stefan@webrtc.org 5eb64f06be Fix BitrateSent() API when having a default RTP module. ...

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/242004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@796 4adac7df-926f-26a2-2b94-8c16560cd09d

2011-10-21 13:42:50 +00:00

..

audio_coding

Adding neteq_tests to modules tests

2011-10-05 20:38:19 +00:00

audio_conference_mixer

refactor the gyp file to gypi file.

2011-09-12 12:24:39 +00:00

audio_device

[Issue 101] Solves memory leak on Windows

2011-10-20 12:11:45 +00:00

audio_processing

Updated the AEC delay logging to output values in ms. PB output updated.

2011-10-19 08:47:40 +00:00

interface

First implementation of simulcast, adds VP8 simulcast to video engine.

2011-10-13 15:19:55 +00:00

media_file

Review URL: http://webrtc-codereview.appspot.com/243001

2011-10-19 18:30:25 +00:00

rtp_rtcp

Fix BitrateSent() API when having a default RTP module.

2011-10-21 13:42:50 +00:00

udp_transport

Removed code under a non-existing define.

2011-10-06 18:14:25 +00:00

utility

First implementation of simulcast, adds VP8 simulcast to video engine.

2011-10-13 15:19:55 +00:00

video_capture

* Add include_internal_video_capture and include_internal_video_render to include/exclude the internal VCM and VRM.

2011-10-19 18:42:17 +00:00

video_coding

Fixes a rate control bug in the VP8 wrapper.

2011-10-21 13:15:16 +00:00

video_processing/main

Fix MSVC issues in AEC to enable SSE2 optimization on Windows.

2011-09-16 22:51:57 +00:00

video_render

* Add include_internal_video_capture and include_internal_video_render to include/exclude the internal VCM and VRM.

2011-10-19 18:42:17 +00:00

modules.gyp

Reorganize the audio_processing source.

2011-10-19 01:40:33 +00:00