I have trouble streaming pulse audio monitor via rtp to an audio player like vlc or gst-launch with udpsrc
this command works and the file has audio that is currently being played
gst-launch-1.0 -v pulsesrc device = "alsa_output.pci-0000_00_1b.0.analog-stereo.monitor" ! opusenc ! oggmux ! filesink location=test.ogg
but when i use this,
gst-launch-1.0 -v pulsesrc device="alsa_output.pci-0000_00_1b.0.analog-stereo.monitor" ! opusenc ! rtpopuspay ! udpsink host=0.0.0.0 port=4000
vlc (from an android phone) tells me that it cannot play the stream with uri rtp://ip-addr:4000
and gst-launch from same machine starts but the resulting file is empty.
gst-launch-1.0 -v udpsrc uri=rtp://0.0.0.0:4000 ! rtpopusdepay ! oggmux ! filesink location=test.ogg
gstreamer version is
$ gst-launch-1.0 --version gst-launch-1.0 version 1.16.0 GStreamer 1.16.0
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Answer
Just started this account, I wanted to add this as a comment, but couldn’t because of rep limitations.
I’m not experienced in using vlc, but at least I get your GStreamer pipelines working if I add the caps, definitions of the rtp stream parameters, to the rtpopusdepay.
So instead of:
gst-launch-1.0 -v udpsrc uri=rtp://0.0.0.0:4000 ! rtpopusdepay ! oggmux ! filesink location=test.ogg
you’ll need to use:
gst-launch-1.0 udpsrc uri=udp://0.0.0.0:4000 ! application/x-rtp,payload=96,encoding-name=OPUS ! rtpopusdepay ! opusdec ! autoaudiosink
for GStreamer. The mandatory parts are the payload and encoding-name, others you can find from gst-inspect-1.0 rtpopuspay/rtpopusdepay. You might need to change the numbers depending on what you define on the server side/what’s the default on your machine.
So in conclusion, I got that GStreamer pipeline working by moving the rtp definitions to the caps for rtpopusdepay. As I said, I’m not familiar with vlc, so I don’t know how to define those GStreamer caps there, if it even depends on those, but I hope this gives some insight on your work.