[Ffmpeg-devel] MPEG4 streaming over RTP

Luca Abeni lucabe72
Fri Jul 22 18:33:51 CEST 2005


On Fri, 2005-07-22 at 17:43 +0200, Michael Niedermayer wrote:

> > Well, I believe that all the rtp_send_* function should be changed, but
> > I've only been able to test the MPEG4 case...
> > I can provide a patch that modifies all the functions to set M to in the
> > last packet of a frame, but I was waiting for the original author's
> > opinion before changing more code.
> well, maybe you should email fabrice directly then, iam not sure if he reads 
> all posts here ...
Ok, I emailed Fabrice directly. In the meanwhile, I did some additional
research on the RFCs, and I found out that
- for audio streams, the M bit should be set to 1 at the beginning of a
"talk-spurt", after silence. So, I think it could be safe to leave it
always to 0
- for mpeg1/2 video, M should be set to 1 for the last packet of a video
frame (it is easy to fix this; I'll send a patch later)
- for mpeg TS, M should be set to 1 only in presence of timestamp
discontinuities. So, I think it could be safe to leave it to 0.


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