alsasrc and caps negotiation

Nox Deleo noxdeleo at googlemail.com
Sat Feb 2 13:16:14 PST 2013


Check out this howto page on the unofficial ALSA wiki:
http://alsa.opensrc.org/.asoundrc

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: http://www.volkerschatz.com/noise/alsa.html

Good luck.
On Feb 2, 2013 8:31 PM, "Gary Thomas" <gary at mlbassoc.com> wrote:

> On 2013-02-02 09:03, Nox Deleo wrote:
>
>> I'd suggest getting a debug graph of this. It should help you see what
>> caps are being used where.
>>
>> See here for instructions:
>> http://gstreamer.freedesktop.**org/wiki/DumpingPipelineGraphs<http://gstreamer.freedesktop.org/wiki/DumpingPipelineGraphs>
>>
>> Also, see these instructions on debugging to get more verbose messages
>> that may help pinpoint your problem:
>> http://gstreamer.freedesktop.**org/data/doc/gstreamer/head/**
>> manual/html/section-checklist-**debug.html<http://gstreamer.freedesktop.org/data/doc/gstreamer/head/manual/html/section-checklist-debug.html>
>>
>> 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.
>>
>
> Any details you can share about how you did this would be much appreciated.
>
>
>>
>> On 2 February 2013 00:22, Gary Thomas <gary at mlbassoc.com <mailto:
>> gary at mlbassoc.com>> wrote:
>>
>>     I'm experimenting with audio streaming, based on this example:
>>     http://cgit.freedesktop.org/__**gstreamer/gst-plugins-good/__**
>> tree/tests/examples/rtp/__**server-alsasrc-PCMA.sh<http://cgit.freedesktop.org/__gstreamer/gst-plugins-good/__tree/tests/examples/rtp/__server-alsasrc-PCMA.sh>
>>     <http://cgit.freedesktop.org/**gstreamer/gst-plugins-good/**
>> tree/tests/examples/rtp/**server-alsasrc-PCMA.sh<http://cgit.freedesktop.org/gstreamer/gst-plugins-good/tree/tests/examples/rtp/server-alsasrc-PCMA.sh>
>> >
>>
>>     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
>>     ------------------------------**__----------------------------**--
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>> >
>>
>>
>>
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> --
> ------------------------------**------------------------------
> Gary Thomas                 |  Consulting for the
> MLB Associates              |    Embedded world
> ------------------------------**------------------------------
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