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

[Anton Pryima]

Hello Matthew,

Yes, you were right, I was adding pad instead of requesting it. But I still
not clear about how to do it correctly.
Is it the correct way to request the pad?
GstPadTemplate *templ =
gst_element_class_get_pad_template(GST_ELEMENT_GET_CLASS(webrtc),
"sink_%u");
GstPad *pad = gst_element_request_pad (webrtc, templ, NULL, NULL);

Then, I link this pad with payer, and start my pipeline. But right
after that, I've got error inside of webrtc element:
ERROR:../../../../opensource/gstreamer/vendor/gst-build2/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c:2291:sdp_media_from_transceiver:
assertion failed: (trans->mline == -1 || trans->mline == media_idx)
Bail out!
ERROR:../../../../opensource/gstreamer/vendor/gst-build2/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c:2291:sdp_media_from_transceiver:
assertion failed: (trans->mline == -1 || trans->mline == media_idx)

Did I miss something?

Best regards,
Anton.

On Tue, Oct 6, 2020 at 2:55 AM Matthew Waters <ystreet00 at gmail.com> wrote:

> How are you adding a pad?
>
> If you're calling gst_element_add_pad() with a newly created pad, that's
> wrong.  You need to request a sink pad from webrtcbin and link to that.
>
> On 6/10/20 4:27 am, Anton Pryima wrote:
>
> Hello Matthew.
>
> I've tried to add one more sink pad to webrtcbin at my sending side.
> Linked it, but right after that, I have:
>  *GStreamer-CRITICAL **: 20:16:08.832: chain on pad sendonly:pad0 but it
> has no chainfunction*
>
> Should I just add pad, or do something more?
>
> Best regards,
> Anton.
>
> On Sat, Oct 3, 2020 at 2:05 PM Matthew Waters <ystreet00 at gmail.com> wrote:
>
>> 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>
>> 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 listgstreamer-devel at lists.freedesktop.orghttps://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>>>
>>>
>>>
>>
>
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