bjornv@webrtc.org 2f047ccede Removed unnecessary variable to avoid compiler error on Win. ...

Review URL: http://webrtc-codereview.appspot.com/267021

git-svn-id: http://webrtc.googlecode.com/svn/trunk@975 4adac7df-926f-26a2-2b94-8c16560cd09d

2011-11-18 12:03:25 +00:00

..

audio_coding

Remove use of exceptions in NetEQ test code

2011-11-18 09:55:01 +00:00

audio_conference_mixer

Add RMS computation for the RTP level indicator.

2011-11-15 16:57:56 +00:00

audio_device

This modification is supposed to fix the webrtc issue 144/145. With this fix, people could set/get mic volume before StartSend().

2011-11-18 02:17:28 +00:00

audio_processing

Removed unnecessary variable to avoid compiler error on Win.

2011-11-18 12:03:25 +00:00

interface

Added parsing of width and height from VP8 header

2011-11-02 23:14:58 +00:00

media_file

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

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

rtp_rtcp

Limit number of send-side BWE increases to one per second.

2011-11-16 07:58:31 +00:00

udp_transport

Removed code under a non-existing define.

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

utility

Restructuring and adding unit test targets on project level instead of in common_audio.

2011-11-17 08:35:47 +00:00

video_capture

Included modules in webrtc.gyp and fixed build errors.

2011-11-16 15:36:44 +00:00

video_coding

Fixing release compilation error

2011-11-17 14:26:21 +00:00

video_processing/main

Restructuring and adding unit test targets on project level instead of in common_audio.

2011-11-17 08:35:47 +00:00

video_render

Included modules in webrtc.gyp and fixed build errors.

2011-11-16 15:36:44 +00:00

modules.gyp

Restructuring and adding dummy unit test target.

2011-11-17 13:56:54 +00:00