,0x3d21 ,
+  0xecac,0x3d03 ,0xec4c,0x3ce4 ,0xebed,0x3cc5 ,
+  0xeb8d,0x3ca5 ,0xeb2e,0x3c85 ,0xeacf,0x3c64 ,
+  0xea70,0x3c42 ,0xea12,0x3c20 ,0xe9b4,0x3bfd ,
+  0xe955,0x3bda ,0xe8f7,0x3bb6 ,0xe89a,0x3b92 ,
+  0xe83c,0x3b6d ,0xe7df,0x3b47 ,0xe782,0x3b21 ,
+  0xe725,0x3afa ,0xe6c9,0x3ad3 ,0xe66d,0x3aab ,
+  0xe611,0x3a82 ,0xe5b5,0x3a59 ,0xe559,0x3a30 ,
+  0xe4fe,0x3a06 ,0xe4a3,0x39db ,0xe448,0x39b0 ,
+  0xe3ee,0x3984 ,0xe394,0x3958 ,0xe33a,0x392b ,
+  0xe2e0,0x38fd ,0xe287,0x38cf ,0xe22d,0x38a1 ,
+  0xe1d5,0x3871 ,0xe17c,0x3842 ,0xe124,0x3812 ,
+  0xe0cc,0x37e1 ,0xe074,0x37b0 ,0xe01d,0x377e ,
+  0xdfc6,0x374b ,0xdf6f,0x3718 ,0xdf19,0x36e5 ,
+  0xdec3,0x36b1 ,0xde6d,0x367d ,0xde18,0x3648 ,
+  0xddc3,0x3612 ,0xdd6e,0x35dc ,0xdd19,0x35a5 ,
+  0xdcc5,0x356e ,0xdc72,0x3537 ,0xdc1e,0x34ff ,
+  0xdbcb,0x34c6 ,0xdb78,0x348d ,0xdb26,0x3453 ,
+  0xdad4,0x3419 ,0xda82,0x33df ,0xda31,0x33a3 ,
+  0xd9e0,0x3368 ,0xd98f,0x332c ,0xd93f,0x32ef ,
+  0xd8ef,0x32b2 ,0xd8a0,0x3274 ,0xd851,0x3236 ,
+  0xd802,0x31f8 ,0xd7b4,0x31b9 ,0xd766,0x3179 ,
+  0xd719,0x3139 ,0xd6cb,0x30f9 ,0xd67f,0x30b8 ,
+  0xd632,0x3076 ,0xd5e6,0x3034 ,0xd59b,0x2ff2 ,
+  0xd550,0x2faf ,0xd505,0x2f6c ,0xd4bb,0x2f28 ,
+  0xd471,0x2ee4 ,0xd428,0x2e9f ,0xd3df,0x2e5a ,
+  0xd396,0x2e15 ,0xd34e,0x2dcf ,0xd306,0x2d88 ,
+  0xd2bf,0x2d41 ,0xd278,0x2cfa ,0xd231,0x2cb2 ,
+  0xd1eb,0x2c6a ,0xd1a6,0x2c21 ,0xd161,0x2bd8 ,
+  0xd11c,0x2b8f ,0xd0d8,0x2b45 ,0xd094,0x2afb ,
+  0xd051,0x2ab0 ,0xd00e,0x2a65 ,0xcfcc,0x2a1a ,
+  0xcf8a,0x29ce ,0xcf48,0x2981 ,0xcf07,0x2935 ,
+  0xcec7,0x28e7 ,0xce87,0x289a ,0xce47,0x284c ,
+  0xce08,0x27fe ,0xcdca,0x27af ,0xcd8c,0x2760 ,
+  0xcd4e,0x2711 ,0xcd11,0x26c1 ,0xccd4,0x2671 ,
+  0xcc98,0x2620 ,0xcc5d,0x25cf ,0xcc21,0x257e ,
+  0xcbe7,0x252c ,0xcbad,0x24da ,0xcb73,0x2488 ,
+  0xcb3a,0x2435 ,0xcb01,0x23e2 ,0xcac9,0x238e ,
+  0xca92,0x233b ,0xca5b,0x22e7 ,0xca24,0x2292 ,
+  0xc9ee,0x223d ,0xc9b8,0x21e8 ,0xc983,0x2193 ,
+  0xc94f,0x213d ,0xc91b,0x20e7 ,0xc8e8,0x2091 ,
+  0xc8b5,0x203a ,0xc882,0x1fe3 ,0xc850,0x1f8c ,
+  0xc81f,0x1f34 ,0xc7ee,0x1edc ,0xc7be,0x1e84 ,
+  0xc78f,0x1e2b ,0xc75f,0x1dd3 ,0xc731,0x1d79 ,
+  0xc703,0x1d20 ,0xc6d5,0x1cc6 ,0xc6a8,0x1c6c ,
+  0xc67c,0x1c12 ,0xc650,0x1bb8 ,0xc625,0x1b5d ,
+  0xc5fa,0x1b02 ,0xc5d0,0x1aa7 ,0xc5a7,0x1a4b ,
+  0xc57e,0x19ef ,0xc555,0x1993 ,0xc52d,0x1937 ,
+  0xc506,0x18db ,0xc4df,0x187e ,0xc4b9,0x1821 ,
+  0xc493,0x17c4 ,0xc46e,0x1766 ,0xc44a,0x1709 ,
+  0xc426,0x16ab ,0xc403,0x164c ,0xc3e0,0x15ee ,
+  0xc3be,0x1590 ,0xc39c,0x1531 ,0xc37b,0x14d2 ,
+  0xc35b,0x1473 ,0xc33b,0x1413 ,0xc31c,0x13b4 ,
+  0xc2fd,0x1354 ,0xc2df,0x12f4 ,0xc2c1,0x1294 ,
+  0xc2a5,0x1234 ,0xc288,0x11d3 ,0xc26d,0x1173 ,
+  0xc251,0x1112 ,0xc237,0x10b1 ,0xc21d,0x1050 ,
+  0xc204,0x0fee ,0xc1eb,0x0f8d ,0xc1d3,0x0f2b ,
+  0xc1bb,0x0eca ,0xc1a4,0x0e68 ,0xc18e,0x0e06 ,
+  0xc178,0x0da4 ,0xc163,0x0d41 ,0xc14f,0x0cdf ,
+  0xc13b,0x0c7c ,0xc128,0x0c1a ,0xc115,0x0bb7 ,
+  0xc103,0x0b54 ,0xc0f1,0x0af1 ,0xc0e0,0x0a8e ,
+  0xc0d0,0x0a2b ,0xc0c0,0x09c7 ,0xc0b1,0x0964 ,
+  0xc0a3,0x0901 ,0xc095,0x089d ,0xc088,0x0839 ,
+  0xc07b,0x07d6 ,0xc06f,0x0772 ,0xc064,0x070e ,
+  0xc059,0x06aa ,0xc04f,0x0646 ,0xc045,0x05e2 ,
+  0xc03c,0x057e ,0xc034,0x051a ,0xc02c,0x04b5 ,
+  0xc025,0x0451 ,0xc01f,0x03ed ,0xc019,0x0388 ,
+  0xc014,0x0324 ,0xc00f,0x02c0 ,0xc00b,0x025b ,
+  0xc008,0x01f7 ,0xc005,0x0192 ,0xc003,0x012e ,
+  0xc001,0x00c9 ,0xc000,0x0065 };
diff --git a/trunk/src/common_audio/signal_processing/webrtc_fft_t_rad.c b/trunk/src/common_audio/signal_processing/webrtc_fft_t_rad.c
new file mode 100644
index 0000000..13fbd9f
--- /dev/null
+++ b/trunk/src/common_audio/signal_processing/webrtc_fft_t_rad.c
@@ -0,0 +1,27 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This file contains the Q14 radix-2 tables used in ARM9E optimization routines.
+ *
+ */
+
+extern const unsigned short t_Q14S_rad8[2];
+const unsigned short t_Q14S_rad8[2] = {  0x0000,0x2d41 };
+
+//extern const int t_Q30S_rad8[2];
+//const int t_Q30S_rad8[2] = {  0x00000000,0x2d413ccd };
+
+extern const unsigned short t_Q14R_rad8[2];
+const unsigned short t_Q14R_rad8[2] = {  0x2d41,0x2d41 };
+
+//extern const int t_Q30R_rad8[2];
+//const int t_Q30R_rad8[2] = {  0x2d413ccd,0x2d413ccd };
diff --git a/trunk/src/common_audio/vad/Android.mk b/trunk/src/common_audio/vad/Android.mk
new file mode 100644
index 0000000..b7be3f0
--- /dev/null
+++ b/trunk/src/common_audio/vad/Android.mk
@@ -0,0 +1,50 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+include $(LOCAL_PATH)/../../../android-webrtc.mk
+
+LOCAL_ARM_MODE := arm
+LOCAL_MODULE_CLASS := STATIC_LIBRARIES
+LOCAL_MODULE := libwebrtc_vad
+LOCAL_MODULE_TAGS := optional
+LOCAL_SRC_FILES := \
+    webrtc_vad.c \
+    vad_core.c \
+    vad_filterbank.c \
+    vad_gmm.c \
+    vad_sp.c
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+    $(MY_WEBRTC_COMMON_DEFS)
+
+LOCAL_C_INCLUDES := \
+    $(LOCAL_PATH)/include \
+    $(LOCAL_PATH)/../.. \
+    $(LOCAL_PATH)/../signal_processing/include 
+
+LOCAL_SHARED_LIBRARIES := \
+    libdl \
+    libstlport
+
+ifeq ($(TARGET_OS)-$(TARGET_SIMULATOR),linux-true)
+LOCAL_LDLIBS += -ldl -lpthread
+endif
+
+ifneq ($(TARGET_SIMULATOR),true)
+LOCAL_SHARED_LIBRARIES += libdl
+endif
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/common_audio/vad/include/webrtc_vad.h b/trunk/src/common_audio/vad/include/webrtc_vad.h
new file mode 100644
index 0000000..3f9e402
--- /dev/null
+++ b/trunk/src/common_audio/vad/include/webrtc_vad.h
@@ -0,0 +1,130 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This header file includes the VAD API calls. Specific function calls are given below.
+ */
+
+#ifndef WEBRTC_COMMON_AUDIO_VAD_INCLUDE_WEBRTC_VAD_H_
+#define WEBRTC_COMMON_AUDIO_VAD_INCLUDE_WEBRTC_VAD_H_
+
+#include "typedefs.h"
+
+typedef struct WebRtcVadInst VadInst;
+
+#ifdef __cplusplus
+extern "C"
+{
+#endif
+
+/****************************************************************************
+ * WebRtcVad_AssignSize(...) 
+ *
+ * This functions get the size needed for storing the instance for encoder
+ * and decoder, respectively
+ *
+ * Input/Output:
+ *      - size_in_bytes : Pointer to integer where the size is returned
+ *
+ * Return value         : 0
+ */
+WebRtc_Word16 WebRtcVad_AssignSize(int *size_in_bytes);
+
+/****************************************************************************
+ * WebRtcVad_Assign(...) 
+ *
+ * This functions Assigns memory for the instances.
+ *
+ * Input:
+ *        - vad_inst_addr :  Address to where to assign memory
+ * Output:
+ *        - vad_inst      :  Pointer to the instance that should be created
+ *
+ * Return value           :  0 - Ok
+ *                          -1 - Error
+ */
+WebRtc_Word16 WebRtcVad_Assign(VadInst **vad_inst, void *vad_inst_addr);
+
+// Creates an instance to the VAD structure.
+//
+// - handle [o] : Pointer to the VAD instance that should be created.
+//
+// returns      : 0 - (OK), -1 - (Error)
+int WebRtcVad_Create(VadInst** handle);
+
+// Frees the dynamic memory of a specified VAD instance.
+//
+// - handle [i] : Pointer to VAD instance that should be freed.
+//
+// returns      : 0 - (OK), -1 - (NULL pointer in)
+int WebRtcVad_Free(VadInst* handle);
+
+/****************************************************************************
+ * WebRtcVad_Init(...)
+ *
+ * This function initializes a VAD instance
+ *
+ * Input:
+ *      - vad_inst      : Instance that should be initialized
+ *
+ * Output:
+ *      - vad_inst      : Initialized instance
+ *
+ * Return value         :  0 - Ok
+ *                        -1 - Error
+ */
+int WebRtcVad_Init(VadInst *vad_inst);
+
+/****************************************************************************
+ * WebRtcVad_set_mode(...)
+ *
+ * This function initializes a VAD instance
+ *
+ * Input:
+ *      - vad_inst      : VAD instance
+ *      - mode          : Aggressiveness setting (0, 1, 2, or 3) 
+ *
+ * Output:
+ *      - vad_inst      : Initialized instance
+ *
+ * Return value         :  0 - Ok
+ *                        -1 - Error
+ */
+int WebRtcVad_set_mode(VadInst *vad_inst, int mode);
+
+/****************************************************************************
+ * WebRtcVad_Process(...)
+ * 
+ * This functions does a VAD for the inserted speech frame
+ *
+ * Input
+ *        - vad_inst     : VAD Instance. Needs to be initiated before call.
+ *        - fs           : sampling frequency (Hz): 8000, 16000, or 32000
+ *        - speech_frame : Pointer to speech frame buffer
+ *        - frame_length : Length of speech frame buffer in number of samples
+ *
+ * Output:
+ *        - vad_inst     : Updated VAD instance
+ *
+ * Return value          :  1 - Active Voice
+ *                          0 - Non-active Voice
+ *                         -1 - Error
+ */
+WebRtc_Word16 WebRtcVad_Process(VadInst *vad_inst,
+                                WebRtc_Word16 fs,
+                                WebRtc_Word16 *speech_frame,
+                                WebRtc_Word16 frame_length);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif  // WEBRTC_COMMON_AUDIO_VAD_INCLUDE_WEBRTC_VAD_H_
diff --git a/trunk/src/common_audio/vad/vad.gypi b/trunk/src/common_audio/vad/vad.gypi
new file mode 100644
index 0000000..f15e651
--- /dev/null
+++ b/trunk/src/common_audio/vad/vad.gypi
@@ -0,0 +1,69 @@
+# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+  'targets': [
+    {
+      'target_name': 'vad',
+      'type': '<(library)',
+      'dependencies': [
+        'signal_processing',
+      ],
+      'include_dirs': [
+        'include',
+      ],
+      'direct_dependent_settings': {
+        'include_dirs': [
+          'include',
+        ],
+      },
+      'sources': [
+        'include/webrtc_vad.h',
+        'webrtc_vad.c',
+        'vad_core.c',
+        'vad_core.h',
+        'vad_defines.h',
+        'vad_filterbank.c',
+        'vad_filterbank.h',
+        'vad_gmm.c',
+        'vad_gmm.h',
+        'vad_sp.c',
+        'vad_sp.h',
+      ],
+    },
+  ], # targets
+   'conditions': [
+    ['build_with_chromium==0', {
+      'targets' : [
+        {
+          'target_name': 'vad_unittests',
+          'type': 'executable',
+          'dependencies': [
+            'vad',
+            '<(webrtc_root)/../test/test.gyp:test_support_main',
+            '<(webrtc_root)/../testing/gtest.gyp:gtest',
+          ],
+          'sources': [
+            'vad_core_unittest.cc',
+            'vad_filterbank_unittest.cc',
+            'vad_gmm_unittest.cc',
+            'vad_sp_unittest.cc',
+            'vad_unittest.cc',
+            'vad_unittest.h',
+          ],
+        }, # vad_unittests
+      ], # targets
+    }], # build_with_chromium
+  ], # conditions
+}
+
+# Local Variables:
+# tab-width:2
+# indent-tabs-mode:nil
+# End:
+# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/common_audio/vad/vad_core.c b/trunk/src/common_audio/vad/vad_core.c
new file mode 100644
index 0000000..2c6e6a9
--- /dev/null
+++ b/trunk/src/common_audio/vad/vad_core.c
@@ -0,0 +1,663 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "vad_core.h"
+
+#include "signal_processing_library.h"
+#include "typedefs.h"
+#include "vad_defines.h"
+#include "vad_filterbank.h"
+#include "vad_gmm.h"
+#include "vad_sp.h"
+
+// Spectrum Weighting
+static const WebRtc_Word16 kSpectrumWeight[6] = { 6, 8, 10, 12, 14, 16 };
+static const WebRtc_Word16 kNoiseUpdateConst = 655; // Q15
+static const WebRtc_Word16 kSpeechUpdateConst = 6554; // Q15
+static const WebRtc_Word16 kBackEta = 154; // Q8
+// Minimum difference between the two models, Q5
+static const WebRtc_Word16 kMinimumDifference[6] = {
+    544, 544, 576, 576, 576, 576 };
+// Upper limit of mean value for speech model, Q7
+static const WebRtc_Word16 kMaximumSpeech[6] = {
+    11392, 11392, 11520, 11520, 11520, 11520 };
+// Minimum value for mean value
+static const WebRtc_Word16 kMinimumMean[2] = { 640, 768 };
+// Upper limit of mean value for noise model, Q7
+static const WebRtc_Word16 kMaximumNoise[6] = {
+    9216, 9088, 8960, 8832, 8704, 8576 };
+// Start values for the Gaussian models, Q7
+// Weights for the two Gaussians for the six channels (noise)
+static const WebRtc_Word16 kNoiseDataWeights[12] = {
+    34, 62, 72, 66, 53, 25, 94, 66, 56, 62, 75, 103 };
+// Weights for the two Gaussians for the six channels (speech)
+static const WebRtc_Word16 kSpeechDataWeights[12] = {
+    48, 82, 45, 87, 50, 47, 80, 46, 83, 41, 78, 81 };
+// Means for the two Gaussians for the six channels (noise)
+static const WebRtc_Word16 kNoiseDataMeans[12] = {
+    6738, 4892, 7065, 6715, 6771, 3369, 7646, 3863, 7820, 7266, 5020, 4362 };
+// Means for the two Gaussians for the six channels (speech)
+static const WebRtc_Word16 kSpeechDataMeans[12] = {
+    8306, 10085, 10078, 11823, 11843, 6309, 9473, 9571, 10879, 7581, 8180, 7483
+};
+// Stds for the two Gaussians for the six channels (noise)
+static const WebRtc_Word16 kNoiseDataStds[12] = {
+    378, 1064, 493, 582, 688, 593, 474, 697, 475, 688, 421, 455 };
+// Stds for the two Gaussians for the six channels (speech)
+static const WebRtc_Word16 kSpeechDataStds[12] = {
+    555, 505, 567, 524, 585, 1231, 509, 828, 492, 1540, 1079, 850 };
+
+// Constants used in GmmProbability().
+//
+// Maximum number of counted speech (VAD = 1) frames in a row.
+static const int16_t kMaxSpeechFrames = 6;
+// Minimum standard deviation for both speech and noise.
+static const int16_t kMinStd = 384;
+
+// Constants in WebRtcVad_InitCore().
+// Default aggressiveness mode.
+static const short kDefaultMode = 0;
+static const int kInitCheck = 42;
+
+// Calculates the probabilities for both speech and background noise using
+// Gaussian Mixture Models. A hypothesis-test is performed to decide which type
+// of signal is most probable.
+//
+// - inst           [i/o] : Pointer to VAD instance
+// - feature_vector [i]   : Feature vector = log10(energy in frequency band)
+// - total_power    [i]   : Total power in audio frame.
+// - frame_length   [i]   : Number of input samples
+//
+// - returns              : the VAD decision (0 - noise, 1 - speech).
+static int16_t GmmProbability(VadInstT *inst, WebRtc_Word16 *feature_vector,
+                              WebRtc_Word16 total_power, int frame_length)
+{
+    int n, k;
+    WebRtc_Word16 backval;
+    WebRtc_Word16 h0, h1;
+    WebRtc_Word16 ratvec, xval;
+    WebRtc_Word16 vadflag;
+    WebRtc_Word16 shifts0, shifts1;
+    WebRtc_Word16 tmp16, tmp16_1, tmp16_2;
+    WebRtc_Word16 diff, nr, pos;
+    WebRtc_Word16 nmk, nmk2, nmk3, smk, smk2, nsk, ssk;
+    WebRtc_Word16 delt, ndelt;
+    WebRtc_Word16 maxspe, maxmu;
+    WebRtc_Word16 deltaN[NUM_TABLE_VALUES], deltaS[NUM_TABLE_VALUES];
+    WebRtc_Word16 ngprvec[NUM_TABLE_VALUES], sgprvec[NUM_TABLE_VALUES];
+    WebRtc_Word32 h0test, h1test;
+    WebRtc_Word32 tmp32_1, tmp32_2;
+    WebRtc_Word32 dotVal;
+    WebRtc_Word32 nmid, smid;
+    WebRtc_Word32 probn[NUM_MODELS], probs[NUM_MODELS];
+    WebRtc_Word16 *nmean1ptr, *nmean2ptr, *smean1ptr, *smean2ptr, *nstd1ptr, *nstd2ptr,
+            *sstd1ptr, *sstd2ptr;
+    WebRtc_Word16 overhead1, overhead2, individualTest, totalTest;
+
+    // Set the thresholds to different values based on frame length
+    if (frame_length == 80)
+    {
+        // 80 input samples
+        overhead1 = inst->over_hang_max_1[0];
+        overhead2 = inst->over_hang_max_2[0];
+        individualTest = inst->individual[0];
+        totalTest = inst->total[0];
+    } else if (frame_length == 160)
+    {
+        // 160 input samples
+        overhead1 = inst->over_hang_max_1[1];
+        overhead2 = inst->over_hang_max_2[1];
+        individualTest = inst->individual[1];
+        totalTest = inst->total[1];
+    } else
+    {
+        // 240 input samples
+        overhead1 = inst->over_hang_max_1[2];
+        overhead2 = inst->over_hang_max_2[2];
+        individualTest = inst->individual[2];
+        totalTest = inst->total[2];
+    }
+
+    if (total_power > MIN_ENERGY)
+    { // If signal present at all
+
+        // Set pointers to the gaussian parameters
+        nmean1ptr = &inst->noise_means[0];
+        nmean2ptr = &inst->noise_means[NUM_CHANNELS];
+        smean1ptr = &inst->speech_means[0];
+        smean2ptr = &inst->speech_means[NUM_CHANNELS];
+        nstd1ptr = &inst->noise_stds[0];
+        nstd2ptr = &inst->noise_stds[NUM_CHANNELS];
+        sstd1ptr = &inst->speech_stds[0];
+        sstd2ptr = &inst->speech_stds[NUM_CHANNELS];
+
+        vadflag = 0;
+        dotVal = 0;
+        for (n = 0; n < NUM_CHANNELS; n++)
+        { // For all channels
+
+            pos = WEBRTC_SPL_LSHIFT_W16(n, 1);
+            xval = feature_vector[n];
+
+            // Probability for Noise, Q7 * Q20 = Q27
+            tmp32_1 = WebRtcVad_GaussianProbability(xval, *nmean1ptr++, *nstd1ptr++,
+                                                    &deltaN[pos]);
+            probn[0] = (WebRtc_Word32)(kNoiseDataWeights[n] * tmp32_1);
+            tmp32_1 = WebRtcVad_GaussianProbability(xval, *nmean2ptr++, *nstd2ptr++,
+                                                    &deltaN[pos + 1]);
+            probn[1] = (WebRtc_Word32)(kNoiseDataWeights[n + NUM_CHANNELS] * tmp32_1);
+            h0test = probn[0] + probn[1]; // Q27
+            h0 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(h0test, 12); // Q15
+
+            // Probability for Speech
+            tmp32_1 = WebRtcVad_GaussianProbability(xval, *smean1ptr++, *sstd1ptr++,
+                                                    &deltaS[pos]);
+            probs[0] = (WebRtc_Word32)(kSpeechDataWeights[n] * tmp32_1);
+            tmp32_1 = WebRtcVad_GaussianProbability(xval, *smean2ptr++, *sstd2ptr++,
+                                                    &deltaS[pos + 1]);
+            probs[1] = (WebRtc_Word32)(kSpeechDataWeights[n + NUM_CHANNELS] * tmp32_1);
+            h1test = probs[0] + probs[1]; // Q27
+            h1 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(h1test, 12); // Q15
+
+            // Get likelihood ratio. Approximate log2(H1/H0) with shifts0 - shifts1
+            shifts0 = WebRtcSpl_NormW32(h0test);
+            shifts1 = WebRtcSpl_NormW32(h1test);
+
+            if ((h0test > 0) && (h1test > 0))
+            {
+                ratvec = shifts0 - shifts1;
+            } else if (h1test > 0)
+            {
+                ratvec = 31 - shifts1;
+            } else if (h0test > 0)
+            {
+                ratvec = shifts0 - 31;
+            } else
+            {
+                ratvec = 0;
+            }
+
+            // VAD decision with spectrum weighting
+            dotVal += WEBRTC_SPL_MUL_16_16(ratvec, kSpectrumWeight[n]);
+
+            // Individual channel test
+            if ((ratvec << 2) > individualTest)
+            {
+                vadflag = 1;
+            }
+
+            // Probabilities used when updating model
+            if (h0 > 0)
+            {
+                tmp32_1 = probn[0] & 0xFFFFF000; // Q27
+                tmp32_2 = WEBRTC_SPL_LSHIFT_W32(tmp32_1, 2); // Q29
+                ngprvec[pos] = (WebRtc_Word16)WebRtcSpl_DivW32W16(tmp32_2, h0);
+                ngprvec[pos + 1] = 16384 - ngprvec[pos];
+            } else
+            {
+                ngprvec[pos] = 16384;
+                ngprvec[pos + 1] = 0;
+            }
+
+            // Probabilities used when updating model
+            if (h1 > 0)
+            {
+                tmp32_1 = probs[0] & 0xFFFFF000;
+                tmp32_2 = WEBRTC_SPL_LSHIFT_W32(tmp32_1, 2);
+                sgprvec[pos] = (WebRtc_Word16)WebRtcSpl_DivW32W16(tmp32_2, h1);
+                sgprvec[pos + 1] = 16384 - sgprvec[pos];
+            } else
+            {
+                sgprvec[pos] = 0;
+                sgprvec[pos + 1] = 0;
+            }
+        }
+
+        // Overall test
+        if (dotVal >= totalTest)
+        {
+            vadflag |= 1;
+        }
+
+        // Set pointers to the means and standard deviations.
+        nmean1ptr = &inst->noise_means[0];
+        smean1ptr = &inst->speech_means[0];
+        nstd1ptr = &inst->noise_stds[0];
+        sstd1ptr = &inst->speech_stds[0];
+
+        maxspe = 12800;
+
+        // Update the model's parameters
+        for (n = 0; n < NUM_CHANNELS; n++)
+        {
+
+            pos = WEBRTC_SPL_LSHIFT_W16(n, 1);
+
+            // Get min value in past which is used for long term correction
+            backval = WebRtcVad_FindMinimum(inst, feature_vector[n], n); // Q4
+
+            // Compute the "global" mean, that is the sum of the two means weighted
+            nmid = WEBRTC_SPL_MUL_16_16(kNoiseDataWeights[n], *nmean1ptr); // Q7 * Q7
+            nmid += WEBRTC_SPL_MUL_16_16(kNoiseDataWeights[n+NUM_CHANNELS],
+                    *(nmean1ptr+NUM_CHANNELS));
+            tmp16_1 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(nmid, 6); // Q8
+
+            for (k = 0; k < NUM_MODELS; k++)
+            {
+
+                nr = pos + k;
+
+                nmean2ptr = nmean1ptr + k * NUM_CHANNELS;
+                smean2ptr = smean1ptr + k * NUM_CHANNELS;
+                nstd2ptr = nstd1ptr + k * NUM_CHANNELS;
+                sstd2ptr = sstd1ptr + k * NUM_CHANNELS;
+                nmk = *nmean2ptr;
+                smk = *smean2ptr;
+                nsk = *nstd2ptr;
+                ssk = *sstd2ptr;
+
+                // Update noise mean vector if the frame consists of noise only
+                nmk2 = nmk;
+                if (!vadflag)
+                {
+                    // deltaN = (x-mu)/sigma^2
+                    // ngprvec[k] = probn[k]/(probn[0] + probn[1])
+
+                    delt = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ngprvec[nr],
+                            deltaN[nr], 11); // Q14*Q11
+                    nmk2 = nmk + (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(delt,
+                            kNoiseUpdateConst,
+                            22); // Q7+(Q14*Q15>>22)
+                }
+
+                // Long term correction of the noise mean
+                ndelt = WEBRTC_SPL_LSHIFT_W16(backval, 4);
+                ndelt -= tmp16_1; // Q8 - Q8
+                nmk3 = nmk2 + (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ndelt,
+                        kBackEta,
+                        9); // Q7+(Q8*Q8)>>9
+
+                // Control that the noise mean does not drift to much
+                tmp16 = WEBRTC_SPL_LSHIFT_W16(k+5, 7);
+                if (nmk3 < tmp16)
+                    nmk3 = tmp16;
+                tmp16 = WEBRTC_SPL_LSHIFT_W16(72+k-n, 7);
+                if (nmk3 > tmp16)
+                    nmk3 = tmp16;
+                *nmean2ptr = nmk3;
+
+                if (vadflag)
+                {
+                    // Update speech mean vector:
+                    // deltaS = (x-mu)/sigma^2
+                    // sgprvec[k] = probn[k]/(probn[0] + probn[1])
+
+                    delt = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sgprvec[nr],
+                            deltaS[nr],
+                            11); // (Q14*Q11)>>11=Q14
+                    tmp16 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(delt,
+                            kSpeechUpdateConst,
+                            21) + 1;
+                    smk2 = smk + (tmp16 >> 1); // Q7 + (Q14 * Q15 >> 22)
+
+                    // Control that the speech mean does not drift to much
+                    maxmu = maxspe + 640;
+                    if (smk2 < kMinimumMean[k])
+                        smk2 = kMinimumMean[k];
+                    if (smk2 > maxmu)
+                        smk2 = maxmu;
+
+                    *smean2ptr = smk2;
+
+                    // (Q7>>3) = Q4
+                    tmp16 = WEBRTC_SPL_RSHIFT_W16((smk + 4), 3);
+
+                    tmp16 = feature_vector[n] - tmp16; // Q4
+                    tmp32_1 = WEBRTC_SPL_MUL_16_16_RSFT(deltaS[nr], tmp16, 3);
+                    tmp32_2 = tmp32_1 - (WebRtc_Word32)4096; // Q12
+                    tmp16 = WEBRTC_SPL_RSHIFT_W16((sgprvec[nr]), 2);
+                    tmp32_1 = (WebRtc_Word32)(tmp16 * tmp32_2);// (Q15>>3)*(Q14>>2)=Q12*Q12=Q24
+
+                    tmp32_2 = WEBRTC_SPL_RSHIFT_W32(tmp32_1, 4); // Q20
+
+                    // 0.1 * Q20 / Q7 = Q13
+                    if (tmp32_2 > 0)
+                        tmp16 = (WebRtc_Word16)WebRtcSpl_DivW32W16(tmp32_2, ssk * 10);
+                    else
+                    {
+                        tmp16 = (WebRtc_Word16)WebRtcSpl_DivW32W16(-tmp32_2, ssk * 10);
+                        tmp16 = -tmp16;
+                    }
+                    // divide by 4 giving an update factor of 0.025
+                    tmp16 += 128; // Rounding
+                    ssk += WEBRTC_SPL_RSHIFT_W16(tmp16, 8);
+                    // Division with 8 plus Q7
+                    if (ssk < kMinStd)
+                        ssk = kMinStd;
+                    *sstd2ptr = ssk;
+                } else
+                {
+                    // Update GMM variance vectors
+                    // deltaN * (feature_vector[n] - nmk) - 1, Q11 * Q4
+                    tmp16 = feature_vector[n] - WEBRTC_SPL_RSHIFT_W16(nmk, 3);
+
+                    // (Q15>>3) * (Q14>>2) = Q12 * Q12 = Q24
+                    tmp32_1 = WEBRTC_SPL_MUL_16_16_RSFT(deltaN[nr], tmp16, 3) - 4096;
+                    tmp16 = WEBRTC_SPL_RSHIFT_W16((ngprvec[nr]+2), 2);
+                    tmp32_2 = (WebRtc_Word32)(tmp16 * tmp32_1);
+                    tmp32_1 = WEBRTC_SPL_RSHIFT_W32(tmp32_2, 14);
+                    // Q20  * approx 0.001 (2^-10=0.0009766)
+
+                    // Q20 / Q7 = Q13
+                    tmp16 = (WebRtc_Word16)WebRtcSpl_DivW32W16(tmp32_1, nsk);
+                    if (tmp32_1 > 0)
+                        tmp16 = (WebRtc_Word16)WebRtcSpl_DivW32W16(tmp32_1, nsk);
+                    else
+                    {
+                        tmp16 = (WebRtc_Word16)WebRtcSpl_DivW32W16(-tmp32_1, nsk);
+                        tmp16 = -tmp16;
+                    }
+                    tmp16 += 32; // Rounding
+                    nsk += WEBRTC_SPL_RSHIFT_W16(tmp16, 6);
+
+                    if (nsk < kMinStd)
+                        nsk = kMinStd;
+
+                    *nstd2ptr = nsk;
+                }
+            }
+
+            // Separate models if they are too close - nmid in Q14
+            nmid = WEBRTC_SPL_MUL_16_16(kNoiseDataWeights[n], *nmean1ptr);
+            nmid += WEBRTC_SPL_MUL_16_16(kNoiseDataWeights[n+NUM_CHANNELS], *nmean2ptr);
+
+            // smid in Q14
+            smid = WEBRTC_SPL_MUL_16_16(kSpeechDataWeights[n], *smean1ptr);
+            smid += WEBRTC_SPL_MUL_16_16(kSpeechDataWeights[n+NUM_CHANNELS], *smean2ptr);
+
+            // diff = "global" speech mean - "global" noise mean
+            diff = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(smid, 9);
+            tmp16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(nmid, 9);
+            diff -= tmp16;
+
+            if (diff < kMinimumDifference[n])
+            {
+
+                tmp16 = kMinimumDifference[n] - diff; // Q5
+
+                // tmp16_1 = ~0.8 * (kMinimumDifference - diff) in Q7
+                // tmp16_2 = ~0.2 * (kMinimumDifference - diff) in Q7
+                tmp16_1 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(13, tmp16, 2);
+                tmp16_2 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(3, tmp16, 2);
+
+                // First Gauss, speech model
+                tmp16 = tmp16_1 + *smean1ptr;
+                *smean1ptr = tmp16;
+                smid = WEBRTC_SPL_MUL_16_16(tmp16, kSpeechDataWeights[n]);
+
+                // Second Gauss, speech model
+                tmp16 = tmp16_1 + *smean2ptr;
+                *smean2ptr = tmp16;
+                smid += WEBRTC_SPL_MUL_16_16(tmp16, kSpeechDataWeights[n+NUM_CHANNELS]);
+
+                // First Gauss, noise model
+                tmp16 = *nmean1ptr - tmp16_2;
+                *nmean1ptr = tmp16;
+
+                nmid = WEBRTC_SPL_MUL_16_16(tmp16, kNoiseDataWeights[n]);
+
+                // Second Gauss, noise model
+                tmp16 = *nmean2ptr - tmp16_2;
+                *nmean2ptr = tmp16;
+                nmid += WEBRTC_SPL_MUL_16_16(tmp16, kNoiseDataWeights[n+NUM_CHANNELS]);
+            }
+
+            // Control that the speech & noise means do not drift to much
+            maxspe = kMaximumSpeech[n];
+            tmp16_2 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(smid, 7);
+            if (tmp16_2 > maxspe)
+            { // Upper limit of speech model
+                tmp16_2 -= maxspe;
+
+                *smean1ptr -= tmp16_2;
+                *smean2ptr -= tmp16_2;
+            }
+
+            tmp16_2 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(nmid, 7);
+            if (tmp16_2 > kMaximumNoise[n])
+            {
+                tmp16_2 -= kMaximumNoise[n];
+
+                *nmean1ptr -= tmp16_2;
+                *nmean2ptr -= tmp16_2;
+            }
+
+            nmean1ptr++;
+            smean1ptr++;
+            nstd1ptr++;
+            sstd1ptr++;
+        }
+        inst->frame_counter++;
+    } else
+    {
+        vadflag = 0;
+    }
+
+    // Hangover smoothing
+    if (!vadflag)
+    {
+        if (inst->over_hang > 0)
+        {
+            vadflag = 2 + inst->over_hang;
+            inst->over_hang = inst->over_hang - 1;
+        }
+        inst->num_of_speech = 0;
+    } else
+    {
+        inst->num_of_speech = inst->num_of_speech + 1;
+        if (inst->num_of_speech > kMaxSpeechFrames)
+        {
+            inst->num_of_speech = kMaxSpeechFrames;
+            inst->over_hang = overhead2;
+        } else
+            inst->over_hang = overhead1;
+    }
+    return vadflag;
+}
+
+// Initialize the VAD. Set aggressiveness mode to default value.
+int WebRtcVad_InitCore(VadInstT* self) {
+  int i;
+
+  if (self == NULL) {
+    return -1;
+  }
+
+  // Initialization of general struct variables.
+  self->vad = 1;  // Speech active (=1).
+  self->frame_counter = 0;
+  self->over_hang = 0;
+  self->num_of_speech = 0;
+
+  // Initialization of downsampling filter state.
+  memset(self->downsampling_filter_states, 0,
+         sizeof(self->downsampling_filter_states));
+
+  // Read initial PDF parameters.
+  for (i = 0; i < NUM_TABLE_VALUES; i++) {
+    self->noise_means[i] = kNoiseDataMeans[i];
+    self->speech_means[i] = kSpeechDataMeans[i];
+    self->noise_stds[i] = kNoiseDataStds[i];
+    self->speech_stds[i] = kSpeechDataStds[i];
+  }
+
+  // Initialize Index and Minimum value vectors.
+  for (i = 0; i < 16 * NUM_CHANNELS; i++) {
+    self->low_value_vector[i] = 10000;
+    self->index_vector[i] = 0;
+  }
+
+  // Initialize splitting filter states.
+  memset(self->upper_state, 0, sizeof(self->upper_state));
+  memset(self->lower_state, 0, sizeof(self->lower_state));
+
+  // Initialize high pass filter states.
+  memset(self->hp_filter_state, 0, sizeof(self->hp_filter_state));
+
+  // Initialize mean value memory, for WebRtcVad_FindMinimum().
+  for (i = 0; i < NUM_CHANNELS; i++) {
+    self->mean_value[i] = 1600;
+  }
+
+  // Set aggressiveness mode to default (=|kDefaultMode|).
+  if (WebRtcVad_set_mode_core(self, kDefaultMode) != 0) {
+    return -1;
+  }
+
+  self->init_flag = kInitCheck;
+
+  return 0;
+}
+
+// Set aggressiveness mode
+int WebRtcVad_set_mode_core(VadInstT *inst, int mode)
+{
+
+    if (mode == 0)
+    {
+        // Quality mode
+        inst->over_hang_max_1[0] = OHMAX1_10MS_Q; // Overhang short speech burst
+        inst->over_hang_max_1[1] = OHMAX1_20MS_Q; // Overhang short speech burst
+        inst->over_hang_max_1[2] = OHMAX1_30MS_Q; // Overhang short speech burst
+        inst->over_hang_max_2[0] = OHMAX2_10MS_Q; // Overhang long speech burst
+        inst->over_hang_max_2[1] = OHMAX2_20MS_Q; // Overhang long speech burst
+        inst->over_hang_max_2[2] = OHMAX2_30MS_Q; // Overhang long speech burst
+
+        inst->individual[0] = INDIVIDUAL_10MS_Q;
+        inst->individual[1] = INDIVIDUAL_20MS_Q;
+        inst->individual[2] = INDIVIDUAL_30MS_Q;
+
+        inst->total[0] = TOTAL_10MS_Q;
+        inst->total[1] = TOTAL_20MS_Q;
+        inst->total[2] = TOTAL_30MS_Q;
+    } else if (mode == 1)
+    {
+        // Low bitrate mode
+        inst->over_hang_max_1[0] = OHMAX1_10MS_LBR; // Overhang short speech burst
+        inst->over_hang_max_1[1] = OHMAX1_20MS_LBR; // Overhang short speech burst
+        inst->over_hang_max_1[2] = OHMAX1_30MS_LBR; // Overhang short speech burst
+        inst->over_hang_max_2[0] = OHMAX2_10MS_LBR; // Overhang long speech burst
+        inst->over_hang_max_2[1] = OHMAX2_20MS_LBR; // Overhang long speech burst
+        inst->over_hang_max_2[2] = OHMAX2_30MS_LBR; // Overhang long speech burst
+
+        inst->individual[0] = INDIVIDUAL_10MS_LBR;
+        inst->individual[1] = INDIVIDUAL_20MS_LBR;
+        inst->individual[2] = INDIVIDUAL_30MS_LBR;
+
+        inst->total[0] = TOTAL_10MS_LBR;
+        inst->total[1] = TOTAL_20MS_LBR;
+        inst->total[2] = TOTAL_30MS_LBR;
+    } else if (mode == 2)
+    {
+        // Aggressive mode
+        inst->over_hang_max_1[0] = OHMAX1_10MS_AGG; // Overhang short speech burst
+        inst->over_hang_max_1[1] = OHMAX1_20MS_AGG; // Overhang short speech burst
+        inst->over_hang_max_1[2] = OHMAX1_30MS_AGG; // Overhang short speech burst
+        inst->over_hang_max_2[0] = OHMAX2_10MS_AGG; // Overhang long speech burst
+        inst->over_hang_max_2[1] = OHMAX2_20MS_AGG; // Overhang long speech burst
+        inst->over_hang_max_2[2] = OHMAX2_30MS_AGG; // Overhang long speech burst
+
+        inst->individual[0] = INDIVIDUAL_10MS_AGG;
+        inst->individual[1] = INDIVIDUAL_20MS_AGG;
+        inst->individual[2] = INDIVIDUAL_30MS_AGG;
+
+        inst->total[0] = TOTAL_10MS_AGG;
+        inst->total[1] = TOTAL_20MS_AGG;
+        inst->total[2] = TOTAL_30MS_AGG;
+    } else if (mode == 3)
+    {
+        // Very aggressive mode
+        inst->over_hang_max_1[0] = OHMAX1_10MS_VAG; // Overhang short speech burst
+        inst->over_hang_max_1[1] = OHMAX1_20MS_VAG; // Overhang short speech burst
+        inst->over_hang_max_1[2] = OHMAX1_30MS_VAG; // Overhang short speech burst
+        inst->over_hang_max_2[0] = OHMAX2_10MS_VAG; // Overhang long speech burst
+        inst->over_hang_max_2[1] = OHMAX2_20MS_VAG; // Overhang long speech burst
+        inst->over_hang_max_2[2] = OHMAX2_30MS_VAG; // Overhang long speech burst
+
+        inst->individual[0] = INDIVIDUAL_10MS_VAG;
+        inst->individual[1] = INDIVIDUAL_20MS_VAG;
+        inst->individual[2] = INDIVIDUAL_30MS_VAG;
+
+        inst->total[0] = TOTAL_10MS_VAG;
+        inst->total[1] = TOTAL_20MS_VAG;
+        inst->total[2] = TOTAL_30MS_VAG;
+    } else
+    {
+        return -1;
+    }
+
+    return 0;
+}
+
+// Calculate VAD decision by first extracting feature values and then calculate
+// probability for both speech and background noise.
+
+WebRtc_Word16 WebRtcVad_CalcVad32khz(VadInstT *inst, WebRtc_Word16 *speech_frame,
+                                     int frame_length)
+{
+    WebRtc_Word16 len, vad;
+    WebRtc_Word16 speechWB[480]; // Downsampled speech frame: 960 samples (30ms in SWB)
+    WebRtc_Word16 speechNB[240]; // Downsampled speech frame: 480 samples (30ms in WB)
+
+
+    // Downsample signal 32->16->8 before doing VAD
+    WebRtcVad_Downsampling(speech_frame, speechWB, &(inst->downsampling_filter_states[2]),
+                           frame_length);
+    len = WEBRTC_SPL_RSHIFT_W16(frame_length, 1);
+
+    WebRtcVad_Downsampling(speechWB, speechNB, inst->downsampling_filter_states, len);
+    len = WEBRTC_SPL_RSHIFT_W16(len, 1);
+
+    // Do VAD on an 8 kHz signal
+    vad = WebRtcVad_CalcVad8khz(inst, speechNB, len);
+
+    return vad;
+}
+
+WebRtc_Word16 WebRtcVad_CalcVad16khz(VadInstT *inst, WebRtc_Word16 *speech_frame,
+                                     int frame_length)
+{
+    WebRtc_Word16 len, vad;
+    WebRtc_Word16 speechNB[240]; // Downsampled speech frame: 480 samples (30ms in WB)
+
+    // Wideband: Downsample signal before doing VAD
+    WebRtcVad_Downsampling(speech_frame, speechNB, inst->downsampling_filter_states,
+                           frame_length);
+
+    len = WEBRTC_SPL_RSHIFT_W16(frame_length, 1);
+    vad = WebRtcVad_CalcVad8khz(inst, speechNB, len);
+
+    return vad;
+}
+
+WebRtc_Word16 WebRtcVad_CalcVad8khz(VadInstT *inst, WebRtc_Word16 *speech_frame,
+                                    int frame_length)
+{
+    WebRtc_Word16 feature_vector[NUM_CHANNELS], total_power;
+
+    // Get power in the bands
+    total_power = WebRtcVad_CalculateFeatures(inst, speech_frame, frame_length,
+                                              feature_vector);
+
+    // Make a VAD
+    inst->vad = GmmProbability(inst, feature_vector, total_power, frame_length);
+
+    return inst->vad;
+}
diff --git a/trunk/src/common_audio/vad/vad_core.h b/trunk/src/common_audio/vad/vad_core.h
new file mode 100644
index 0000000..c82fbce
--- /dev/null
+++ b/trunk/src/common_audio/vad/vad_core.h
@@ -0,0 +1,106 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This header file includes the descriptions of the core VAD calls.
+ */
+
+#ifndef WEBRTC_COMMON_AUDIO_VAD_VAD_CORE_H_
+#define WEBRTC_COMMON_AUDIO_VAD_VAD_CORE_H_
+
+#include "typedefs.h"
+#include "vad_defines.h"
+
+typedef struct VadInstT_
+{
+
+    WebRtc_Word16 vad;
+    WebRtc_Word32 downsampling_filter_states[4];
+    WebRtc_Word16 noise_means[NUM_TABLE_VALUES];
+    WebRtc_Word16 speech_means[NUM_TABLE_VALUES];
+    WebRtc_Word16 noise_stds[NUM_TABLE_VALUES];
+    WebRtc_Word16 speech_stds[NUM_TABLE_VALUES];
+    // TODO(bjornv): Change to |frame_count|.
+    WebRtc_Word32 frame_counter;
+    WebRtc_Word16 over_hang; // Over Hang
+    WebRtc_Word16 num_of_speech;
+    // TODO(bjornv): Change to |age_vector|.
+    WebRtc_Word16 index_vector[16 * NUM_CHANNELS];
+    WebRtc_Word16 low_value_vector[16 * NUM_CHANNELS];
+    // TODO(bjornv): Change to |median|.
+    WebRtc_Word16 mean_value[NUM_CHANNELS];
+    WebRtc_Word16 upper_state[5];
+    WebRtc_Word16 lower_state[5];
+    WebRtc_Word16 hp_filter_state[4];
+    WebRtc_Word16 over_hang_max_1[3];
+    WebRtc_Word16 over_hang_max_2[3];
+    WebRtc_Word16 individual[3];
+    WebRtc_Word16 total[3];
+
+    int init_flag;
+
+} VadInstT;
+
+// Initializes the core VAD component. The default aggressiveness mode is
+// controlled by |kDefaultMode| in vad_core.c.
+//
+// - self [i/o] : Instance that should be initialized
+//
+// returns      : 0 (OK), -1 (NULL pointer in or if the default mode can't be
+//                set)
+int WebRtcVad_InitCore(VadInstT* self);
+
+/****************************************************************************
+ * WebRtcVad_set_mode_core(...)
+ *
+ * This function changes the VAD settings
+ *
+ * Input:
+ *      - inst      : VAD instance
+ *      - mode      : Aggressiveness degree
+ *                    0 (High quality) - 3 (Highly aggressive)
+ *
+ * Output:
+ *      - inst      : Changed  instance
+ *
+ * Return value     :  0 - Ok
+ *                    -1 - Error
+ */
+
+int WebRtcVad_set_mode_core(VadInstT* inst, int mode);
+
+/****************************************************************************
+ * WebRtcVad_CalcVad32khz(...) 
+ * WebRtcVad_CalcVad16khz(...) 
+ * WebRtcVad_CalcVad8khz(...) 
+ *
+ * Calculate probability for active speech and make VAD decision.
+ *
+ * Input:
+ *      - inst          : Instance that should be initialized
+ *      - speech_frame  : Input speech frame
+ *      - frame_length  : Number of input samples
+ *
+ * Output:
+ *      - inst          : Updated filter states etc.
+ *
+ * Return value         : VAD decision
+ *                        0 - No active speech
+ *                        1-6 - Active speech
+ */
+WebRtc_Word16 WebRtcVad_CalcVad32khz(VadInstT* inst, WebRtc_Word16* speech_frame,
+                                     int frame_length);
+WebRtc_Word16 WebRtcVad_CalcVad16khz(VadInstT* inst, WebRtc_Word16* speech_frame,
+                                     int frame_length);
+WebRtc_Word16 WebRtcVad_CalcVad8khz(VadInstT* inst, WebRtc_Word16* speech_frame,
+                                    int frame_length);
+
+#endif  // WEBRTC_COMMON_AUDIO_VAD_VAD_CORE_H_
diff --git a/trunk/src/common_audio/vad/vad_core_unittest.cc b/trunk/src/common_audio/vad/vad_core_unittest.cc
new file mode 100644
index 0000000..141b796
--- /dev/null
+++ b/trunk/src/common_audio/vad/vad_core_unittest.cc
@@ -0,0 +1,99 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdlib.h>
+
+#include "gtest/gtest.h"
+#include "typedefs.h"
+#include "vad_unittest.h"
+
+extern "C" {
+#include "vad_core.h"
+}
+
+namespace {
+
+TEST_F(VadTest, InitCore) {
+  // Test WebRtcVad_InitCore().
+  VadInstT* self = reinterpret_cast<VadInstT*>(malloc(sizeof(VadInstT)));
+
+  // NULL pointer test.
+  EXPECT_EQ(-1, WebRtcVad_InitCore(NULL));
+
+  // Verify return = 0 for non-NULL pointer.
+  EXPECT_EQ(0, WebRtcVad_InitCore(self));
+  // Verify init_flag is set.
+  EXPECT_EQ(42, self->init_flag);
+
+  free(self);
+}
+
+TEST_F(VadTest, set_mode_core) {
+  VadInstT* self = reinterpret_cast<VadInstT*>(malloc(sizeof(VadInstT)));
+
+  // TODO(bjornv): Add NULL pointer check if we take care of it in
+  // vad_core.c
+
+  ASSERT_EQ(0, WebRtcVad_InitCore(self));
+  // Test WebRtcVad_set_mode_core().
+  // Invalid modes should return -1.
+  EXPECT_EQ(-1, WebRtcVad_set_mode_core(self, -1));
+  EXPECT_EQ(-1, WebRtcVad_set_mode_core(self, 1000));
+  // Valid modes should return 0.
+  for (size_t j = 0; j < kModesSize; ++j) {
+    EXPECT_EQ(0, WebRtcVad_set_mode_core(self, kModes[j]));
+  }
+
+  free(self);
+}
+
+TEST_F(VadTest, CalcVad) {
+  VadInstT* self = reinterpret_cast<VadInstT*>(malloc(sizeof(VadInstT)));
+  int16_t speech[kMaxFrameLength];
+
+  // TODO(bjornv): Add NULL pointer check if we take care of it in
+  // vad_core.c
+
+  // Test WebRtcVad_CalcVadXXkhz()
+  // Verify that all zeros in gives VAD = 0 out.
+  memset(speech, 0, sizeof(speech));
+  ASSERT_EQ(0, WebRtcVad_InitCore(self));
+  for (size_t j = 0; j < kFrameLengthsSize; ++j) {
+    if (ValidRatesAndFrameLengths(8000, kFrameLengths[j])) {
+      EXPECT_EQ(0, WebRtcVad_CalcVad8khz(self, speech, kFrameLengths[j]));
+    }
+    if (ValidRatesAndFrameLengths(16000, kFrameLengths[j])) {
+      EXPECT_EQ(0, WebRtcVad_CalcVad16khz(self, speech, kFrameLengths[j]));
+    }
+    if (ValidRatesAndFrameLengths(32000, kFrameLengths[j])) {
+      EXPECT_EQ(0, WebRtcVad_CalcVad32khz(self, speech, kFrameLengths[j]));
+    }
+  }
+
+  // Construct a speech signal that will trigger the VAD in all modes. It is
+  // known that (i * i) will wrap around, but that doesn't matter in this case.
+  for (int16_t i = 0; i < kMaxFrameLength; ++i) {
+    speech[i] = (i * i);
+  }
+  for (size_t j = 0; j < kFrameLengthsSize; ++j) {
+    if (ValidRatesAndFrameLengths(8000, kFrameLengths[j])) {
+      EXPECT_EQ(1, WebRtcVad_CalcVad8khz(self, speech, kFrameLengths[j]));
+    }
+    if (ValidRatesAndFrameLengths(16000, kFrameLengths[j])) {
+      EXPECT_EQ(1, WebRtcVad_CalcVad16khz(self, speech, kFrameLengths[j]));
+    }
+    if (ValidRatesAndFrameLengths(32000, kFrameLengths[j])) {
+      EXPECT_EQ(1, WebRtcVad_CalcVad32khz(self, speech, kFrameLengths[j]));
+    }
+  }
+
+  free(self);
+}
+}  // namespace
diff --git a/trunk/src/common_audio/vad/vad_defines.h b/trunk/src/common_audio/vad/vad_defines.h
new file mode 100644
index 0000000..5d1539d
--- /dev/null
+++ b/trunk/src/common_audio/vad/vad_defines.h
@@ -0,0 +1,93 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+
+/*
+ * This header file includes the macros used in VAD.
+ */
+
+#ifndef WEBRTC_VAD_DEFINES_H_
+#define WEBRTC_VAD_DEFINES_H_
+
+#define NUM_CHANNELS        6   // Eight frequency bands
+#define NUM_MODELS          2   // Number of Gaussian models
+#define NUM_TABLE_VALUES    NUM_CHANNELS * NUM_MODELS
+
+#define MIN_ENERGY          10
+#define ALPHA1              6553    // 0.2 in Q15
+#define ALPHA2              32439   // 0.99 in Q15
+// Mode 0, Quality thresholds - Different thresholds for the different frame lengths
+#define INDIVIDUAL_10MS_Q   24
+#define INDIVIDUAL_20MS_Q   21      // (log10(2)*66)<<2 ~=16
+#define INDIVIDUAL_30MS_Q   24
+
+#define TOTAL_10MS_Q        57
+#define TOTAL_20MS_Q        48
+#define TOTAL_30MS_Q        57
+
+#define OHMAX1_10MS_Q       8  // Max Overhang 1
+#define OHMAX2_10MS_Q       14 // Max Overhang 2
+#define OHMAX1_20MS_Q       4  // Max Overhang 1
+#define OHMAX2_20MS_Q       7  // Max Overhang 2
+#define OHMAX1_30MS_Q       3
+#define OHMAX2_30MS_Q       5
+
+// Mode 1, Low bitrate thresholds - Different thresholds for the different frame lengths
+#define INDIVIDUAL_10MS_LBR 37
+#define INDIVIDUAL_20MS_LBR 32
+#define INDIVIDUAL_30MS_LBR 37
+
+#define TOTAL_10MS_LBR      100
+#define TOTAL_20MS_LBR      80
+#define TOTAL_30MS_LBR      100
+
+#define OHMAX1_10MS_LBR     8  // Max Overhang 1
+#define OHMAX2_10MS_LBR     14 // Max Overhang 2
+#define OHMAX1_20MS_LBR     4
+#define OHMAX2_20MS_LBR     7
+
+#define OHMAX1_30MS_LBR     3
+#define OHMAX2_30MS_LBR     5
+
+// Mode 2, Very aggressive thresholds - Different thresholds for the different frame lengths
+#define INDIVIDUAL_10MS_AGG 82
+#define INDIVIDUAL_20MS_AGG 78
+#define INDIVIDUAL_30MS_AGG 82
+
+#define TOTAL_10MS_AGG      285 //580
+#define TOTAL_20MS_AGG      260
+#define TOTAL_30MS_AGG      285
+
+#define OHMAX1_10MS_AGG     6  // Max Overhang 1
+#define OHMAX2_10MS_AGG     9  // Max Overhang 2
+#define OHMAX1_20MS_AGG     3
+#define OHMAX2_20MS_AGG     5
+
+#define OHMAX1_30MS_AGG     2
+#define OHMAX2_30MS_AGG     3
+
+// Mode 3, Super aggressive thresholds - Different thresholds for the different frame lengths
+#define INDIVIDUAL_10MS_VAG 94
+#define INDIVIDUAL_20MS_VAG 94
+#define INDIVIDUAL_30MS_VAG 94
+
+#define TOTAL_10MS_VAG      1100 //1700
+#define TOTAL_20MS_VAG      1050
+#define TOTAL_30MS_VAG      1100
+
+#define OHMAX1_10MS_VAG     6  // Max Overhang 1
+#define OHMAX2_10MS_VAG     9  // Max Overhang 2
+#define OHMAX1_20MS_VAG     3
+#define OHMAX2_20MS_VAG     5
+
+#define OHMAX1_30MS_VAG     2
+#define OHMAX2_30MS_VAG     3
+
+#endif // WEBRTC_VAD_DEFINES_H_
diff --git a/trunk/src/common_audio/vad/vad_filterbank.c b/trunk/src/common_audio/vad/vad_filterbank.c
new file mode 100644
index 0000000..2f5db44
--- /dev/null
+++ b/trunk/src/common_audio/vad/vad_filterbank.c
@@ -0,0 +1,335 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "vad_filterbank.h"
+
+#include <assert.h>
+
+#include "signal_processing_library.h"
+#include "typedefs.h"
+#include "vad_defines.h"
+
+// Constants used in LogOfEnergy().
+static const int16_t kLogConst = 24660;  // 160*log10(2) in Q9.
+static const int16_t kLogEnergyIntPart = 14336;  // 14 in Q10
+
+// Coefficients used by HighPassFilter, Q14.
+static const int16_t kHpZeroCoefs[3] = { 6631, -13262, 6631 };
+static const int16_t kHpPoleCoefs[3] = { 16384, -7756, 5620 };
+
+// Allpass filter coefficients, upper and lower, in Q15.
+// Upper: 0.64, Lower: 0.17
+static const int16_t kAllPassCoefsQ15[2] = { 20972, 5571 };
+
+// Adjustment for division with two in SplitFilter.
+static const int16_t kOffsetVector[6] = { 368, 368, 272, 176, 176, 176 };
+
+// High pass filtering, with a cut-off frequency at 80 Hz, if the |data_in| is
+// sampled at 500 Hz.
+//
+// - data_in      [i]   : Input audio data sampled at 500 Hz.
+// - data_length  [i]   : Length of input and output data.
+// - filter_state [i/o] : State of the filter.
+// - data_out     [o]   : Output audio data in the frequency interval
+//                        80 - 250 Hz.
+static void HighPassFilter(const int16_t* data_in, int data_length,
+                           int16_t* filter_state, int16_t* data_out) {
+  int i;
+  const int16_t* in_ptr = data_in;
+  int16_t* out_ptr = data_out;
+  int32_t tmp32 = 0;
+
+
+  // The sum of the absolute values of the impulse response:
+  // The zero/pole-filter has a max amplification of a single sample of: 1.4546
+  // Impulse response: 0.4047 -0.6179 -0.0266  0.1993  0.1035  -0.0194
+  // The all-zero section has a max amplification of a single sample of: 1.6189
+  // Impulse response: 0.4047 -0.8094  0.4047  0       0        0
+  // The all-pole section has a max amplification of a single sample of: 1.9931
+  // Impulse response: 1.0000  0.4734 -0.1189 -0.2187 -0.0627   0.04532
+
+  for (i = 0; i < data_length; i++) {
+    // All-zero section (filter coefficients in Q14).
+    tmp32 = WEBRTC_SPL_MUL_16_16(kHpZeroCoefs[0], *in_ptr);
+    tmp32 += WEBRTC_SPL_MUL_16_16(kHpZeroCoefs[1], filter_state[0]);
+    tmp32 += WEBRTC_SPL_MUL_16_16(kHpZeroCoefs[2], filter_state[1]);
+    filter_state[1] = filter_state[0];