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
d66929995f