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7ae9108b60
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andresp@webrtc.org 7ae9108b60 Remove more unused tsan suppressions and fix call test passing the same decoder to multiple received streams. 
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6651 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-10 10:35:12 +00:00
..
base Change Timing::WallTimeNow to be static. 2014-07-08 12:48:29 +00:00
build Remove more unused tsan suppressions and fix call test passing the same decoder to multiple received streams. 2014-07-10 10:35:12 +00:00
common_audio common_audio: Removes macro WEBRTC_SPL_SHIFT_W16 2014-07-03 13:38:53 +00:00
common_video GN: Implement BUILD.gn for common_video. 2014-07-03 17:04:12 +00:00
examples WebRTCDemo: set local SSRC for loopback test, otherwise receiver would reset it due to ssrc clash, which would cause delayed remote rendering. 2014-07-03 05:59:22 +00:00
modules Refactor registerable callbacks for FrameCountObserver from rtp_rtcp module into vie_channel. 2014-07-10 09:39:23 +00:00
overrides/webrtc/base Rename webrtc/base's IS_ALIGNED macro to RTC_IS_ALIGNED to avoid conflict between webrtc/base/basictypes.h and third_party/.../vpx_codec.h. 2014-05-21 16:52:14 +00:00
system_wrappers Clean data races from system_wrappers_unittests. 2014-07-07 20:37:39 +00:00
test Remove more unused tsan suppressions and fix call test passing the same decoder to multiple received streams. 2014-07-10 10:35:12 +00:00
tools Pass GYP DEPTH variable to isolate. 2014-06-13 09:02:15 +00:00
video Support VP8 encoder settings in VideoSendStream. 2014-07-10 10:13:37 +00:00
video_engine Refactor registerable callbacks for FrameCountObserver from rtp_rtcp module into vie_channel. 2014-07-10 09:39:23 +00:00
voice_engine Some refactoring inside rtp_rtcp/. 2014-07-08 12:10:51 +00:00
.gitignore .gitignore: Add *.mk, created as part of ChromiumOS build 2013-01-04 21:25:42 +00:00
BUILD.gn GN: Refactor base/BUILD.gn and fix dbus-glib error. 2014-06-28 18:05:22 +00:00
call.h Remove GetDefaultConfigs() from Call. 2014-07-07 04:45:15 +00:00
common_types.h Add boilerplate code for H.264. 2014-07-04 12:42:07 +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 Add ToString() to VideoSendStream::Config. 2014-05-15 09:35:06 +00:00
engine_configurations.h Add boilerplate code for H.264. 2014-07-04 12:42:07 +00:00
experiments.h Refactor ramp-up tests to have separate help files for the test classes, to make things more reusable. 2014-07-08 13:59:46 +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
supplement.gypi TSan: Move suppressions to source file. 2014-06-27 09:18:51 +00:00
transport.h Rename newapi::Transport::SendRTP()->SendRtp(). 2013-11-20 12:17:04 +00:00
typedefs.h Remove ALLOW_UNUSED. 2014-05-05 18:18:02 +00:00
video_engine_tests.isolate Pass GYP DEPTH variable to isolate. 2014-06-13 09:02:15 +00:00
video_receive_stream.h Remove GetDefaultConfigs() from Call. 2014-07-07 04:45:15 +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 pacing by default and remove the option to disable it from the new API. 2014-06-12 15:12:25 +00:00
webrtc_examples.gyp WebRTCDemo: set local SSRC for loopback test, otherwise receiver would reset it due to ssrc clash, which would cause delayed remote rendering. 2014-07-03 05:59:22 +00:00
webrtc_perf_tests.isolate Pass GYP DEPTH variable to isolate. 2014-06-13 09:02:15 +00:00
webrtc_tests.gypi Refactor ramp-up tests to have separate help files for the test classes, to make things more reusable. 2014-07-08 13:59:46 +00:00
webrtc.gyp Add ToString() to VideoSendStream::Config. 2014-05-15 09:35:06 +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.