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aef0779dab
webrtc/webrtc
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tommi@webrtc.org aef0779dab Rewrite ThreadWindows. 
* Remove "dead" and "alive" variables.
* Remove critical section
* Skip synchronizing with the worker thread to verify startup (no need).
* Remove implementation of SetNotAlive()
* Always set thread name
* Add thread checks for correct usage.

Also added some TODOs for myself for the ThreadWrapper interface.

I'm removing the HasNoMonitorThread test since it is no longer relevant and ends up checking the wrong thing (ProcessThread - a generic thread type) in the wrong way (parsing a debug log) :)  I think it served a purpose some years ago, but things have changed since.

BUG=2902
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37069004

Cr-Commit-Position: refs/heads/master@{#8220}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8220 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-30 15:06:44 +00:00
..
base Control the max IPv6 Networks used by WebRTC. 2015-01-30 00:09:42 +00:00
build Move use of DEPTH into build_with_chromium==1. 2015-01-30 14:55:20 +00:00
common_audio Move channel_buffer.{h,cc} to common_audio. 2015-01-28 19:57:44 +00:00
common_video Make it easier to use external libyuv + cleanup GYP files. 2015-01-26 19:17:26 +00:00
examples Add arraysize() macro from Chromium, and make use of it in a few places. 2015-01-28 21:37:13 +00:00
libjingle Enable Clang warning implicit-fallthrough and annotate the code. 2015-01-28 18:38:13 +00:00
modules Refactor senders into senders and sources in the simulation framework. 2015-01-30 14:37:09 +00:00
overrides Remove COMPILE_ASSERT and use static_assert everywhere 2015-01-14 10:51:54 +00:00
p2p Adding constness. 2015-01-28 17:33:45 +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 Rewrite ThreadWindows. 2015-01-30 15:06:44 +00:00
test Remove ChangeUniqueID. 2015-01-29 12:14:13 +00:00
tools Re-land "Remove <(webrtc_root) from source file entries." 2015-01-29 14:30:41 +00:00
video Add tracing for slow paths in new video API. 2015-01-29 12:33:42 +00:00
video_engine Revert 8203 "Reducing locking in OveruseFrameDetector and increa..." 2015-01-29 16:09:07 +00:00
voice_engine Fixing a VoE test to set correct rate for iSAC 2015-01-30 13:04:47 +00:00
.gitignore .gitignore: Add *.mk, created as part of ChromiumOS build 2013-01-04 21:25:42 +00:00
BUILD.gn Make it easier to use external libyuv + cleanup GYP files. 2015-01-26 19:17:26 +00:00
call.h Use int64_t more consistently for times, in particular for RTT values. 2015-01-12 21:51:21 +00:00
codereview.settings Add codereview.settings to the /webrtc subdirectory 2014-12-05 13:43:35 +00:00
common_types.h Update StreamDataCounter with FEC bytes. 2015-01-27 12:17:29 +00:00
common.gyp Make it easier to use external libyuv + cleanup GYP files. 2015-01-26 19:17:26 +00:00
common.h Add a Config class interface to AudioProcessing for passing options. 2013-07-25 18:28:29 +00:00
config.cc Revert 8028 "Support associated payload type when registering Rt..." 2015-01-09 20:22:46 +00:00
config.h Revert 8028 "Support associated payload type when registering Rt..." 2015-01-09 20:22:46 +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 Enable Clang warning implicit-fallthrough and annotate the code. 2015-01-28 18:38:13 +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 Use int64_t more consistently for times, in particular for RTT values. 2015-01-12 21:51:21 +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 Log configs when creating video streams in Call. 2015-01-15 10:09:39 +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 Log configs when creating video streams in Call. 2015-01-15 10:09:39 +00:00
webrtc_examples.gyp Move internal capture+render to build_with_chromium==0 condition 2015-01-20 11:40:45 +00:00
webrtc_perf_tests.isolate Update isolate files for Android APK tests. 2014-11-13 08:35:05 +00:00
webrtc_tests.gypi Re-land "Remove <(webrtc_root) from source file entries." 2015-01-29 14:30:41 +00:00
webrtc.gyp Make it easier to use external libyuv + cleanup GYP files. 2015-01-26 19:17:26 +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.