braveyao@webrtc.org 113f851cc3 Merge Chromium issue 95797 into WebRTC. ... Bug = 450 Test = Manual test Review URL: https://webrtc-codereview.appspot.com/551004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2192 4adac7df-926f-26a2-2b94-8c16560cd09d		2012-05-08 09:28:39 +00:00
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audio_coding	Hi Tina,	2012-05-07 20:36:22 +00:00
audio_conference_mixer	Rename AudioFrame members.	2012-05-02 23:56:37 +00:00
audio_device	Merge Chromium issue 95797 into WebRTC.	2012-05-08 09:28:39 +00:00
audio_processing	Add test to verify identical input channels result in identical output channels.	2012-05-05 00:32:00 +00:00
bitrate_controller	Fixed some memory leaks.	2012-05-03 11:32:25 +00:00
interface	Rename AudioFrame members.	2012-05-02 23:56:37 +00:00
media_file	Fix wrong data type in ReadWavHeader	2012-04-03 15:11:01 +00:00
rtp_rtcp	Renamed all _test.cc files to _unittest.cc, to conform to convention	2012-05-04 08:13:57 +00:00
udp_transport	Fixed some memory leaks.	2012-05-03 11:32:25 +00:00
utility	Rename AudioFrame members.	2012-05-02 23:56:37 +00:00
video_capture	Bug fix and refactor video capture code on android	2012-05-04 17:06:32 +00:00
video_coding	Revert VP8 Deblocker.	2012-05-08 09:06:31 +00:00
video_processing/main	VPM: fix to coverity issues 10255-10258 (unintended sign extension).	2012-04-27 15:56:02 +00:00
video_render	Two bug fixs in android surface render	2012-05-07 20:29:43 +00:00
modules.gyp	Break out of send side bandwidth estimation and controll.	2012-04-19 12:13:52 +00:00