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pbos@webrtc.org 27e5898f45 Explicitly unpoison FDs for MSan. 
MSan doesn't handle inline assembly that's used by FD_ZERO causing a
false positive.

R=earthdok@chromium.org, henrike@webrtc.org
BUG=chromium:344505

Review URL: https://webrtc-codereview.appspot.com/25799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7388 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-07 17:56:53 +00:00
..
base Explicitly unpoison FDs for MSan. 2014-10-07 17:56:53 +00:00
build Remove potential deadlock in WebRtcVideoEngine2. 2014-10-07 14:27:27 +00:00
common_audio Import LappedTransform and friends. 2014-10-01 17:42:18 +00:00
common_video GN: Add common configs to all targets. 2014-09-28 17:37:22 +00:00
examples Split video engine android initialization into each internal module initialization. 2014-09-17 11:44:51 +00:00
libjingle/xmllite Mac: adds missing _DEBUG flag to mac debug builds. 2014-10-06 22:04:11 +00:00
modules Roll chromium_revision c264a05..fc668e2 (297113:298195) 2014-10-07 12:49:34 +00:00
overrides webrtc/overrides: add OWNERS-file. 2014-09-17 08:04:28 +00:00
sound rtc_unittest: turned sound's test gyp into gypi to speed up GYP generation. 2014-10-01 16:33:03 +00:00
system_wrappers Import LappedTransform and friends. 2014-10-01 17:42:18 +00:00
test GN: Add common configs to tools and test. 2014-09-30 19:07:58 +00:00
tools Check on the existence of report directory 2014-10-06 17:21:27 +00:00
video Wire up CPU adaptation in WebRtcVideoEngine2. 2014-10-03 11:25:45 +00:00
video_engine GN: Add common configs to all targets. 2014-09-28 17:37:22 +00:00
voice_engine Reland "Remove DTMF status methods from Voice Engine" r7276 2014-10-01 08:23:21 +00:00
.gitignore .gitignore: Add *.mk, created as part of ChromiumOS build 2013-01-04 21:25:42 +00:00
BUILD.gn GN: Enable libvpx, add link target and convert some test targets 2014-09-30 18:05:02 +00:00
call.h Wire up CPU adaptation in WebRtcVideoEngine2. 2014-10-03 11:25:45 +00:00
common_types.h Move pacer to fully use webrtc::Clock instead of webrtc::TickTime. 2014-07-11 13:44:02 +00:00
common.gyp Add ToString() to VideoSendStream::Config. 2014-05-15 09:35:06 +00:00
common.h Add a Config class interface to AudioProcessing for passing options. 2013-07-25 18:28:29 +00:00
config.cc Add ToString() to VideoSendStream::Config. 2014-05-15 09:35:06 +00:00
config.h Config struct for VideoEncoder. 2014-09-19 12:30:25 +00:00
engine_configurations.h Add boilerplate code for H.264. 2014-07-04 12:42:07 +00:00
experiments.h Remove no longer used SkipEncodingUnusedStreams. 2014-07-22 07:17:17 +00:00
frame_callback.h Wire up statistics in video receive stream of new API 2014-02-07 12:06:29 +00:00
LICENSE Move src/ -> webrtc/ 2012-10-22 18:19:23 +00:00
LICENSE_THIRD_PARTY Consolidate all third party licenses in LICENSE_THIRD_PARTY. 2013-05-05 18:54:10 +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 Adds isolate for rtc_unittests and moves sound's unittests to rtc_unittest. 2014-09-30 14:21:10 +00:00
supplement.gypi Roll chromium_revision 289723:291647 2014-08-25 14:16:32 +00:00
transport.h Rename newapi::Transport::SendRTP()->SendRtp(). 2013-11-20 12:17:04 +00:00
typedefs.h Add CHECK and friends from Chromium. 2014-08-28 16:28:26 +00:00
video_encoder.h Expose VP8/H264 defaults through video_encoder.h. 2014-09-18 12:42:28 +00:00
video_engine_tests.isolate Pass GYP DEPTH variable to isolate. 2014-06-13 09:02:15 +00:00
video_frame.h Revert 7151 "Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h."" 2014-09-17 09:02:25 +00:00
video_receive_stream.h Change return value for number of discarded packets to be int. 2014-09-04 07:07:44 +00:00
video_renderer.h Separate Call API/build files from video_engine/. 2013-10-28 16:32:01 +00:00
video_send_stream.h Config struct for VideoEncoder. 2014-09-19 12:30:25 +00:00
webrtc_examples.gyp Roll chromium_revision 6455c69..deaf2f7 (293954:295079) 2014-09-27 18:10:30 +00:00
webrtc_perf_tests.isolate Pass GYP DEPTH variable to isolate. 2014-06-13 09:02:15 +00:00
webrtc_tests.gypi Moves xmllite's unittests to rtc_unittest. 2014-10-02 18:43:47 +00:00
webrtc.gyp Moves xmllite's unittests to rtc_unittest. 2014-10-02 18:43:47 +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.