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Per 040f68e360 Fix VideoCapturer::OnMessage override 
Somehow this bug is now detected after landing https://codereview.webrtc.org/1689923002/
See https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Builder/builds/9638/steps/compile/logs/stdio

TBR=pthatcher@webrtc.org
BUG=webrtc:5426

Review URL: https://codereview.webrtc.org/1692923002 .

Cr-Commit-Position: refs/heads/master@{#11598}
2016-02-12 13:00:25 +00:00
..
api
This reland https://codereview.webrtc.org/1655793003/ with the change that cricket::VideoCapturer::SignalVideoFrame is added back and used for frame forwarding. It is used in Chrome remoting.
2016-02-12 12:31:08 +00:00
audio
Clean up of CongestionController.
2016-02-08 13:31:53 +00:00
base
rtc::Buffer: Replace an internal rtc::scoped_ptr with std::unique_ptr
2016-02-11 21:36:57 +00:00
build
Move talk/session/media -> webrtc/pc
2016-02-12 05:48:10 +00:00
call
Initialize VideoSendStream members in constructor.
2016-02-11 12:51:18 +00:00
common_audio
Move gtest_prod_util.h out of webrtc/test tree.
2016-02-05 08:23:57 +00:00
common_video
H264: Improve FFmpeg decoder performance by using I420BufferPool.
2016-02-05 16:08:39 +00:00
examples
Move talk/app/webrtc to webrtc/api
2016-02-10 09:53:26 +00:00
libjingle
Move talk/session/media -> webrtc/pc
2016-02-12 05:48:10 +00:00
media
Fix VideoCapturer::OnMessage override
2016-02-12 13:00:25 +00:00
modules
Fix negative shift exponent in WPDTree
2016-02-11 19:03:04 +00:00
p2p
Create QuicSession
2016-02-09 01:35:55 +00:00
pc
This reland https://codereview.webrtc.org/1655793003/ with the change that cricket::VideoCapturer::SignalVideoFrame is added back and used for frame forwarding. It is used in Chrome remoting.
2016-02-12 12:31:08 +00:00
sound
OWNERS: Add * to .gyp{i,} everywhere.
2015-12-16 19:44:39 +00:00
system_wrappers
Integrate helper macros for calling histograms with different names (real-time vs screenshare and rampup metrics).
2016-02-04 08:33:29 +00:00
test
Added A/V sync tests with drifting clocks.
2016-02-10 18:54:52 +00:00
tools
Move gtest_prod_util.h out of webrtc/test tree.
2016-02-05 08:23:57 +00:00
video
Use std::vector in the PayloadRouter interface.
2016-02-11 22:37:36 +00:00
voice_engine
Remove spammy GetRTPStatistics() log.
2016-02-12 10:35:18 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
audio_receive_stream.h
Add send-side BWE to WebRtcVoiceEngine under a finch experiment.
2016-02-04 12:12:31 +00:00
audio_send_stream.h
Prepare the AudioSendStream to be hooked up to send-side BWE.
2015-12-07 09:26:32 +00:00
audio_state.h
Add comments about the Audio parts of the public Call API being WIP.
2015-12-03 12:06:31 +00:00
BUILD.gn
Move gtest_prod_util.h out of webrtc/test tree.
2016-02-05 08:23:57 +00:00
call.h
Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats().
2015-11-06 23:34:58 +00:00
codereview.settings
Improve informative message in codereview.settings.
2015-11-10 10:08:36 +00:00
common_types.cc
Separating voice activity flag from audio level in RtpHeaderExtension.
2015-08-10 13:08:46 +00:00
common_types.h
Remove extra_options from VideoCodec.
2016-01-19 15:26:24 +00:00
common.gyp
Move gtest_prod_util.h out of webrtc/test tree.
2016-02-05 08:23:57 +00:00
common.h
Re-land: "Use an explicit identifier in Config"
2016-01-14 12:32:51 +00:00
config.cc
Support multiple rtx codecs.
2016-02-03 12:30:10 +00:00
config.h
Re-land: "Use an explicit identifier in Config"
2016-01-14 12:32:51 +00:00
engine_configurations.h
Remove VIDEOCODEC_* from engine_configurations.h.
2015-11-24 12:56:05 +00:00
frame_callback.h
Measure encoding time on encode callbacks.
2016-02-05 10:13:41 +00:00
libjingle_media_unittest.isolate
Move talk/media to webrtc/media
2016-02-05 07:52:35 +00:00
libjingle_p2p_unittest.isolate
Move talk/session/media -> webrtc/pc
2016-02-12 05:48:10 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Add the Ooura FFT to RealFourier.
2015-03-19 20:07:43 +00:00
OWNERS
…
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests.isolate
…
stream.h
Wire up PacketTime to ReceiveStreams.
2015-09-08 12:36:23 +00:00
supplement.gypi
Roll chromium_revision 68cf0b8..aa8e58a (361406:361601)
2015-11-25 21:20:11 +00:00
transport.h
Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
2015-12-09 11:13:40 +00:00
typedefs.h
Update attributes to match gclibc's ansidecl.h
2016-01-08 12:51:47 +00:00
video_decoder.h
Implement NullVideoDecoder to avoid crash on unsupported decoders.
2016-02-03 13:51:56 +00:00
video_encoder.h
Expose codec implementation names in stats.
2015-12-18 15:01:23 +00:00
video_engine_tests.isolate
Add audio streams to CallTest and a first A/V call test.
2016-01-07 16:43:31 +00:00
video_frame.h
Added EncodedImage::GetBufferPaddingBytes.
2016-01-21 13:43:18 +00:00
video_receive_stream.h
Expose codec implementation names in stats.
2015-12-18 15:01:23 +00:00
video_renderer.h
Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
2015-12-09 11:13:40 +00:00
video_send_stream.h
Measure encoding time on encode callbacks.
2016-02-05 10:13:41 +00:00
webrtc_examples.gyp
Move talk/session/media -> webrtc/pc
2016-02-12 05:48:10 +00:00
webrtc_nonparallel_tests.isolate
Create isolate files for nonparallel tests.
2015-10-29 14:29:15 +00:00
webrtc_perf_tests.isolate
Enable GoogleWifiTrace3Mbps simulations.
2015-05-04 12:26:26 +00:00
webrtc_tests.gypi
Move talk/session/media -> webrtc/pc
2016-02-12 05:48:10 +00:00
webrtc.gyp
Move talk/session/media -> webrtc/pc
2016-02-12 05:48:10 +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.