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pbos@webrtc.org e9e4253a3c Sleep in ThreadTest thread functions. 
Prevents busy loops that really mess up Valgrind's thread scheduling,
this brings runtimes from up to minutes down to milliseconds.

BUG=
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6728 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-18 10:12:50 +00:00
..
base
Raw packet loss rate reported by RTP_RTCP module may vary too drastically over time. This CL is to add a filter to the value in VoE before lending it to audio coding module.
2014-07-16 21:28:26 +00:00
build
Move pacer to fully use webrtc::Clock instead of webrtc::TickTime.
2014-07-11 13:44:02 +00:00
common_audio
int16<->float conversions: Use size_t for array length argument, not int
2014-07-16 08:36:52 +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
AudioBuffer: Optimize const accesses to arrays that autoconvert int16<->float
2014-07-18 07:50:29 +00:00
overrides/webrtc/base
…
system_wrappers
Sleep in ThreadTest thread functions.
2014-07-18 10:12:50 +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
…
video
Make RTCP sender report send media bytes.
2014-07-15 15:25:39 +00:00
video_engine
Raw packet loss rate reported by RTP_RTCP module may vary too drastically over time. This CL is to add a filter to the value in VoE before lending it to audio coding module.
2014-07-16 21:28:26 +00:00
voice_engine
Raw packet loss rate reported by RTP_RTCP module may vary too drastically over time. This CL is to add a filter to the value in VoE before lending it to audio coding module.
2014-07-16 21:28:26 +00:00
.gitignore
…
BUILD.gn
GN: Fix include paths for WebRTC in Chromium build.
2014-07-13 09:02:54 +00:00
call.h
Remove GetDefaultConfigs() from Call.
2014-07-07 04:45:15 +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
…
common.h
…
config.cc
…
config.h
Move pacer to fully use webrtc::Clock instead of webrtc::TickTime.
2014-07-11 13:44:02 +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
…
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
GN: Add BUILD.gn files + kjellander to OWNERS
2014-06-23 19:21:07 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
supplement.gypi
TSan: Move suppressions to source file.
2014-06-27 09:18:51 +00:00
transport.h
…
typedefs.h
…
video_engine_tests.isolate
…
video_receive_stream.h
Remove GetDefaultConfigs() from Call.
2014-07-07 04:45:15 +00:00
video_renderer.h
…
video_send_stream.h
Move pacer to fully use webrtc::Clock instead of webrtc::TickTime.
2014-07-11 13:44:02 +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
…
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
…

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.