henrik.lundin@webrtc.org 895870b68f Adding marker bit to RTPanalyze results ...

Review URL: http://webrtc-codereview.appspot.com/254005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@867 4adac7df-926f-26a2-2b94-8c16560cd09d

2011-11-02 08:44:42 +00:00

..

audio_coding

Adding marker bit to RTPanalyze results

2011-11-02 08:44:42 +00:00

audio_conference_mixer

Fixes crash due to r841.

2011-10-31 23:53:04 +00:00

audio_device

This CL is about to manually reset the ShutdownRenderEvent at StopPlayout().

2011-10-28 21:30:30 +00:00

audio_processing

Use __inline in NS-fixed.

2011-11-01 17:07:46 +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

Solves two bandwidth estimation issues and measures the sent video bitrate.

2011-10-27 16:08:29 +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

Only allow increasing capture time.

2011-11-01 17:10:49 +00:00

video_coding

Resolving a crash related to strncopy followed by a strcat

2011-11-01 15:19:44 +00:00

video_processing/main

updating vpm unit_test following r858

2011-11-01 22:07:16 +00:00

video_render

Changed Android makefile to make the lastest video render code run.

2011-10-31 21:29:50 +00:00

modules.gyp

Reorganize the audio_processing source.

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