Check out this howto page on the unofficial ALSA wiki: <a href="http://alsa.opensrc.org/.asoundrc">http://alsa.opensrc.org/.asoundrc</a> The section you want is about splitting outputs. For inputs, it's the same deal but with a dsnoop plugin rather than dmix. If you want a better understanding of ALSA, I found this page to be very helpful: <a href="http://www.volkerschatz.com/noise/alsa.html">http://www.volkerschatz.com/noise/alsa.html</a> Good luck. <div class="gmail_quote">On Feb 2, 2013 8:31 PM, "Gary Thomas" <<a href="mailto:gary@mlbassoc.com">gary@mlbassoc.com</a>> wrote: <blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"> On 2013-02-02 09:03, Nox Deleo wrote: <blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"> I'd suggest getting a debug graph of this. It should help you see what caps are being used where. See here for instructions: <a href="http://gstreamer.freedesktop.org/wiki/DumpingPipelineGraphs" target="_blank">http://gstreamer.freedesktop.org/wiki/DumpingPipelineGraphs</a> Also, see these instructions on debugging to get more verbose messages that may help pinpoint your problem: <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/manual/html/section-checklist-debug.html" target="_blank">http://gstreamer.freedesktop.org/data/doc/gstreamer/head/manual/html/section-checklist-debug.html</a> I've just been doing some work with ALSA and GStreamer, and I ended up writing a custom ALSA config file to split up my sound cards 8 ins and outs into different devices, so I could access them with separate alsasrc/sink elements. I found it alot easier than dealing with interleaving and channel positions. I don't know what you're trying to do exactly, but that's always an option if you need it. </blockquote> Any details you can share about how you did this would be much appreciated. <blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"> On 2 February 2013 00:22, Gary Thomas <<a href="mailto:gary@mlbassoc.com" target="_blank">gary@mlbassoc.com</a> <mailto:<a href="mailto:gary@mlbassoc.com" target="_blank">gary@mlbassoc.com</a>>> wrote: I'm experimenting with audio streaming, based on this example: <a href="http://cgit.freedesktop.org/__gstreamer/gst-plugins-good/__tree/tests/examples/rtp/__server-alsasrc-PCMA.sh" target="_blank">http://cgit.freedesktop.org/__gstreamer/gst-plugins-good/__tree/tests/examples/rtp/__server-alsasrc-PCMA.sh</a> <<a href="http://cgit.freedesktop.org/gstreamer/gst-plugins-good/tree/tests/examples/rtp/server-alsasrc-PCMA.sh" target="_blank">http://cgit.freedesktop.org/gstreamer/gst-plugins-good/tree/tests/examples/rtp/server-alsasrc-PCMA.sh</a>> When paired with the appropriate client, I can run it fine when the audio source is 'audiotestsrc'. If I change it to be 'alsasrc', I get incompatible caps and it fails. This seems to be because the rtppcmapay element only wants one channel. I've not been able to figure out how to select only one channel of my ALSA source, either by mixing or just choosing one or the other. Here's what I tried that now doesn't complain about the caps being wrong (but I'm sure the issue remains), rather it gives me an "internal data flow error" gst-launch -vvv gstrtpbin name=rtpbin alsasrc device=hw:0,1 \ ! deinterleave name=d d.src0 ! queue ! audioconvert \ ! alawenc ! rtppcmapay ! rtpbin.send_rtp_sink_0 \ rtpbin.send_rtp_src_0 ! udpsink port=5002 host=192.168.1.114 \ rtpbin.send_rtcp_src_0 ! udpsink port=5003 host=192.168.1.114 sync=false async=false \ udpsrc port=5007 ! rtpbin.recv_rtcp_sink_0 Any suggestions on how I make this happy? Thanks -- ------------------------------__------------------------------ Gary Thomas | Consulting for the MLB Associates | Embedded world ------------------------------__------------------------------ _________________________________________________ gstreamer-devel mailing list gstreamer-devel@lists.__<a href="http://freedesktop.org" target="_blank">freedesktop.org</a> <mailto:<a href="mailto:gstreamer-devel@lists.freedesktop.org" target="_blank">gstreamer-devel@lists.freedesktop.org</a>> <a href="http://lists.freedesktop.org/__mailman/listinfo/gstreamer-__devel" target="_blank">http://lists.freedesktop.org/__mailman/listinfo/gstreamer-__devel</a> <<a href="http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel" target="_blank">http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel</a>> _______________________________________________ gstreamer-devel mailing list <a href="mailto:gstreamer-devel@lists.freedesktop.org" target="_blank">gstreamer-devel@lists.freedesktop.org</a> <a href="http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel" target="_blank">http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel</a> </blockquote> -- ------------------------------------------------------------ Gary Thomas | Consulting for the MLB Associates | Embedded world ------------------------------------------------------------ _______________________________________________ gstreamer-devel mailing list <a href="mailto:gstreamer-devel@lists.freedesktop.org" target="_blank">gstreamer-devel@lists.freedesktop.org</a> <a href="http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel" target="_blank">http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel</a> </blockquote></div>