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andrew@webrtc.org 587c844741 Query the capture volume immediately on Win Core. 
This allows the capture volume to be queried immediately at capture
startup, rather than waiting the usual one second interval.

Review URL: http://webrtc-codereview.appspot.com/297003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1064 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-30 17:43:05 +00:00
..
build Use clang by default on Mac. 2011-11-30 01:16:06 +00:00
common_audio Added support for 96 kHz sampling frequency. 2011-11-23 13:50:41 +00:00
common_video libyuv: Updating paths for test files 2011-11-29 17:50:07 +00:00
modules Query the capture volume immediately on Win Core. 2011-11-30 17:43:05 +00:00
system_wrappers Fixes broken unit test http://code.google.com/p/webrtc/issues/detail?id=154 2011-11-29 18:21:46 +00:00
video_engine Since the CocoaRenderView is forward declared with @class instead of imported, 2011-11-29 23:55:19 +00:00
voice_engine Removing deprecated NetEQ APIs 2011-11-30 16:21:22 +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 Removing statistics API from NetEQ 2011-11-23 09:36:23 +00:00
engine_configurations.h To be able to get webrtc into chrome, we need to reduce the size of the binary and the usage of memory. 2011-11-30 15:35:44 +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.