AAC streaming

[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 <devfrederik at gmail.com> 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
>
>
>
>
> --
> Sent from: http://gstreamer-devel.966125.n4.nabble.com/
> _______________________________________________
> gstreamer-devel mailing list
> gstreamer-devel at lists.freedesktop.org
> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>


-- 
g. Marc
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://lists.freedesktop.org/archives/gstreamer-devel/attachments/20200828/278ed78f/attachment.htm>