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phoglund@webrtc.org 49d0d34ed5 Making sure neteq gets compiled with OPUS. 
All WebRTC calls on GN were failing because we failed to add OPUS as a
receive codec. The reason was that the WEBRTC_CODEC_OPUS define wasn't
set for the audio_decoder_impl.cc file; this CL fixes that.

BUG=None
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42309004

Cr-Commit-Position: refs/heads/master@{#8672}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8672 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-10 16:23:47 +00:00
..
base
Chromium build fix: Include new .cc files in rtc_base
2015-03-10 01:00:17 +00:00
build
Adds C++/JNI/Java unit test for audio device module on Android.
2015-03-09 12:40:43 +00:00
common_audio
Enabling common_audio building with NEON on ARM64
2015-03-05 19:14:21 +00:00
common_video
Revert "Make the entry point for VideoFrames to webrtc const ref I420VideoFrame."
2015-03-10 15:13:13 +00:00
examples
Adds C++/JNI/Java unit test for audio device module on Android.
2015-03-09 12:40:43 +00:00
libjingle
Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away
2015-02-26 14:43:50 +00:00
modules
Making sure neteq gets compiled with OPUS.
2015-03-10 16:23:47 +00:00
overrides
Re-landing perf improvement for libjingle logging after reverting the general change.
2015-03-07 12:18:14 +00:00
p2p
Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro
2015-03-04 13:04:54 +00:00
sound
Use std::min and std::max instead of self-defined functions such as rtc::_min/_max.
2015-02-12 11:55:32 +00:00
system_wrappers
Hide assembly symbols.
2015-03-09 23:21:42 +00:00
test
Revert "Make the entry point for VideoFrames to webrtc const ref I420VideoFrame."
2015-03-10 15:13:13 +00:00
tools
Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro
2015-03-04 13:04:54 +00:00
video
Revert "Make the entry point for VideoFrames to webrtc const ref I420VideoFrame."
2015-03-10 15:13:13 +00:00
video_engine
Revert "Make the entry point for VideoFrames to webrtc const ref I420VideoFrame."
2015-03-10 15:13:13 +00:00
voice_engine
Adds C++/JNI/Java unit test for audio device module on Android.
2015-03-09 12:40:43 +00:00
.gitignore
.gitignore: Add *.mk, created as part of ChromiumOS build
2013-01-04 21:25:42 +00:00
BUILD.gn
Let Chromium declare the mips_dsp_rev build variable.
2015-03-04 09:51:17 +00:00
call.h
Use int64_t more consistently for times, in particular for RTT values.
2015-01-12 21:51:21 +00:00
codereview.settings
Add codereview.settings to the /webrtc subdirectory
2014-12-05 13:43:35 +00:00
common_types.cc
Remove DCHECK from common_types.cc
2015-03-04 20:10:38 +00:00
common_types.h
Add CVO support to RTP sender side.
2015-03-04 22:55:43 +00:00
common.gyp
Fix style violations in common_types.h and config.h
2015-02-26 14:01:28 +00:00
common.h
Add a Config class interface to AudioProcessing for passing options.
2013-07-25 18:28:29 +00:00
config.cc
Fix style violations in common_types.h and config.h
2015-02-26 14:01:28 +00:00
config.h
Fix style violations in common_types.h and config.h
2015-02-26 14:01:28 +00:00
engine_configurations.h
Delete all codec-specific subclasses of ACMGenericCodec
2015-02-23 09:26:51 +00:00
experiments.h
Remove no longer used SkipEncodingUnusedStreams.
2014-07-22 07:17:17 +00:00
frame_callback.h
Changing include guard in frame_callback.h.
2015-02-03 14:51:39 +00:00
LICENSE
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
LICENSE_THIRD_PARTY
Remove webrtc/system_wrappers/interface/scoped_ptr.h
2015-03-03 10:17:49 +00:00
OWNERS
GN: Add BUILD.gn files + kjellander to OWNERS
2014-06-23 19:21:07 +00:00
PATENTS
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
PRESUBMIT.py
PRESUBMIT.py: accept variants on the copyright message that are present in the codebase.
2014-05-23 17:27:18 +00:00
README.chromium
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
rtc_unittests.isolate
Update all .isolate files for the new format.
2014-10-31 18:08:09 +00:00
supplement.gypi
Roll chromium_revision 601e6f3..b0c3ed3 (315263:316737)
2015-02-18 10:38:11 +00:00
transport.h
Rename newapi::Transport::SendRTP()->SendRtp().
2013-11-20 12:17:04 +00:00
typedefs.h
Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro
2015-03-04 13:04:54 +00:00
video_decoder.h
Add support for VP9 in webrtc::Call and video_loopback.
2014-11-04 19:41:15 +00:00
video_encoder.h
Remove default arguments in EncodedImageCallback.
2015-02-09 09:14:48 +00:00
video_engine_tests.isolate
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
video_frame.h
Revert "Make the entry point for VideoFrames to webrtc const ref I420VideoFrame."
2015-03-10 15:13:13 +00:00
video_receive_stream.h
Add decoder-timing stats to VideoReceiveStream.
2015-02-25 10:42:45 +00:00
video_renderer.h
Use VideoReceiveStream as an ExternalRenderer.
2015-02-09 15:15:24 +00:00
video_send_stream.h
Revert "Make the entry point for VideoFrames to webrtc const ref I420VideoFrame."
2015-03-10 15:13:13 +00:00
webrtc_examples.gyp
Improve cleaning for Android demo applications
2015-02-28 11:18:20 +00:00
webrtc_perf_tests.isolate
Offline screenshare quality test, plus loopback.
2015-02-18 12:46:44 +00:00
webrtc_tests.gypi
rtc_unittests on Android
2015-03-04 08:48:17 +00:00
webrtc.gyp
Rename GYP and GN targets for video capture+render.
2015-02-11 07:47:47 +00:00

README.chromium 

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.