AAC/RTP streaming

[Martin Vachovski]

Hi Nicolas,

Now the pipeline works.

Many thanks!
Martin

________________________________________
From: gstreamer-devel <gstreamer-devel-bounces at lists.freedesktop.org> on behalf of Nicolas Dufresne <nicolas at ndufresne.ca>
Sent: Tuesday, September 26, 2017 7:21 PM
To: Discussion of the development of and with GStreamer
Subject: Re: AAC/RTP streaming

Le mardi 26 septembre 2017 à 11:05 +0000, Martin Vachovski a écrit :
> Hello everybody,
>
> I am trying to implement AAC streaming overt RTP, but the receiving
> pipeline doesn't play any audio (although it says it goes to PLAYING
> state).
> I am using the following pipelines with gstreamer 1.12.2:
>
> TX:
> gst-launch-1.0.exe -v audiotestsrc ! audioconvert ! avenc_aac !
> rtpmp4apay ! udpsink
>
> RX:
> gst-launch-1.0.exe -v udpsrc ! application/x-rtp,clock-rate=44100 !
> rtpjitterbuffer ! rtpmp4adepay ! avdec_aac ! audioconvert !
> autoaudiosink

You are sending RAW AAC, this means the other side require the config
header. It's good, since it's pretty low latency, but you have to
signal that config and add config=... to your receiver caps. For this
pipeline, try:

gst-launch-1.0.exe -v udpsrc ! application/x-rtp,clock-rate=44100,config=40002410adca00 ! rtpjitterbuffer ! rtpmp4adepay ! avdec_aac ! audioconvert ! autoaudiosink

>
> (I have also tried replacing the "avdec_aac" element with a
> "decodebin").
> The pipelines don't report any errors, but the second one doesn't
> play any sound.
> I wonder if there is any other setting, caps I need to specify? I
> know that the Vorbis codec,
> for instance, needs the DCT coefficients of the transform as extra
> data when streaming over the network?
>
> Also, if I merge the two pipelines together (omitting the
> udpsink/udpsrc pair), the pipeline works:
> audiotestsrc volume=0.1 ! audioconvert ! avenc_aac ! rtpmp4apay !
> application/x-rtp,clock-rate=44100 ! rtpjitterbuffer ! rtpmp4adepay !
> avdec_aac ! audioconvert ! wavescope ! videoconvert ! autovideosink
>
> I have also tried to supply all the CAPS as reported by the SRC pad
> of "rtpmp4apay " before the "rtpjitterbuffer", still no luck
> /GstPipeline:pipeline0/GstRtpMP4APay:rtpmp4apay0.GstPad:src: caps =
> application/x-rtp, media=(string)audio, clock-rate=(int)44100,
> encoding-name=(string)MP4A-LATM, cpresent=(string)0,
> config=(string)40002410adca00, payload=(int)96,
> ssrc=(uint)3854776221, timestamp-offset=(uint)466007625, seqnum-
> offset=(uint)12210
>
> Many thanks!
> Martin
>
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