4 Commits

Author SHA1 Message Date
stefan@webrtc.org
a678a3baee Move video_coding to new Clock interface and remove fake clock implementations from RTP module tests.
TEST=video_coding_unittests, video_coding_integrationtests, rtp_rtcp_unittests, trybots

Review URL: https://webrtc-codereview.appspot.com/1044004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3393 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-21 07:42:11 +00:00
stefan@webrtc.org
20ed36dada Break out RtpClock to system_wrappers and make it more generic.
The goal with this new clock interface is to have something which is used all
over WebRTC to make it easier to switch clock implementation depending on where
the components are used. This is a first step in that direction.

Next steps will be to, step by step, move all modules, video engine and voice
engine over to the new interface, effectively deprecating the old clock
interfaces. Long-term my vision is that we should be able to deprecate the clock
of WebRTC and rely on the user providing the implementation.

TEST=vie_auto_test, rtp_rtcp_unittests, trybots

Review URL: https://webrtc-codereview.appspot.com/1041004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3381 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-17 14:01:20 +00:00
mflodman@webrtc.org
2f225cadde Add logs when no RTCP RR has been received for three regular RTCP intervals.
BUG=1267
TEST=Unittest added.

Review URL: https://webrtc-codereview.appspot.com/1019006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3346 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-09 13:54:43 +00:00
andrew@webrtc.org
14b43beb7c Move src/ -> webrtc/
TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/915006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-22 18:19:23 +00:00