webrtc: Padname src_0 is not unique in element webrtcbin0, not adding

[Matthew Waters]

Add a new sink pad on the webrtcbin be requesting a new sink pad, wait
for caps to reach the new pad and then perform another sdp negotiation
cycle.

On 3/10/20 6:49 pm, Anton Pryima wrote:
> Hello Matthew, 
>
> Thanks for your quick response.
>
> What is the best way to achieve this? Add one more transceiver at the
> sending side for the same pipeline? Or I need to create another
> pipeline with the same webrtc sink?
>
> Best regards,
> Anton.
>
> On Sat, Oct 3, 2020 at 2:16 AM Matthew Waters <ystreet00 at gmail.com
> <mailto:ystreet00 at gmail.com>> wrote:
>
>     It sounds like you're attempting a renegotiation of the stream
>     format for the same mline.  That is currently an entirely
>     unsupported reconfiguration scenario :).  For now should add a new
>     stream and remove the old stream if you want to change formats.
>
>     Cheers
>     -Matt
>
>     On 3/10/20 4:09 am, Anton Pryima wrote:
>>     Hello all.
>>
>>     I have an issue with the webrtcbin and changing pipes. 
>>     My sender is:
>>
>>     appsrc->parser->payer->webrtcbin
>>
>>     My receiver is:
>>
>>     webrtcbin->depayer->parser->decodebin->autovideosink.
>>
>>     My app source can push 2 kinds of samples: H264 and H265. I start
>>     my send/recv pipelines on the H264 codec. 
>>     Then, when the pipeline is PLAYING and I have the h265 sample, I
>>     put a blocking probe and dynamically reconfigure the sending pipe
>>     to *appsrc->h265parser->h265payer->webrtcbin*. And everything is
>>     fine and my sending pipe continues working.
>>     On the receiving side, I've got the signal through the data
>>     channel about codec change from the sending side and configure my
>>     receiving pipeline
>>     to*webrtcbin->h265depay->h265parse->decodebin->autovideosink*.
>>     But right after that, I hove:
>>      *GStreamer-CRITICAL **: 16:24:34.926: Padname src_0 is not
>>     unique in element webrtcbin0, not adding*
>>     Error and frames stop flowing through webrtc src_0 pad.
>>
>>     Can anyone suggest what am I doing wrong and how to resolve this?
>>
>>     Best regards,
>>     Anton.
>>
>>     _______________________________________________
>>     gstreamer-devel mailing list
>>     gstreamer-devel at lists.freedesktop.org <mailto:gstreamer-devel at lists.freedesktop.org>
>>     https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>

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