[Live-devel] fDurationInMicroseconds

[Ross Finlayson]

>It seems as if fDurationInMicroseconds is not being calculated for some of 
>the RTPSources, at least for audio codecs.

Yes, to date I've been a bit lax about setting "fDurationInMicroseconds" 
for "RTPSource" subclasses, for a couple of reasons:

1/ For "RTPSource"s, it isn't really needed, because any client that reads 
from a "RTPSource" doesn't really need to know how long each incoming frame 
lasts - instead, it just waits until the next incoming RTP packet arrives.

2/ For some "RTPSource"s, computing "fDurationInMicroseconds" for incoming 
RTP packet data would be non-trivial.  E.g., doing this for MP3 data would 
require parsing the MP3 header bytes - something that otherwise isn't needed.

>Where should this parameter be set, for example in streaming WAVs?

If you were to set "fDurationInMicroseconds", you'd probably set it in the 
same part of the code where you set "fFrameSize".


	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>