[gst-devel] AAC RTP Doubt

[Danilo Freire]

Hi,

Im trying to encapsulate a aac stream in rtp packets.

I seem that there is two possible ways. One using the rtpmp4gpay and another
using the rtpmp4apay. Both claims to receive a complete AU (audio frame?).
Which coder should I use with them? I'm using the faac, but it has two
possible output formats (ADTS and AAC RAW), I have to use the AAC RAW?

Is there some plugin to extract the AU from the ADTS stream? like a
adtsparse?

[]s

-- 
Danilo Freire
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