Webrtc wit appsrc
Matthew Waters
ystreet00 at gmail.com
Wed Aug 5 05:20:06 UTC 2020
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
> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
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