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pbos@webrtc.org 124b9c70f9 Suppress races in event tracing code. 
Due to lack of atomics our tracing code is broken and triggering real
errors in ThreadSanitizer.

R=kjellander@webrtc.org
BUG=2497
TEST=out-tsan/out/Debug/libjingle_media_unittest --gtest_filter=WebRtcVideoMediaChannelTest.GetStatsMultipleRecvStreams + verifying that "race:*trace_event_unique_catstatic*" exists in the list of matched suppressions.

Review URL: https://webrtc-codereview.appspot.com/35719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8022 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-08 12:38:42 +00:00
..
base
Add a dummy implemenation of SChannelAdapter::SetMode that makes sure that StartSSL fails if the mode is set to DTLS.
2015-01-07 18:01:07 +00:00
build
Suppress races in event tracing code.
2015-01-08 12:38:42 +00:00
common_audio
Remove WebRtcSpl_ScaleAndAddVectorsWithRoundNeon
2015-01-07 21:34:23 +00:00
common_video
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
examples
Update Android projects to API level 21.
2014-10-31 23:26:10 +00:00
libjingle
Move the Jingle-specific network code into webrtc/libjingle.
2014-12-23 22:14:15 +00:00
modules
Reenable dlclose() for InternalUnloadDll on TSan.
2015-01-07 18:02:39 +00:00
overrides
Remove or rename typedefs with _t prefixes.
2014-12-17 13:43:55 +00:00
p2p
Disable flaky RelayServerTest.TestExpiration on all platforms.
2015-01-05 23:56:19 +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
Hack clock_unittest fix for parallel execution.
2015-01-05 19:08:58 +00:00
test
Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess.
2014-12-15 22:09:40 +00:00
tools
Merge in AGC manager and AGC tools.
2014-12-15 16:33:16 +00:00
video
Fix the ramp-up-down-up test which was using ts-offset extension with the abs-send-time estimator.
2014-12-19 18:00:21 +00:00
video_engine
Add histograms stats for sent/received fraction loss for a stream:
2015-01-08 07:50:56 +00:00
voice_engine
Rtp-Rtcp sender cleanup.
2014-12-19 13:49:55 +00:00
.gitignore
.gitignore: Add *.mk, created as part of ChromiumOS build
2013-01-04 21:25:42 +00:00
BUILD.gn
Enabling building with NEON on ARM64
2014-11-26 17:01:40 +00:00
call.h
Wire up RTT statistics to webrtc::Call.
2014-12-11 13:26:09 +00:00
codereview.settings
Add codereview.settings to the /webrtc subdirectory
2014-12-05 13:43:35 +00:00
common_types.h
Refactor some receive-side stats.
2014-12-18 13:50:16 +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
Log formatting fix for VideoEncoderConfig.
2014-11-06 09:35:08 +00:00
config.h
Refactor some receive-side stats.
2014-12-18 13:50:16 +00:00
engine_configurations.h
Add VP9 codec to VCM and vie_auto_test.
2014-11-01 06:10:48 +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
Update all .isolate files for the new format.
2014-10-31 18:08:09 +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
Annotate COMPILE_ASSERT with __attribute__((unused)).
2014-11-17 13:47:38 +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
Report encoded frame size in VideoSendStream.
2014-12-01 15:23:21 +00:00
video_engine_tests.isolate
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
video_frame.h
Report encoded frame size in VideoSendStream.
2014-12-01 15:23:21 +00:00
video_receive_stream.h
Wire up bandwidth stats to the new API and webrtcvideoengine2.
2014-11-05 14:05:29 +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
Enable payload-based padding by default and remove the API.
2014-12-19 15:33:17 +00:00
webrtc_examples.gyp
Add macros and APIs for webrtc histograms.
2014-10-23 12:57:56 +00:00
webrtc_perf_tests.isolate
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
webrtc_tests.gypi
Add AGC manager tests.
2014-12-16 14:48:47 +00:00
webrtc.gyp
Merge in AGC manager and AGC tools.
2014-12-15 16:33:16 +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.