webrtc

History

phoglund@webrtc.org 01530a2ac2 Rewrote the rcp_rtcp test. Finished rewriting the rtp_rtcp test. Rewrote first RTP RTCP test BUG= TEST= Review URL: http://webrtc-codereview.appspot.com/342007 git-svn-id: http://webrtc.googlecode.com/svn/trunk@1386 4adac7df-926f-26a2-2b94-8c16560cd09d		2012-01-11 12:26:34 +00:00
..
build	Enable warnings-as-errors on Mac.	2011-12-21 16:36:56 +00:00
common_audio	Refactor vad_filterbank: Some restructuring.	2012-01-10 13:48:09 +00:00
common_video	Correcting typo in libyuv.h.	2012-01-09 10:27:46 +00:00
modules	Implementation if mono-to-stereo and vice versa in ACM.	2012-01-11 10:12:54 +00:00
system_wrappers	Bugfix mac pid_t	2012-01-11 08:57:47 +00:00
video_engine	Added RTX to ViE.	2012-01-10 14:09:18 +00:00
voice_engine	Rewrote the rcp_rtcp test.	2012-01-11 12:26:34 +00:00
common_settings.gypi	git-svn-id: http://webrtc.googlecode.com/svn/trunk@156 4adac7df-926f-26a2-2b94-8c16560cd09d	2011-07-07 08:21:25 +00:00
common_types.h	Adding two new network metrics to NetEQ	2012-01-04 13:09:55 +00:00
engine_configurations.h	Forcing external transport to be on in Chrome.	2011-12-21 07:38:54 +00:00
LICENSE	Aligning license file with file header	2011-11-02 09:31:39 +00:00
LICENSE_THIRD_PARTY	Add references to src/ copies for LICENSE etc.	2011-10-26 01:05:07 +00:00
PATENTS	Adding copies of license files to src/ so that Chromium will get those as well.	2011-07-14 08:00:33 +00:00
README.chromium	git-svn-id: http://webrtc.googlecode.com/svn/trunk@156 4adac7df-926f-26a2-2b94-8c16560cd09d	2011-07-07 08:21:25 +00:00
typedefs.h	git-svn-id: http://webrtc.googlecode.com/svn/trunk@785 4adac7df-926f-26a2-2b94-8c16560cd09d	2011-10-20 12:30:35 +00:00

README.chromium

Name: WebRTC
URL: http://www.webrtc.org
Version: 90
License: BSD
License File: LICENSE

Description:
WebRTC provides real time voice and video processing
functionality to enable the implementation of 
PeerConnection/MediaStream.

Third party code used in this project is described 
in the file LICENSE_THIRD_PARTY.