mflodman@webrtc.org 1ce66e4dfb Don't report error when failing to send RTCP BYE. ...

Review URL: http://webrtc-codereview.appspot.com/337002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1293 4adac7df-926f-26a2-2b94-8c16560cd09d

2011-12-22 18:40:15 +00:00

..

audio_coding

Cleaned up leaky symbols in G722.

2011-12-22 12:20:06 +00:00

audio_conference_mixer

Removed usage of the deprecated critical section constructor in audio_conference_mixer.

2011-12-16 23:00:30 +00:00

audio_device

Disable API tests on ALSA since the tests don't work for all the alsa devices.

2011-12-09 14:05:29 +00:00

audio_processing

Cleaned up leaky symbols in NS.

2011-12-22 12:33:08 +00:00

interface

Review URL: http://webrtc-codereview.appspot.com/295010

2011-12-16 14:31:37 +00:00

media_file

Removed usage of the deprecated critical section constructor in media_file.

2011-12-14 17:27:58 +00:00

rtp_rtcp

Don't report error when failing to send RTCP BYE.

2011-12-22 18:40:15 +00:00

udp_transport

Removed unused function messing up the symbols.

2011-12-22 09:48:48 +00:00

utility

Removed usage of the deprecated critical section constructor in modules/utility.

2011-12-16 19:53:46 +00:00

video_capture

Remove warnings in VideoEngine, capture module and render module.

2011-12-09 10:12:57 +00:00

video_coding

Remove unused variable.

2011-12-22 08:34:31 +00:00

video_processing/main

Including Brighten function in namespace VideoProcessing

2011-12-20 15:33:49 +00:00

video_render

Fix last Mac/clang compile error.

2011-12-20 22:23:21 +00:00

modules.gyp

Restructuring and adding dummy unit test target.

2011-11-17 13:56:54 +00:00