WebRTC to MKV or hlssink2 - problems with audio+video streams together

[Jan Schmidt]

Hi,

On 3/11/20 12:26 am, d3im wrote:
> Hi,
> I'm using gst 1.16.2 and trying to modify webrtc-sendrecv.c to save received
> streams to file (HLS would be perfect) instead of play with autosinks.
>
> When I save only one stream it seem OK - audio alone is fine video alone
> also OK.
>
> When I try to put them together it quits:
> ERROR:/var/tmp/portage/media-libs/gst-plugins-good-1.16.2/work/gst-plugins-good-1.16.2/gst/multifile/gstsplitmuxsink.c:2008:handle_gathered_gop:
> assertion failed: (queued_gop_time >= 0)
> Bail out!
> ERROR:/var/tmp/portage/media-libs/gst-plugins-good-1.16.2/work/gst-plugins-good-1.16.2/gst/multifile/gstsplitmuxsink.c:2008:handle_gathered_gop:
> assertion failed: (queued_gop_time >= 0)

I fixed that problem recently in 
https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/commit/1316dd9c653dc8e6e6eff23ba922549745e4daf3 
- but it's only in git master at this point.

I think one solution for you would be to add a buffer probe on the audio 
stream and drop any audio before the first video frame arrives.

Cheers,

Jan.

> (SIGABRT)
>
> Is there any technique how to do it correctly?
>
> I enclose graph of it:
> 0.dot <http://gstreamer-devel.966125.n4.nabble.com/file/t379376/0.dot>
> 0.pdf <http://gstreamer-devel.966125.n4.nabble.com/file/t379376/0.pdf>
>
>
>
> --
> Sent from: http://gstreamer-devel.966125.n4.nabble.com/
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