[Live-devel] receiving and decoding AAC-hbr

[Jon Burgess]

Hi,

I have subclassed MediaSink in an attempt to provide an adapter class
between live555 and the audio APIs on iOS (iPhone).  The RTSP session is set
up using the openRTSP test program (except my custom subclass of MediaSink
is used in place of FileSink).

The stream being received is an AAC-hbr stream.

I am getting *some* audio to decode/render, but it seems as though the data
is being consumed faster than it is being supplied, resulting in audio
rendering to stop (it's quite likely a problem down stream from live555, but
just trying to rule things out).

I just want to double check that each frame received from the
MPEG4GenericRTPSource is actually a depacketized, "raw" AAC frame - i.e.
stripped of all RTP and AU headers etc, and ready to be sent on for
decoding.

Also, the server (in this case Darwin 5.5.5) is sending out multiframed RTP
packets (about 6 or 7 frames per RTP packets) - my sink is being notified of
new data but with the same presentation timestamp for 6 or 7 frames in a row
(it then increases).  I'm guessing that the MPEG4GenericeRTPSource is
setting the pts based on the RTP timestsamp, and not calculating for each
frame within the packet, and that's why I'm seeing this behaviour.  Is this
correct?  The pts is ignored by my code anyway - again, just trying to
understand things.

Regards,
Jon Burgess
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.live555.com/pipermail/live-devel/attachments/20101101/4bb01142/attachment-0001.html>