[FFmpeg-devel] [PATCH] RTP packetizer for AAC audio

[Luca Abeni]

Hi all,

I attach the code for supporting AAC audio in RTP streams (the SDP 
generator already contains the necessary bits).

- aac_encapsulation.diff implements the basic packetization algorithm 
(without fragmentation). Some notes:
     + The patch introduces a new rtp_aac.c file (since this is really 
the "AAC-hbr" mode of RFC3640, maybe the file should be called 
rtp_rfc3640.c, in case someone implements the other modes in the future?).
     + The maximum number of AAC frames per packet is fixed (maybe we 
can make it runtime-configurable in the future).
     + Instead of adding a new field to the RTPDemuxContext structure, I 
reused the "read_buf_index" field (which was only used in receiving AAC 
audio). If this is not ok, I'll fix the problem.
     + The patch depends on the two "RTP timestamps" patches that I 
posted some time ago
     + Fragmentation support is not implemented because I've not been 
able to generate big AAC frames, so I could not test it
     + As usual, the patch has been tested using vlc and openrtsp as 
clients, and it works ok

- aac_fragmented_frames.diff implements fragmentation support in the AAC 
packetizer. As said, I've not been able to test this code (this is why 
it is in a second patch).


				Luca
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