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aleloi 1655e45d85 Elimiteted race condition in the AudioMixer. 
The mixer allocates an audio frame for each added data source. This
audio frame was deallocated when a source was removed from the
mixer. Source removal could happen during the mixing, and the existing
locking scheme (and the Clang thread checker) was not sufficient to
prevent a data race.

After this change, the mixer doesn't release its lock until it is
finished with the sources' Audio frames. Since multi-threaded access to
the mixer only happens when a source is added or removed, we believe
that this change wouldn't have any noticeable performance impact.

NOTRY=True

BUG=webrtc:6346

Review-Url: https://codereview.webrtc.org/2439283002
Cr-Commit-Position: refs/heads/master@{#14744}
2016-10-24 13:57:03 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
RTCTransportStats[1] added, supporting all members.
2016-10-24 11:00:12 +00:00
audio
Add a placeholder stat for logging the estimated residual echo likelihood.
2016-10-21 11:10:08 +00:00
base
Add tests and fix thread annotations
2016-10-24 13:07:33 +00:00
build
Delete DataLog abstraction, which was almost unused.
2016-10-21 08:53:01 +00:00
call
Integrate FlexfecReceiveStream with Call.
2016-10-24 06:37:24 +00:00
common_audio
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds.
2016-10-17 06:56:20 +00:00
common_video
Reland of Delete webrtc::VideoFrame::CopyFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2397943003/ )
2016-10-24 07:44:17 +00:00
examples
Fix org.mockito.Matchers deprecation warnings in DirectRTCClientTest.
2016-10-20 12:49:12 +00:00
libjingle
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds.
2016-10-17 06:56:20 +00:00
logging
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds.
2016-10-17 06:56:20 +00:00
media
Implement framesEncoded stat in video send ssrc stats.
2016-10-24 09:09:39 +00:00
modules
Elimiteted race condition in the AudioMixer.
2016-10-24 13:57:03 +00:00
p2p
Fix some flaky tests by using longer timeout and/or fake clock.
2016-10-20 18:47:19 +00:00
pc
Remove cricket::VideoCodec with, height and framerate properties
2016-10-24 08:21:24 +00:00
sdk
iOS: Optimize video scaling and cropping
2016-10-20 10:34:32 +00:00
stats
RTCTransportStats[1] added, supporting all members.
2016-10-24 11:00:12 +00:00
system_wrappers
Delete DataLog abstraction, which was almost unused.
2016-10-21 08:53:01 +00:00
test
GN: move webrtc/video/ targets from webrtc/BUILD.gn into webrtc/video/BUILD.gn
2016-10-24 09:09:40 +00:00
tools
During a fix of an unrelated issue, a bug was introduced in the rtp analyzer tool: when the number of data points was divisible by RTPStatitstics.PLOT_RESOLUTION_MS (which is 50), pyplot.plot was called with arrays of different lengths. One of the arrays could be one element larger.
2016-10-18 08:44:50 +00:00
video
GN: move webrtc/video/ targets from webrtc/BUILD.gn into webrtc/video/BUILD.gn
2016-10-24 09:09:40 +00:00
voice_engine
Move audio frame memory handling inside AudioMixer.
2016-10-20 21:24:46 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
BUILD.gn
Add tests and fix thread annotations
2016-10-24 13:07:33 +00:00
call.h
Integrate FlexfecReceiveStream with Call.
2016-10-24 06:37:24 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
common_types.cc
Reland of Ignore Camera and Flip bits in CVO when parsing video rotation (patchset #1 id:1 of https://codereview.webrtc.org/2300323002/ )
2016-09-08 10:25:05 +00:00
common_types.h
Enable the -Wundef warning for clang
2016-09-29 00:42:08 +00:00
common.gyp
Delete old video defines in engine config.
2016-10-07 05:07:36 +00:00
config.cc
Add FlexfecReceiveStream.
2016-10-20 11:54:51 +00:00
config.h
Add FlexfecReceiveStream.
2016-10-20 11:54:51 +00:00
DEPS
Delete old video defines in engine config.
2016-10-07 05:07:36 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete talk directory, and references from build_ios_libs.sh.
2016-09-02 10:07:13 +00:00
OWNERS
OWNERS: Make everyone able to change *.gn,*.gni files.
2016-09-09 12:51:48 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
supplement.gypi
Cleanup MIPS specific link configuration
2016-06-20 05:39:25 +00:00
transport.h
Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
2015-12-09 11:13:40 +00:00
typedefs.h
[MIPS] Fix build issue for mips64el
2016-06-23 10:52:35 +00:00
video_decoder.h
Replace scoped_ptr with unique_ptr in webrtc/common_video/
2016-03-02 11:41:40 +00:00
video_encoder.h
Make MediaCodecEncoder fallback to a software encoder on failure.
2016-10-06 12:55:19 +00:00
video_frame.h
Reland of Delete webrtc::VideoFrame::CopyFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2397943003/ )
2016-10-24 07:44:17 +00:00
video_receive_stream.h
Rename FecConfig to UlpfecConfig in config.h.
2016-10-05 06:28:43 +00:00
video_send_stream.h
Implement framesEncoded stat in video send ssrc stats.
2016-10-24 09:09:39 +00:00
voice_engine_configurations.h
Delete old video defines in engine config.
2016-10-07 05:07:36 +00:00
webrtc_examples.gyp
Reland of name AppRTCDemo on Android and iOS to AppRTCMobile (patchset #1 id:1 of https://codereview.webrtc.org/2358133003/ )
2016-09-27 08:10:52 +00:00
webrtc_tests.gypi
GYP: Remove targets inside include_tests==1 that are converted to GN.
2016-09-15 11:57:39 +00:00
webrtc.gyp
Moved RtcEventLog files from call/ to logging/
2016-10-04 01:31:32 +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.