[Ffmpeg-devel] RTP patches & RFC

Ryan Martell rdm4
Wed Oct 11 03:08:04 CEST 2006

This is a rough draft of my rtp_h264 code.  This will not compile  
without adding some diffs, but having never done this before, I  
figured I would post the file and see what I need to fix (formatting  

I ran it through indent, as per the RFC, and it nuked my tabs (some  
may have been reintroduced through editing though- i'll re-indent it  
before I post the final version.  I'm not a fan of the format (i like  
more spaces), but that's just me whining. ;-)

My other reason for posting is to see if anyone can address any of my  
current issues:
  1) Fix video/audio syncing! (frame rate wanders; sometimes it's 10,  
sometimes 29.97.  I suspect I'm not doing something right.
  2) I hand packets to the decoder as I get them, since h264 can be  
backward and forward looking.  i am only updating pts though, and  
those are not strictly increasing.  Is this right?
  3) This works with files hinted by darwin streaming server, which  
only uses FU-A's and STAP-A's (AFAIK).  I need to extend for the  
other rtp/h264 types.
  4) I am using my own packet queues.  This means that a packet comes  
in (from network), gets allocated and copied to a packet, and once a  
full frame is received, is conglomerated and
     copied into an AVPacket.  I got rid of the extra allocations and  
reallocations and copies with the fragmentation packets (see the  
  5) Move the protocol specific rtp extradata up a structure; it's  
currently tagged off the extradata for AAC (sorta)
  6) Fix the double packet allocation in AAC streaming (rtp.c) (not  
my bug)
  7) Figure out why at 120 seconds I lose video, consistently.  I  
suspect it's a parameter in Darwin Streaming Server; it's not getting  
a statistics packet back from the client (or something),
     so it's closing down.  It's very consistent at 120 seconds.
  8) Verify I'm using the correct endianness macros.

Any thoughts on any of the above, plus any and all criticism/ 
commentary is appreciated!
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