Playback issues when saving streamed aac audio to file

[erlend_ne]

Greetings,

I need to send aac encoded audio over rtp and then save it as a mp4 track. I
have gotten it to work somewhat, however I get various errors / warnings
from ffmpeg when trying to play the saved file.

Send pipeline:

gst-launch-1.0 -v audiotestsrc do-timestamp=true is-live=true ! audioconvert
! avenc_aac ! aacparse ! rtpmp4apay !
application/x-rtp,clock-rate=44100,config=40002410adca00 ! udpsink
host=127.0.0.1 port=5000

Receive and save pipeline:

gst-launch-1.0 -v -e udpsrc port=5000 do-timestamp=false !
application/x-rtp,clock-rate=44100,config=40002410adca00 ! rtpjitterbuffer !
rtpmp4adepay ! aacparse ! mp4mux ! filesink location=aac.mp4

The ffmpeg error messages vary from each time. There are various
combinations of the messages "Sample rate index in program config element
does not match the sample rate index configured by the container", "channel
element 2.5 is not allocated", " Multiple frames in a packet", "Reserved bit
set", and "Prediction is not allowed in AAC-LC".

I suspect the issue has to do with the fact that I only send raw aac,
without adts headers. That's why I included the config parameter in the
caps, as described here:
http://gstreamer-devel.966125.n4.nabble.com/AAC-RTP-streaming-td4684775.html
. However, since I still have issues, I would very much appreciate if anyone
knows what I'm doing wrong, or if there are payloader elements or code
examples for including adts headers in the aac rtp stream.

Best regards,
Erlend



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