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wu@webrtc.org a1930427af When WEBRTC_VIDEO_EXTERNAL_CAPTURE_AND_RENDER is defined, we should never try to use _ptrCaptureDeviceInfo. 
Review URL: http://webrtc-codereview.appspot.com/167001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@631 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-20 17:38:57 +00:00
..
build refactor the gyp file to gypi file. 2011-09-12 12:24:39 +00:00
common_audio Add a unit testing framework. 2011-09-14 17:02:44 +00:00
common_video refactor the gyp file to gypi file. 2011-09-12 12:24:39 +00:00
modules Added support for new RTCP message REMB (remote estimated max bitrate) 2011-09-20 13:52:04 +00:00
system_wrappers Integrate the built-in WASAPI AEC DMO to VoE. 2011-09-13 17:17:49 +00:00
video_engine When WEBRTC_VIDEO_EXTERNAL_CAPTURE_AND_RENDER is defined, we should never try to use _ptrCaptureDeviceInfo. 2011-09-20 17:38:57 +00:00
voice_engine Enable OPENELSE defination when compile voice engine 2011-09-20 16:41:09 +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 Add missing intrinsic casts for VS 2005. 2011-09-19 18:48:25 +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.