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b7540b0322
webrtc/src
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mikhal@google.com b7540b0322 RTP: Changing the behavior in case of a send video packet error 
Review URL: http://webrtc-codereview.appspot.com/74005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@217 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-07-15 21:01:08 +00:00
..
build git-svn-id: http://webrtc.googlecode.com/svn/trunk@174 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-07-07 11:17:52 +00:00
common_audio Change android makefile to be able to build for x86 2011-07-14 18:23:07 +00:00
common_video U/V buffer fix for bilinear scale. 2011-07-13 00:07:40 +00:00
modules RTP: Changing the behavior in case of a send video packet error 2011-07-15 21:01:08 +00:00
system_wrappers Removed DISALLOW_* macros from the system_wrappers interface files. 2011-07-14 15:43:02 +00:00
video_engine Add include path to common_video/interface to android build 2011-07-15 17:25:27 +00:00
voice_engine Porting GTalk bugs: 2011-07-15 18:21:34 +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 git-svn-id: http://webrtc.googlecode.com/svn/trunk@156 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-07-07 08:21:25 +00:00
engine_configurations.h git-svn-id: http://webrtc.googlecode.com/svn/trunk@156 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-07-07 08:21:25 +00:00
LICENSE Adding copies of license files to src/ so that Chromium will get those as well. 2011-07-14 08:00:33 +00:00
LICENSE_THIRD_PARTY Adding copies of license files to src/ so that Chromium will get those as well. 2011-07-14 08:00:33 +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@156 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-07-07 08:21:25 +00:00
video_engine.gyp git-svn-id: http://webrtc.googlecode.com/svn/trunk@156 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-07-07 08:21:25 +00:00
voice_engine.gyp Ensures that all test files in VoE and ADM are read from 2011-07-07 14:10:34 +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.