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Peter Boström 02bafc6379 Add a race-checking mechanism. 
Permits CHECKing/DCHECKing that methods are being accessed in a
thread-safe manner, even if they are not used by one single thread
(thread pools such as VideoToolbox OK).

BUG=
R=danilchap@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/2097403002 .

Cr-Commit-Position: refs/heads/master@{#13358}
2016-07-01 10:45:29 +00:00
..
androidjunit
Reland of Android: Change camera fps range selection (patchset #1 id:1 of https://codereview.webrtc.org/2021233002/ )
2016-06-01 07:44:07 +00:00
api
Revert of Combine webrtc/api/java/android and webrtc/api/java/src. (patchset #1 id:1 of https://codereview.webrtc.org/2111823002/ )
2016-07-01 07:37:49 +00:00
audio
Revert of Move RtcEventLog object from inside VoiceEngine to Call. (patchset #16 id:420001 of https://codereview.webrtc.org/1748403002/ )
2016-06-30 07:59:49 +00:00
base
Add a race-checking mechanism.
2016-07-01 10:45:29 +00:00
build
Remove dead GYP target audio_device_module_java
2016-06-20 08:43:42 +00:00
call
Revert of Move RtcEventLog object from inside VoiceEngine to Call. (patchset #16 id:420001 of https://codereview.webrtc.org/1748403002/ )
2016-06-30 07:59:49 +00:00
common_audio
Keep track of the user-facing number of channels in a ChannelBuffer
2016-06-30 22:33:47 +00:00
common_video
Add a race-checking mechanism.
2016-07-01 10:45:29 +00:00
examples
Dispose audio source correctly in AppRTC Demo on Android.
2016-06-30 14:57:22 +00:00
libjingle
GN: Update xmpp and p2p to match Chromium build
2016-06-18 11:23:08 +00:00
media
Revert of Workaround for clang bug http://llvm.org/PR28348. (patchset #1 id:1 of https://codereview.webrtc.org/2110043003/ )
2016-06-30 09:32:37 +00:00
modules
Add a race-checking mechanism.
2016-07-01 10:45:29 +00:00
p2p
Add config to prune low-priority TURN ports for creating connections
2016-07-01 03:52:16 +00:00
pc
Revert of Move RtcEventLog object from inside VoiceEngine to Call. (patchset #16 id:420001 of https://codereview.webrtc.org/1748403002/ )
2016-06-30 07:59:49 +00:00
sdk
Revert of Fix for RtcEventLog ObjC interface (patchset #3 id:40001 of https://codereview.webrtc.org/2106103003/ )
2016-06-30 07:55:07 +00:00
system_wrappers
FileWrapper[Impl] modifications and actually remove the "Impl" class.
2016-06-15 17:30:18 +00:00
test
Revert of Move RtcEventLog object from inside VoiceEngine to Call. (patchset #16 id:420001 of https://codereview.webrtc.org/1748403002/ )
2016-06-30 07:59:49 +00:00
tools
New rtc dump analyzing tool in Python
2016-06-20 14:39:21 +00:00
video
Expanded error message for unexpected packet
2016-06-29 13:29:48 +00:00
voice_engine
Revert of Move RtcEventLog object from inside VoiceEngine to Call. (patchset #16 id:420001 of https://codereview.webrtc.org/1748403002/ )
2016-06-30 07:59:49 +00:00
.gitignore
…
audio_receive_stream.h
Add AudioReceiveStream::SetGain() method and use that in WVoMC::SetOutputVolume().
2016-06-17 15:30:58 +00:00
audio_send_stream.h
Add AudioSendStream::SetMuted() method and use it in WVoMC::MuteStream().
2016-06-16 17:53:28 +00:00
audio_sink.h
…
audio_state.h
…
BUILD.gn
Build webrtc_nonparallel_tests under GN.
2016-06-27 17:31:56 +00:00
call.h
Revert of Move RtcEventLog object from inside VoiceEngine to Call. (patchset #16 id:420001 of https://codereview.webrtc.org/1748403002/ )
2016-06-30 07:59:49 +00:00
codereview.settings
…
common_types.cc
FileWrapper[Impl] modifications and actually remove the "Impl" class.
2016-06-15 17:30:18 +00:00
common_types.h
Avoid unnecessary HW video encoder reconfiguration
2016-06-16 19:08:11 +00:00
common.gyp
Revert of Add missing headers and fix some missing dependencies (patchset #1 id:20001 of https://codereview.webrtc.org/1990593002/ )
2016-05-18 12:00:55 +00:00
common.h
New module for the adaptive level controlling functionality in the audio processing module
2016-06-29 22:26:19 +00:00
config.cc
Avoid unnecessary HW video encoder reconfiguration
2016-06-16 19:08:11 +00:00
config.h
Avoid unnecessary HW video encoder reconfiguration
2016-06-16 19:08:11 +00:00
DEPS
Moved creation of AudioDecoderFactory to inside PeerConnectionFactory.
2016-06-13 14:35:01 +00:00
engine_configurations.h
…
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
Add kwiberg@webrtc.org as root owner.
2016-06-14 09:39:40 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests_apk.isolate
…
rtc_unittests.isolate
…
supplement.gypi
Cleanup MIPS specific link configuration
2016-06-20 05:39:25 +00:00
transport.h
…
typedefs.h
[MIPS] Fix build issue for mips64el
2016-06-23 10:52:35 +00:00
video_decoder.h
…
video_encoder.h
…
video_engine_tests_apk.isolate
…
video_engine_tests.isolate
…
video_frame.h
Add TODO comments on deprecated VideoFrame methods.
2016-06-22 15:47:52 +00:00
video_receive_stream.h
Movable support for VideoReceiveStream::Config and avoid copies.
2016-06-10 15:58:12 +00:00
video_send_stream.h
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
webrtc_examples.gyp
Add GN targets for AppRTC Demo on Android.
2016-06-29 12:55:12 +00:00
webrtc_nonparallel_tests_apk.isolate
…
webrtc_nonparallel_tests.isolate
…
webrtc_perf_tests_apk.isolate
…
webrtc_perf_tests.isolate
Fix missing resource file in webrtc_perf_tests.isolate
2016-06-17 07:39:49 +00:00
webrtc_tests.gypi
New module for the adaptive level controlling functionality in the audio processing module
2016-06-29 22:26:19 +00:00
webrtc.gyp
Remove webrtc_all target
2016-06-14 09:09:24 +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.