braveyao@webrtc.org 59d6cec291 Fix the crash at playing 48kHz stereo wav file. ... http://code.google.com/p/webrtc/issues/detail?id=208 Review URL: https://webrtc-codereview.appspot.com/396001 git-svn-id: http://webrtc.googlecode.com/svn/trunk@1681 4adac7df-926f-26a2-2b94-8c16560cd09d		2012-02-14 18:17:16 +00:00
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audio_coding	Adding missing timestamp calculation to RTPencode.	2012-02-08 13:20:36 +00:00
audio_conference_mixer	Fix enumeral comparison error.	2012-01-25 19:38:16 +00:00
audio_device	This CL makes the playout delay value thread safe.	2012-02-09 13:50:48 +00:00
audio_processing	Restore AECM Coverity fix.	2012-02-09 17:16:18 +00:00
interface	AudioFrame payload shouldn't be mutable.	2012-02-03 17:47:32 +00:00
media_file	Fix the crash at playing 48kHz stereo wav file.	2012-02-14 18:17:16 +00:00
rtp_rtcp	Update receive only channels with RTT.	2012-02-14 12:49:59 +00:00
udp_transport	Bugfix EnableIPV6 issue 255	2012-02-08 08:08:37 +00:00
utility	Removing encoder reset. Function did not make sence.	2012-02-13 09:03:53 +00:00
video_capture	Enable all modules without building errors	2012-01-19 22:11:37 +00:00
video_coding	Allow for spatial-downsampling without reinitializaing encoder. Change of frame size will automatically trigger new key frame in codec. This feature is set off in vie_encoder until we upgrade to the new libvpx.	2012-02-14 17:26:24 +00:00
video_processing/main	Disable spatial subsampling for denoiser variance estimation.	2012-02-08 08:27:31 +00:00
video_render	Coverity fix for VideoRenderModule including issues 10084, 10226, 10267 and 10340.	2012-02-08 00:14:25 +00:00
modules.gyp	Added RTX to ViE.	2012-01-10 14:09:18 +00:00