webrtc

History

stefan@webrtc.org ddfdfed3b5 Pass capture time (wallclock) to the RTP sender to compute transmission offset - Change how the transmission offset is calculated, to incorporate the time since the frame was captured. - Break out RtpRtcpClock and move it to system_wrappers. - Use RtpRtcpClock to set the capture time in ms in the capture module. We must use the same clock as in the RTP module to be able to measure the time from capture until transmission. - Enables the RTP header extension for packet transmission time offsets. BUG= TEST=trybots Review URL: https://webrtc-codereview.appspot.com/666006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2489 4adac7df-926f-26a2-2b94-8c16560cd09d		2012-07-03 13:21:22 +00:00
..
build	Land: https://webrtc-codereview.appspot.com/678010/	2012-07-03 08:19:24 +00:00
common_audio	Move test to src/test.	2012-06-27 01:41:54 +00:00
common_video	Pass capture time (wallclock) to the RTP sender to compute transmission offset	2012-07-03 13:21:22 +00:00
modules	Pass capture time (wallclock) to the RTP sender to compute transmission offset	2012-07-03 13:21:22 +00:00
system_wrappers	Move test to src/test.	2012-06-27 01:41:54 +00:00
test	Wrote ClusterFuzz test for WebRTC GetUserMedia.	2012-07-02 11:39:22 +00:00
video_engine	Pass capture time (wallclock) to the RTP sender to compute transmission offset	2012-07-03 13:21:22 +00:00
voice_engine	Pass capture time (wallclock) to the RTP sender to compute transmission offset	2012-07-03 13:21:22 +00:00
common_types.h	Expose a set of options to the OveruseDetector supporting experiments	2012-06-26 10:47:04 +00:00
engine_configurations.h	Enable iSAC_FIX on android	2012-04-18 16:52:00 +00:00
LICENSE	Aligning license file with file header	2011-11-02 09:31:39 +00:00
LICENSE_THIRD_PARTY	Introduced ARM version of WebRtcSpl_SqrtFloor(). Function cycles reduced by ~ 30% in a real time VOE test in an android device (Nexus-S, ARMv7a).	2012-02-07 17:15:15 +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	[MIPS] Adding support for MIPS architecture for WebRTC.	2012-06-27 22:24:43 +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.