RTP playback issue

nicolas at ndufresne.ca nicolas at ndufresne.ca
Sun Dec 11 03:16:55 UTC 2016


Le vendredi 09 décembre 2016 à 19:22 -0800, Vinod Kesti a écrit :
> gst-launch-1.0 udpsrc port=9000  !
> application/x-rtp,clock-rate=90000,payload=33 ! rtpjitterbuffer !
> rtpmp2tdepay ! decodebin name=dec dec.! video/x-raw ! fakesink dec.!
> audio/x-raw ! fakesink  --gst-debug=3 
> 
> Pipeline started working when capssetter element was replaced by
> capsfilter
> element int the pipeline.

Indeed, capssetter for your use case would require you to set
join=false. The default behavior is to only overwrite the incoming
fields. udpsrc will produce ANY, which has no field set.

The capsfilter will work with intersection, which means the operation
always return the caps you have set.

regards,
Nicolas
-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 181 bytes
Desc: This is a digitally signed message part
URL: <https://lists.freedesktop.org/archives/gstreamer-devel/attachments/20161210/5dbf7676/attachment.sig>


More information about the gstreamer-devel mailing list