webrtc

History

stefan@webrtc.org 61b4abf1f8 Proper use of frame rate argument in generic_codec_test. BUG= TEST= Review URL: http://webrtc-codereview.appspot.com/181005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@686 4adac7df-926f-26a2-2b94-8c16560cd09d		2011-10-04 06:40:21 +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	Proper use of frame rate argument in generic_codec_test.	2011-10-04 06:40:21 +00:00
system_wrappers	Added compare methods for TickInterval class.	2011-09-23 11:33:31 +00:00
video_engine	Reverting changelist 666 since it broke the build on Mac.	2011-10-03 07:37:41 +00:00
voice_engine	Returning errors if any of the Init() settings in VoE fail.	2011-10-03 15:22:28 +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.