4 Commits

Author	SHA1	Message	Date
asapersson	fdca66910a	Potential division by zero in RtpToNtpMs() in rtp_to_ntp.cc. ... CalculateFrequency() results in zero frequency (floating point) if the RTP timestamps in the RTCP list are equal. Added check in UpdateRtcpList to not insert RTCP SR with the same RTP timestamp. BUG=webrtc:5780 Review URL: https://codereview.webrtc.org/1891703002 Cr-Commit-Position: refs/heads/master@{#12429}	2016-04-19 14:04:52 +00:00
asapersson	f8cdd184d5	Add histogram stats for AV sync stream offset: ... "WebRTC.Video.AVSyncOffsetInMs" The absolute value of the sync offset between a rendered video frame and the latest played audio frame is measured per video frame. The average offset per received video stream is recorded when a stream is removed. Updated sync tests in call_perf_tests.cc to use this implementation. BUG=webrtc:5493 Review URL: https://codereview.webrtc.org/1756193005 Cr-Commit-Position: refs/heads/master@{#11993}	2016-03-15 08:00:54 +00:00
Henrik Kjellander	98f53510b2	system_wrappers: rename interface -> include ... BUG=webrtc:5095 R=tommi@webrtc.org Review URL: https://codereview.webrtc.org/1413333002 . Cr-Commit-Position: refs/heads/master@{#10438}	2015-10-28 17:17:50 +00:00
wu@webrtc.org	66773a032a	Move timestamp_extrapolator and rtp_to_ntp to system wrapper so that it can be shared by both audio and video engine. ... BUG=3111 TEST=try bots R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13459004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6074 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-05-07 17:09:44 +00:00