webrtc_m130

History

henrika@webrtc.org bb8ada686e TSan v2 reports data races in WebRTCAudioDeviceTest.FullDuplexAudioWithAGC

BUG=226044
TEST=content_unittests in Chrome with TSan v2 enabled

Review URL: https://webrtc-codereview.appspot.com/1201010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3760 4adac7df-926f-26a2-2b94-8c16560cd09d

2013-04-04 08:39:09 +00:00

build

Use ninja with merge_libs.py on Windows and clean up.

2013-02-26 02:47:50 +00:00

common_audio

Refactored inline assembly code in complex_fft.c, by combining the individual __asm lines into a single block, to avoid potential register usage problems when building with different tools.

2013-03-01 23:01:14 +00:00

common_video

Fixing/disabling Windows x64 warnings

2013-02-04 10:07:17 +00:00

modules

TSan v2 reports data races in WebRTCAudioDeviceTest.FullDuplexAudioWithAGC

2013-04-04 08:39:09 +00:00

system_wrappers

Fixes build break in previous cl (https://code.google.com/p/webrtc/source/detail?r=3739 ) found by Android bots.

2013-03-28 16:31:51 +00:00

test

Add some VoE and AudioProcessing mocks.

2013-03-25 21:20:38 +00:00

tools

Fix frame_editing_unittest reference file handling.

2013-03-11 15:30:33 +00:00

video_engine

Remove UDP transport API from ViE

2013-04-02 20:37:14 +00:00

voice_engine

TSan v2 reports data races in WebRTCAudioDeviceTest.FullDuplexAudioWithAGC

2013-04-04 08:39:09 +00:00

.gitignore

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

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

common_types.h

Removed all code enclosed in WEBRTC_SRTP #ifdefs, and the unsupported VoE SRTP APIs. Test stubs are left in place as we still have the (De)RegisterExternalEncryption() APIs, although they are currently untested.

2013-03-28 09:14:36 +00:00

engine_configurations.h

2013-03-28 09:14:36 +00:00

LICENSE

Move src/ -> webrtc/

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

LICENSE_THIRD_PARTY

Move src/ -> webrtc/

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

PATENTS

Move src/ -> webrtc/

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

README.chromium

Move src/ -> webrtc/

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

typedefs.h

Add some VoE and AudioProcessing mocks.

2013-03-25 21:20:38 +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.