[Live-devel] getNormalPlayTime calculation

[Ross Finlayson]

>So my question: is the rtpSource->curPacketRTPTimestamp() always 
>supposed to be larger than rtpInfo.timestamp?

No, RTP timestamps can wrap around (because they are 32-bit numbers).


>In MediaSubsession::getNormalPlayTime(...), when rtpInfo.infoIsNew, 
>an offset is set as such:
>       u_int32_t timestampOffset = 
>rtpSource()->curPacketRTPTimestamp() - rtpInfo.timestamp;

This is correct.  If "rtpSource()->curPacketRTPTimestamp()" happens 
to be smaller than "rtpInfo.timestamp" (because the RTP timestamp has 
wrapped around), then the actual offset should be
	(2^32 +  rtpSource()->curPacketRTPTimestamp()) - rtpInfo.timestamp
but this is just the same as
	rtpSource()->curPacketRTPTimestamp() - rtpInfo.timestamp
if expressed as a 32-bit unsigned number.

Therefore the current code is correct, I believe.

(This assumes, however, that the RTP timestamp wraps around *at most 
once* before the first RTCP "SR" packet is received (so that we can 
compute a properly-synchronized presentation time).  If your RTP 
timestamp wraps around more than once, then that suggests that either 
your server's RTCP rate is too low, or else your client is not 
receiving RTCP at all.)
-- 

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/