Hi All,
I was testing out audio mixing from a network source in the following way:
Network sender:
gst-launch-1.0 audiotestsrc is-live=true freq=880 ! audioconvert !
audioresample ! opusenc ! rtpopuspay ! udpsink host=127.0.0.1 port=5000
Network receiver:
GST_DEBUG=6 gst-launch-1.0 -vvvv audiomixer name=mix mix. ! queue !
audioconvert ! audioresample ! autoaudiosink audiotestsrc is-live=true
volume=0.2 ! mix. udpsrc port=5000
caps="application/x-rtp,media=(string)audio,clock-rate=48000,encoding-params=2,encoding-name=(string)OPUS"
! rtpopusdepay ! opusdec ! mix.
Results in the following problem, but it is unclear where the source of the
issue is coming from:
GStreamer error: clock problem.
Any insights into correcting this?
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