AAC streaming

[gotsring]

Using the pipelines from  this post
<http://gstreamer-devel.966125.n4.nabble.com/AAC-RTP-streaming-td4684775.html> 
, I can get it to work. The main difference is the use of rtpmp4*a*pay vs
rtpmp4*g*pay (and depay). I guess this means you're limited to audio, if
that's ok.

Server:
gst-launch-1.0 audiotestsrc ! 'audio/x-raw, rate=48000, channels=2' !
avenc_aac ! rtpmp4apay ! 'application/x-rtp, mode=AAC-hbr' ! udpsink
host=127.0.0.1 port=12000

Client:
gst-launch-1.0 udpsrc port=12000 ! 'application/x-rtp, clock-rate=48000,
config=40002320adca00' ! rtpjitterbuffer ! rtpmp4adepay ! avdec_aac !
autoaudiosink


FYI, I got the config value in the client caps from looking at the pipeline
of the server from the exported .dot file. See  here for instructions
<https://embeddedartistry.com/blog/2018/02/22/generating-gstreamer-pipeline-graphs/> 
, it's a helpful tool.



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