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2d08d43206
webrtc/src
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amyfong@webrtc.org 2d08d43206 * Added modification of Start Bit Rate to vie_auto_test_custom_call 
* Added minor spacing and ":" for user input during vie_auto_test_custom_call
* Changed the default Video Port to 11111 and Audio Port to be 11113 to bring it inline with the WindowsTest application for ViE
Review URL: http://webrtc-codereview.appspot.com/181001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@654 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-27 17:46:45 +00:00
..
build refactor the gyp file to gypi file. 2011-09-12 12:24:39 +00:00
common_audio In spl, introduced function WebRtcSpl_Sat32To16(), and changed file resample_by_2.c, both for optimization in ARMv7. 2011-09-26 16:35:25 +00:00
common_video refactor the gyp file to gypi file. 2011-09-12 12:24:39 +00:00
modules video_coding: Updating media opt test - fixing call to protection callback. 2011-09-27 16:30:59 +00:00
system_wrappers Added compare methods for TickInterval class. 2011-09-23 11:33:31 +00:00
video_engine * Added modification of Start Bit Rate to vie_auto_test_custom_call 2011-09-27 17:46:45 +00:00
voice_engine Get the right guid str for GetRecordingDeviceName 2011-09-27 14:43:06 +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.