webrtc_m130

History

wu@webrtc.org 6049787252 Add protection to few more methods of AudioDeviceLinuxALSA. Those methods can be called from

a different thread.

One example is the _playWarning can be changed in AudioDeviceLinuxALSA::Init, which is called on the application's thread. At the same time it can be read via PlayoutWarning() on the VoE's process_thread.

RISK=P2
TESTED=try bots and tsan test:
tools/valgrind-webrtc/webrtc_tests.sh --tool=tsan -t out/Debug/libjingle_peerconnection_unittest --gtest_filter=PeerConnectionFactoryTestInternal.CreatePCUsingInternalModules
BUG=1205
R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2315004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4866 4adac7df-926f-26a2-2b94-8c16560cd09d

2013-09-27 18:19:25 +00:00

build

Recognize armv7 target_arch for ios support in webrtc common.gyp

2013-09-05 16:46:36 +00:00

common_audio

MIPS optimizations for the functions WebRtcSpl_SqrtFloor, WebRtcSpl_CrossCorrelation, WebRtcSpl_ScaleAndAddVectorsWithRound and the inline functions from signal_processing spl_inl.h file.

2013-09-18 17:40:46 +00:00

common_video

Refactor frame generation code so it can be used by multiple modules.

2013-09-19 12:14:03 +00:00

modules

Add protection to few more methods of AudioDeviceLinuxALSA. Those methods can be called from

2013-09-27 18:19:25 +00:00

system_wrappers

Disable tests for TSan v2

2013-09-19 07:15:36 +00:00

test

Remove include_dirs from test.

2013-09-26 14:03:57 +00:00

tools

Remove include_dirs from tools.

2013-09-26 14:04:27 +00:00

video_engine

Remove test parameters from CallTest.

2013-09-27 10:54:10 +00:00

voice_engine

Fix for Heap-use-after-free in webrtc::voe::Channel::SendRTCPPacket

2013-09-24 15:42:49 +00:00

.gitignore

.gitignore: Add *.mk, created as part of ChromiumOS build

2013-01-04 21:25:42 +00:00

common_types.h

Remove redundant STR_CASE_CMP macro definitions.

2013-09-09 17:50:10 +00:00

common_unittest.cc

Fix unnamed-type-template-args warnings on clang.

2013-06-27 10:18:09 +00:00

common.h

Add a Config class interface to AudioProcessing for passing options.

2013-07-25 18:28:29 +00:00

engine_configurations.h

Adds a new voice engine warning for the typing noise off state.

2013-09-17 18:09:20 +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

PATENTS

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

PRESUBMIT.py

Modified the presubmit checks such that difference license templates are checked for in webrtc and talk folder.

2013-07-22 22:32:50 +00:00

README.chromium

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

typedefs.h

Disable clang C++11 warnings to permit OVERRIDE keyword.

2013-05-08 08:34:34 +00:00

webrtc.gyp

Makes webrtc and libjingle build from the same gyp-file. Also, the libjingle and webrtc DEPS revisions were mismatching. This cl takes the most recent revision of mismatches. Also disables 64 bit Mac builds for libjingle

2013-07-23 18:15:11 +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.