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7 Commits

Author SHA1 Message Date
kwiberg
ac9f876bc0 Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/ 
gmock.h and gtest.h were moved (or rather, got wrappers so that we
could put some icky compatibility hacks in one place instead of 500)
in this CL: https://codereview.webrtc.org/2358993004/

NOPRESUBMIT=true
BUG=webrtc:6398

Review-Url: https://codereview.webrtc.org/2381013002
Cr-Commit-Position: refs/heads/master@{#14464}
 2016-10-01 05:29:53 +00:00
asapersson
3cc47ebd2d Add sanity check for decreasing RTP timestamp in RtpToNtpMs. 
The capture time for a frame (capture_ms) is set later (in ViEEncoder::IncomingCapturedFrame) than the timestamp.
Could potentially cause the RTP timestamp in consecutive RTCP SR to decrease.
Example:
// Frame1 46371: timestamp:2732, capture_ms:46373, rtcp SR ms: 46423 -> estimated current RTP timestamp:2732+(46423-46373)*90 = 7232
// Frame2 46404: timestamp:5702, capture_ms:46412, rtcp SR ms: 46428 -> estimated current RTP timestamp:5702+(46428-46412)*90 = 7142
// Diff:  33 ms:          33 ms,            39 ms,              5 ms

BUG=b/31154867

Review-Url: https://codereview.webrtc.org/2354843003
Cr-Commit-Position: refs/heads/master@{#14454}
 2016-09-30 10:16:26 +00:00
kwiberg
77eab70470 Enable the -Wundef warning for clang 
NOPRESUBMIT=true
BUG=webrtc:6398

Review-Url: https://codereview.webrtc.org/2358993004
Cr-Commit-Position: refs/heads/master@{#14425}
 2016-09-29 00:42:08 +00:00
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