webrtc

History

mflodman@webrtc.org 5eec6cf29a Started rewriting video_engine tests to use GUnit. - Added comments to the new test. - Added a new mode to the vie_auto_test binary. It is now possible to pass --automated to it to make it run noninteractively. - To be precise, it will run everything that has been rewritten as GUnit tests, which currently is one "test suite" in the binary. BUG= TEST= Review URL: http://webrtc-codereview.appspot.com/168002 Patch from Patrik Hoglund <phoglund@webrtc.org>. git-svn-id: http://webrtc.googlecode.com/svn/trunk@666 4adac7df-926f-26a2-2b94-8c16560cd09d		2011-09-29 12:24:13 +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	Bit-exact with non-Neon version.	2011-09-28 16:03:38 +00:00
system_wrappers	Added compare methods for TickInterval class.	2011-09-23 11:33:31 +00:00
video_engine	Started rewriting video_engine tests to use GUnit.	2011-09-29 12:24:13 +00:00
voice_engine	Remove assert "currentVoEMicLevel <= kMaxVolumeLevel". We ran into an issue on a Linux system where the currentVoEMicLevel was in fact greater than the kMaxVolumeLevel. Therefore we are removing this assert and capping the currentMicLevel to the maxVolumeLevel when this case is detected.	2011-09-28 17:45:03 +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.