asapersson@webrtc.org 43b8fc5c0d Review URL: http://webrtc-codereview.appspot.com/345011 ...

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1435 4adac7df-926f-26a2-2b94-8c16560cd09d

2012-01-16 13:49:04 +00:00

..

audio_coding

Fix issue 218 with new solution

2012-01-13 07:46:50 +00:00

audio_conference_mixer

Remove unused variable from mixer module.

2012-01-13 17:54:57 +00:00

audio_device

Review URL: http://webrtc-codereview.appspot.com/347012

2012-01-13 10:22:44 +00:00

audio_processing

Use -msse2 for SSE2 optimized code.

2012-01-13 19:43:09 +00:00

interface

Removed Version function from all modules.

2012-01-04 15:00:12 +00:00

media_file

Removed Version function from all modules.

2012-01-04 15:00:12 +00:00

rtp_rtcp

Review URL: http://webrtc-codereview.appspot.com/345011

2012-01-16 13:49:04 +00:00

udp_transport

Renaming 47 files from .cpp to .cc

2012-01-12 10:23:41 +00:00

utility

Renaming 47 files from .cpp to .cc

2012-01-12 10:23:41 +00:00

video_capture

Restoring unintentially renamed MS DirectShow source files in

2012-01-12 12:22:03 +00:00

video_coding

Ported more jitter buffer tests to unit tests.

2012-01-16 11:59:01 +00:00

video_processing/main

Use -msse2 for SSE2 optimized code.

2012-01-13 19:43:09 +00:00

video_render

Renaming 47 files from .cpp to .cc

2012-01-12 10:23:41 +00:00

modules.gyp

Added RTX to ViE.

2012-01-10 14:09:18 +00:00