FLAC-encoded audio via WebRTC

[Soebirk, Thorsten]

Thanks Tim.

I've changed the pipeline to:

... ! queue ! flacenc ! rtpgstpay ! queue ! capsfilter caps=application/x-rtp,media=application,encoding-name=X-GST

Now I don’t get any errors, but the streams never start. ICE negotiation takes place as usual, but the receiving side no longer gets pad-added signals - neither for audio nor video (which is unchanged). The sending side uses a blocking appsrc to send the audio. The program now pushes buffers for only a few seconds before being blocked - another indication that nothing is streaming.

Am I overlooking something additional that needs to be changed or added on either the sending or receiving side? 

Thanks!

/ Thorsten

-----Original Message-----
From: gstreamer-devel [mailto:gstreamer-devel-bounces at lists.freedesktop.org] On Behalf Of Tim Müller
Sent: 31 August 2020 13:51
To: Discussion of the development of and with GStreamer <gstreamer-devel at lists.freedesktop.org>
Subject: Re: FLAC-encoded audio via WebRTC

On Mon, 2020-08-31 at 09:50 +0000, Soebirk, Thorsten wrote:

Hi Thorsten,

> I have tried changing it to:
> 
> appsrc ! audio/x-raw, channels=1, rate=16000, format=S16LE, 
> layout=interleaved ! audioconvert ! audioresample ! queue ! flacenc !
> rtpgstpay ! queue ! capsfilter caps=application/x-
> rtp,media=audio,encoding-name=FLAC,payload=96
> 

rtpgstpay will output

   application/x-rtp,media=application,encoding-name=X-GST 

Note that rtpgstpay is something that will only work with GStreamer clients, browsers won't be able to handle it.

Cheers
 Tim

--
Tim Müller, Centricular Ltd - http://www.centricular.com

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