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

..

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