webrtc

History

phoglund@webrtc.org d056abd62f Prepared tests for running on build bot.

The changes will mostly clarify why PSNR and SSIM thresholds are chosen. I think those explanations will be good since these tests may well be a bit flaky.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/343017

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1496 4adac7df-926f-26a2-2b94-8c16560cd09d

2012-01-20 12:48:13 +00:00

build

Add target and config info to merged lib name.

2012-01-20 01:32:16 +00:00

common_audio

Incorrect pointer inputs in filter test.

2012-01-19 10:41:39 +00:00

common_video

Correcting typo in libyuv.h.

2012-01-09 10:27:46 +00:00

modules

Fixing Valgrind problem detected by video_processing_unittests.

2012-01-20 08:40:55 +00:00

system_wrappers

CpuLinux file handle leak.

2012-01-16 09:31:39 +00:00

video_engine

Prepared tests for running on build bot.

2012-01-20 12:48:13 +00:00

voice_engine

Remove the diamond inheritance pattern from VoEVideoSyncImpl in attempt to see if this fixes coverity reports.

2012-01-19 15:05:36 +00:00

common_settings.gypi

git-svn-id: http://webrtc.googlecode.com/svn/trunk@156 4adac7df-926f-26a2-2b94-8c16560cd09d

2011-07-07 08:21:25 +00:00

common_types.h

Removing unused code.

2012-01-17 12:45:47 +00:00

engine_configurations.h

Forcing external transport to be on in Chrome.

2011-12-21 07:38:54 +00:00

LICENSE

Aligning license file with file header

2011-11-02 09:31:39 +00:00

LICENSE_THIRD_PARTY

Add references to src/ copies for LICENSE etc.

2011-10-26 01:05:07 +00:00

PATENTS

Adding copies of license files to src/ so that Chromium will get those as well.

2011-07-14 08:00:33 +00:00

README.chromium

git-svn-id: http://webrtc.googlecode.com/svn/trunk@156 4adac7df-926f-26a2-2b94-8c16560cd09d

2011-07-07 08:21:25 +00:00

typedefs.h

git-svn-id: http://webrtc.googlecode.com/svn/trunk@785 4adac7df-926f-26a2-2b94-8c16560cd09d

2011-10-20 12:30:35 +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.