Webrtc wit appsrc

[Matthew Waters]

It does, but codec-preferences has only been tested for receive only
streams. i.e. configuring which media format to receive.

On 5/8/20 9:40 pm, Anton Pryima wrote:
> So, setting caps on the transceiver is not enough? It does not
> influent SDP generation?
>
> On Wed, Aug 5, 2020 at 2:04 PM Matthew Waters <ystreet00 at gmail.com
> <mailto:ystreet00 at gmail.com>> wrote:
>
>     I'm not sure on the correct way, I have never done appsrc into
>     webrtcbin.  I only know that webrtcbin needs the caps event on all
>     of its sink pads before a coherent sdp can be generated.  How that
>     occurs, you would need to debug, or ask someone who can, help you
>     debug.
>
>     Cheers
>     -Matt
>
>     On 5/8/20 8:41 pm, Anton Pryima wrote:
>>     Hello Matthew,
>>
>>     Thanks for quick reply,
>>     Basically, I don't provide caps. But I'm pushing samples to the
>>     appsrc - so it has caps by default. But I was trying to set caps
>>     explicitly - with no success.
>>     BTW, I should set caps on webrtcbin or on appsrc sinkpad?
>>     Basically, I was able to proceed, with dirty hack - calling
>>     on_negotiation_needed(webrtcbin, userdata) callback function
>>     manually, right after I configure a transceiver.
>>
>>     But what is the correct way?
>>
>>     Best regards,
>>     Anton.
>>
>>     On Wed, Aug 5, 2020 at 8:20 AM Matthew Waters
>>     <ystreet00 at gmail.com <mailto:ystreet00 at gmail.com>> wrote:
>>
>>         Are you providing caps to your appsrc?  I can't remember if
>>         appsrc will delay the caps event until the first buffer or
>>         not so that may be a reason.
>>
>>         On 5/8/20 5:55 am, Anton Pryima wrote:
>>>         Hello all.
>>>
>>>         I have a pipe:
>>>         appsrc->rtph264pay->webrtcbin.sink
>>>
>>>         But, when I setting up everything, and pipeline is not
>>>         running yet, I received an error: 
>>>         DEBUG              webrtcbin
>>>         gstwebrtcbin.c:5722:gst_webrtc_bin_change_state: changing
>>>         state: NULL => READY
>>>         LOG                webrtcbin
>>>         gstwebrtcbin.c:1341:_check_if_negotiation_is_needed:<sendonly>
>>>         checking if negotiation is needed
>>>         LOG                webrtcbin
>>>         gstwebrtcbin.c:1346:_check_if_negotiation_is_needed:<sendonly>
>>>         no negotiation possible until caps have been received on all
>>>         sink pads
>>>
>>>         After that, I'm starting pipeline and it working fine, no
>>>         issues:
>>>
>>>          DEBUG              webrtcbin
>>>         gstwebrtcbin.c:5722:gst_webrtc_bin_change_state: changing
>>>         state: READY => PAUSED
>>>         DEBUG              webrtcbin
>>>         gstwebrtcbin.c:5722:gst_webrtc_bin_change_state: changing
>>>         state: PAUSED => PLAYING
>>>
>>>         And that's all. No more on-negotiation-needed callback. Nothing.
>>>
>>>         How to proceed further with webrtc connection? How to
>>>         re-init it after the pipeline is running to make it call
>>>         on-negotiation-needed callback?
>>>
>>>         Thank you in advance,
>>>         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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