webrtc/webrtc
tommi@webrtc.org 019955d770 Revert 8749 "We changed Encode() and EncodeInternal() return typ..."
The reason is that this cl adds a static initializer so we can't roll webrtc into Chromium.
See audio_encoder.cc and 'sizes' regression here:
http://build.chromium.org/p/chromium/builders/Linux%20x64/builds/186

> We changed Encode() and EncodeInternal() return type from bool to void in this issue:
> https://webrtc-codereview.appspot.com/38279004/
> Now we don't have to pass EncodedInfo as output parameter, but can return it instead. This also adds the benefit of making clear that EncodeInternal() needs to fill in this info.
> 
> R=kwiberg@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/43839004

TBR=jmarusic@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49449004

Cr-Commit-Position: refs/heads/master@{#8772}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8772 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-18 06:38:40 +00:00
..
base Revert 8753 "Use atomic operations for setting/reading the trace..." 2015-03-17 15:35:41 +00:00
build Adding build_opus as a switch in GYP. 2015-03-17 14:05:18 +00:00
common_audio Clean up LappedTransform and Blocker. 2015-03-12 23:24:19 +00:00
common_video Add I420 buffer pool to avoid unnecessary allocations 2015-03-17 11:41:15 +00:00
examples Roll chromium_revision 00e438c..8d51d96 (320241:320682) 2015-03-16 09:00:41 +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 Revert 8749 "We changed Encode() and EncodeInternal() return typ..." 2015-03-18 06:38:40 +00:00
overrides Re-landing perf improvement for libjingle logging after reverting the general change. 2015-03-07 12:18:14 +00:00
p2p Remove troublesome Windows line ending. 2015-03-17 21:50:29 +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 Fix FYI build - add a missing include to event_tracer.h in system_wrappers. 2015-03-17 22:15:28 +00:00
test Revert "Changed argument occurences of const I420VideoFrame* to const I420VideoFrame& and non-const I420VideoFrame& to I420VideoFrame*." 2015-03-16 13:48:18 +00:00
tools Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro 2015-03-04 13:04:54 +00:00
video Fix screenshare loopback target bitrate which isn't correctly configured 2015-03-17 16:28:11 +00:00
video_engine Change ThreadPosix to use an auto-reset event instead of manual reset now that we know the problem we had with EventWrapper::Wait was simply a bug in the EventWrapper. Also removing |started_| since we can just check the thread_ instead. 2015-03-16 16:06:16 +00:00
voice_engine Supporting Opus DTX in Voice Engine. 2015-03-13 09:38:55 +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 Parsing of transport wide sequence number rtp extension header. 2015-03-17 14:33:46 +00:00
common_types.h Parsing of transport wide sequence number rtp extension header. 2015-03-17 14:33:46 +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 Add CVO support to Vie layer. 2015-03-12 20:51:50 +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 Revert "Changed argument occurences of const I420VideoFrame* to const I420VideoFrame& and non-const I420VideoFrame& to I420VideoFrame*." 2015-03-16 13:48:18 +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 I420VideoFrame.CreateFrame: Removed unnecessary buffer size arguments. 2015-03-16 13:26:41 +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

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.