Hey folks,
I still working on this conferencing solution, but I am also still stuck at the same thing. The problem with the audio part is, that each participant should not hear himself. So I constructed the pipeline as in the picture. It doesn't put out any packets though. If I use a rtpmuxer and only one udpsink / multiudpsink at the end, it works, but then any participant gets the muxed audio of all participants. Any comment is appreciated! -- Michael Niemand Albusstr. 17 60313 Frankfurt/M Germany
Tel.: +49 (0) 69 200 130 60 _______________________________________________ gstreamer-devel mailing list [hidden email] http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel Pipe.png (69K) Download Attachment |
Hi,
Such a pipeline is the right way to do it, but you're very likely to hit some corner case. You may want to make the sinks have async=false for a start. I've been meaning for a long time to write this kind of application with Farstream, but I never got to it. Olivier On Mon, 2012-02-27 at 13:51 +0100, Michael Niemand wrote: > Hey folks, > > I still working on this conferencing solution, but I am also still > stuck at the same thing. > > The problem with the audio part is, that each participant should not > hear himself. So I constructed the pipeline as in the picture. It > doesn't put out any packets though. > > If I use a rtpmuxer and only one udpsink / multiudpsink at the end, it > works, but then any participant gets the muxed audio of all > participants. > > Any comment is appreciated! > > > > -- > Michael Niemand > > Albusstr. 17 > 60313 Frankfurt/M > Germany > > > Tel.: +49 (0) 69 200 130 60 > Mobile: +49 (0) 171 645 29 07 > > > eMail: [hidden email] > sip:[hidden email] > > _______________________________________________ > gstreamer-devel mailing list > [hidden email] > http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel Olivier Crête [hidden email] _______________________________________________ gstreamer-devel mailing list [hidden email] http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel signature.asc (205 bytes) Download Attachment |
For everybody that is interested: I managed to get it working. My problem was in my java code, not gstreamer. Once I got a working pipeline in the command line, the java code was easy to adapt.
here is the working command line:
GST_DEBUG_DUMP_DOT_DIR=/tmp/ gst-launch --gst-debug-level=2 gstrtpbin name=rtpbin latency=2 sync=false udpsrc name=udpsrc0 caps="application/x-rtp, media=audio, clock-rate=8000, payload=0, encoding-name=PCMU" port=5000 ! rtpbin.recv_rtp_sink_0 rtpbin. ! rtppcmudepay ! mulawdec ! tee name=t0 ! queue max-size-buffers=150 ! liveadder name=mix1 ! mulawenc ! rtppcmupay ! udpsink name=udpsink1 host=192.168.0.101 port=5050 t0. ! queue max-size-buffers=150 ! liveadder name=mix2 ! mulawenc ! rtppcmupay ! udpsink name=udpsink2 host=192.168.0.112 port=5050 rtpbin. ! rtppcmudepay ! mulawdec ! tee name=t1 ! queue max-size-buffers=150 ! liveadder name=mix0 ! mulawenc ! rtppcmupay ! udpsink name=udpsink0 host=192.168.0.101 port=5051 t1. ! queue max-size-buffers=150 ! mix2. rtpbin. ! rtppcmudepay ! mulawdec ! tee name=t2 ! queue max-size-buffers=150 ! mix0. t2. ! queue max-size-buffers=150 ! mix1.
yes, it's a hell of a pipeline, I know but it does what it should and also puts out a dot-file that you can use to generate a PNG of the pipeline which is very useful for debugging. I hope putting this here helps somebody with similar problems!
regards, Michael
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