leozwang@webrtc.org 7fdb909339 Reformat and add more debug info into ViESurfaceRenderer ...

BUG=
TEST=test on android
Review URL: https://webrtc-codereview.appspot.com/546004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2163 4adac7df-926f-26a2-2b94-8c16560cd09d

2012-05-02 16:45:55 +00:00

..

audio_coding

Fix coverity issues in ACM.

2012-04-27 17:06:48 +00:00

audio_conference_mixer

Atomic32Wrapper -> Atomic32

2012-04-19 14:28:45 +00:00

audio_device

Minor changes to remove dead code in opensl es

2012-04-30 14:42:17 +00:00

audio_processing

Add tests for downmixing and no processing.

2012-05-02 00:04:10 +00:00

bitrate_controller

Fixing the header file path in gypi file.

2012-04-26 23:03:15 +00:00

interface

Fixes for coverity warnings.

2012-03-23 16:01:15 +00:00

media_file

Fix wrong data type in ReadWavHeader

2012-04-03 15:11:01 +00:00

rtp_rtcp

Timeout tests for TMMBR

2012-05-02 09:56:45 +00:00

udp_transport

Disabled UnremovedSocketsGetCollectedAtManagerDeletion in UdpSocketManager unittest.

2012-04-26 06:40:00 +00:00

utility

Split of stereo packets moved

2012-04-12 11:02:38 +00:00

video_capture

Hi Magnus, I added some of the changes that you suggested before. Let me know what you think.

2012-04-23 21:24:02 +00:00

video_coding

Wire up pading.

2012-04-23 14:52:15 +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

Reformat and add more debug info into ViESurfaceRenderer

2012-05-02 16:45:55 +00:00

modules.gyp

Break out of send side bandwidth estimation and controll.

2012-04-19 12:13:52 +00:00