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In the case where a network glitch causes a packet to arrive so late that the jitter buffer has gone into expand mode, the playout timestamp could have been increased to a value that is larger than the RTP timestamp of the late packet when it finally arrives. This causes the difference to be negative, and would make the value wrap (unsigned). With this fix, the difference is set to zero when the playout timestamp is ahead of the incoming RTP timestamp. Further down in the method, a zero-value will lead to the averaging filter not being updated. BUG=3080 R=henrika@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/10199004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5735 4adac7df-926f-26a2-2b94-8c16560cd09d
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.