Dear all, I am developing a pipeline to send video and audio muxed via UDP to receive it in another computer. The only problem is that in case of a bidirectional conversation I receive echo, hence I would like to apply the filter: webrtcechoprobe and webrtcdsp, but I do not how to do this. This is the pipeline to send: gst-launch-1.0 -v v4l2src device=/dev/video0 ! video/x-raw, height=480 ! videoconvert \ ! x264enc tune=zerolatency \ ! queue ! mpegtsmux name=mux ! queue ! udpsink host=ipvalue port=portvalue alsasrc \ ! queue ! audioconvert \ ! queue ! opusenc ! queue ! mux. -e And the pipeline to receive: gst-launch-1.0 -v udpsrc port=portvalue ! queue ! tsdemux name=demux demux. \ ! queue ! h264parse ! queue ! avdec_h264 ! queue ! videoconvert \ ! autovideosink demux. ! queue ! opusdec ! queue ! audioconvert ! queue ! alsasink sync=false Would you give me an example about how to perform it? Thanks for your valuable time, Regards, Andrés. _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
Le mar. 22 déc. 2020 10 h 00, Andrés Meseguer Valenzuela <[hidden email]> a écrit :
To use these elements, you need to send and receive in the same pipeline.
Then the webrtcdsp goes here, right after alsasrc. ! queue ! audioconvert \ And the echoprobe right before alsasink. The probe is timestamp driven, so you probably want to use sync=true async=false on that sink. Though the latency will be too hight with mpegtsdemux, hopefully you use a recent GStreamer release and have a latency property to reduce it from the 700ms default. Another note, our TS demuxer does not reorder UDP packets, so avoid this protocol over the internet, use RTP instead. queue ! alsasink sync=false
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