AAC streaming

[Frederik]

Hi,
can anybody help me with making a AAC server and client.
Here is what I got
server:
 gst-launch-1.0 -v  audiotestsrc ! audio/x-raw,rate=48000,channels=2
!audioconvert ! audioresample ! avenc_aac ! rtpmp4gpay ! application/x-rtp,
mode=AAC-hbr, encoding-name=MPEG4-GENERIC ! udpsink host=127.0.0.1
port=12000

client:
gst-launch-1.0.exe -v udpsrc port=12000 caps="application/x-rtp,
media=(string)audio, clock-rate=(int)48000,
encoding-name=(string)MPEG4-GENERIC, mode=AAC-hbr" ! rtpmp4gdepay  !
avdec_aac ! autoaudiosink

Everything seems to work out fine, but there is NO sound.
This is from gstreamer log, and it just continues
WARN                   libav
gstavauddec.c:756:gst_ffmpegauddec_handle_frame:<avdec_aac0> decoding error


Can anybody help me out here ?

Thanks



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[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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[Marc Leeman]

See if your data ends up at the end of the pipeline: replace autoaudiosink with 'fakesink dump=1'

If you see hex scrolling on the console, your audio is decoded and sent to your device, something is wrong there.

If not, you can gradually remove elements in your pipeline to figure out where the data gets stuck for some reason.

On Fri, 28 Aug 2020 at 10:15, Frederik <[hidden email]> wrote:

Hi,
thanks for the answer.
Trying to use the exact same pipes a you describe, I get no sound and just a
log full of
basetransform
gstbasetransform.c:1364:gst_base_transform_setcaps:<audioconvert0> transform
could not transform audio/x-raw, format=(string)F32LE,
layout=(string)non-interleaved, rate=(int)44100, channels=(int)1 in anything
we support

Using
client:
udpsrc port=12000  ! application/x-rtp,clock-rate=44100,
config=40002410adca00  ! rtpjitterbuffer ! rtpmp4adepay ! avdec_aac !
audioconvert ! autoaudiosink
server:
audiotestsrc  ! audioconvert ! avenc_aac !  rtpmp4apay !  udpsink
host=127.0.0.1 port=12000




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[Frederik]

Hi,
thank you both for your answers.
The config parameter actually made the difference. As you proposed, I copied
it from the gstreamer dump graph.

Thank you very much for giving me a hint about solving this problem.



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