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?
From: gstreamer-devel [mailto:[hidden email]] On Behalf Of Tim Müller
Sent: 31 August 2020 13:51
To: Discussion of the development of and with GStreamer <[hidden email]>
Subject: Re: FLAC-encoded audio via WebRTC
On Mon, 2020-08-31 at 09:50 +0000, Soebirk, Thorsten wrote:
> 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-