4 Commits

Author SHA1 Message Date
qiangchen
067121ab3f Bug Fix: WebRTC Receiver Timestamp Jump Detection
RTCVideoEncoder does not propagate RTP timestamps properly for encoded video frames, and as such whenever switching between simulcast layers there's a large timestamp gap that causes the incoming stream to freeze (timestamps look like they're either too far ahead or too far behind the previous frame).

Ideally RTCVideoEncoder would propagate these timestamps, but even so, when there's a large timestamp gap it would seem reasonable that the receiver resets quickly and consider this to be a new stream.

This CL detects the large jump for timestamps, if that happens, we reset the time extrapolator, which is the class for convertion from RTP timestamp to clock time.

BUG=chromium:705679

Review-Url: https://codereview.webrtc.org/2776813002
Cr-Commit-Position: refs/heads/master@{#17770}
2017-04-19 16:57:37 +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
andrew@webrtc.org
cc476aa038 Fix a name collision with Android libc++
The Android libc++ has a symbol called '_P'
This CL renames a property called _P in webrtc.

BUG=chromium:427718
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30009004

Patch from Fabrice de Gans-Riberi <fdegans@chromium.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7579 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31 16:01:25 +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