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Tristan,<br>
<br>
If I add 'host' to the udpsink it works ...<br>
<br>
<pre wrap="">gst-launch -v jackaudiosrc connect=none ! audio/x-raw-float, channels=10
! vorbisenc ! rtpvorbispay ! udpsink port=10000 host=127.0.0.1</pre>
<br>
Then both jackaudiosink and jackaudiosrc show up.<br>
<br>
Thanks for the help!<br>
<br>
Best, Dirk<br>
<blockquote cite="mid:4AB77775.5040408@sat.qc.ca" type="cite">
  <pre wrap="">Are you copying the caps from udpsink's sink pad directly from the
sender pipeline? Vorbis caps (i.e. the codebook) will change for
different configurations.

-T

Dirk Griffioen wrote:
  </pre>
  <blockquote type="cite">
    <pre wrap="">Hi Tristan,

Thanks for the replies. They are really helpfull! (And I will have a
further look at 'miville' - it looks really nice).
    </pre>
    <blockquote type="cite">
      <pre wrap="">This example might help:
sender

gst-launch -v jackaudiosrc connect=none ! audio/x-raw-float, channels=10
! vorbisenc ! rtpvorbispay ! udpsink port=10000

receiver

gst-launch -v udpsrc caps="$CAPS" port=10000 ! rtpvorbisdepay !
vorbisdec ! queue max-size-buffers=3 ! jackaudiosink connect=none

where $CAPS are the caps of the udpsink from the first pipeline.


      </pre>
    </blockquote>
    <pre wrap="">This does not work for me, jackaudiosink does not pop up in qjackctl
... I tried some other configurations, but nothing.

However, somehow my first pipeline with rtp decided to work, with 24
channels and from gst-launch. Still, I get:

WARNING: from element /GstPipeline:pipeline0/GstVorbisDec:vorbisdec0:
Could not decode stream.
Additional debug info:
vorbisdec.c(670): vorbis_handle_identification_packet ():
/GstPipeline:pipeline0/GstVorbisDec:vorbisdec0:
Using NONE channel layout for more than 8 channels

Maybe this can interpreted as 'cannot read layout from stream,
defaulting to NONE' - as the audio streams fine.
    </pre>
    <blockquote type="cite">
      <pre wrap="">-Tristan

Tristan Matthews wrote:

      </pre>
      <blockquote type="cite">
        <pre wrap="">As far as I know this should be fine in python, though I haven't tried
it. Our app (<a class="moz-txt-link-freetext" href="https://svn.sat.qc.ca/trac/miville">https://svn.sat.qc.ca/trac/miville</a>) does this for up to 8
channels of vorbis or raw audio (with rtpL16pay/depay) and gstrtpbin. We
haven't tried (yet) to implement support for more than 8 channels. Here
we don't set the channel positions and it works, but I do get that same
"warning could not decode stream" even though the sound if fine.
The element to set the number of channels is a caps filter element, so
the equivalent in C would be:

GstElement *capsfilter;
gst_element_factory_make(capsfilter, NULL);
g_object_set(capsfilter, "caps", "audio/x-raw-float, channels=8", NULL);

and then link it in between jackaudiosrc and vorbisenc.

-Tristan

Dirk Griffioen wrote:


        </pre>
        <blockquote type="cite">
          <pre wrap="">Hi Tristan,


          </pre>
          <blockquote type="cite">
            <pre wrap="">You probably have to set the "channel-positions" property on the
interleave element, which you can't do with gst-launch as I recall (i.e.
you need to write c or python app for it) as it is an array. However for
more than 8 channels the positions might have to all be set to
GST_AUDIO_CHANNEL_POSITION_NONE




            </pre>
          </blockquote>
          <pre wrap="">Yes I read about that
(<a class="moz-txt-link-freetext" href="http://tristanswork.blogspot.com/2008/08/multichannel-audio-with-gstreamer.html">http://tristanswork.blogspot.com/2008/08/multichannel-audio-with-gstreamer.html</a>),
and can I do this in python as well? (I dont mind C, but python will
be quicker).



          </pre>
          <blockquote type="cite">
            <pre wrap="">Are you just trying to do something like this?

gst-launch -v jackaudiosrc connect=none ! audio/x-raw-float, channels=24
! vorbisenc ! vorbisdec ! jackaudiosink connect=none




            </pre>
          </blockquote>
          <pre wrap="">Well, almost :)

I am trying to put rtp in between the vorbisencoder and decoder so I
can stream n channels from A to B over a single rtp session

I get the following on the receiving end (after copying the new config
string from A to B):

GstPipeline:pipeline0/GstVorbisDec:vorbisdec0.GstPad:sink: caps =
audio/x-vorbis
WARNING: from element /GstPipeline:pipeline0/GstVorbisDec:vorbisdec0:
Could not decode stream.
Additional debug info:
vorbisdec.c(670): vorbis_handle_identification_packet ():
/GstPipeline:pipeline0/GstVorbisDec:vorbisdec0:
Using NONE channel layout for more than 8 channels

Which is weird because it knows this:

/GstPipeline:pipeline0/GstVorbisDec:vorbisdec0.GstPad:src: caps =
audio/x-raw-float, rate=(int)48000, channels=(int)24,
endianness=(int)1234, width=(int)32,
channel-positions=(GstAudioChannelPosition)&lt;
GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE,
GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE,
GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE,
GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE,
GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE,
GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE,
GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE,
GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE,
GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE,
GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE,
GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE,
GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE &gt;

Do you have any tips? (Maybe the NONE layout is not in the vorbis
config string ...)

Regards, Dirk

PS - how do you call the 'audio/x-raw-float, channels=24' element?



          </pre>
          <blockquote type="cite">
            <pre wrap="">(note that the bottleneck here will probably be your soundcard).

-Tristan

Dirk Griffioen wrote:



            </pre>
            <blockquote type="cite">
              <pre wrap="">Thanks for the answer!




              </pre>
              <blockquote type="cite">
                <pre wrap="">I think you need to interleave several mono channels from audiotestsrc
into a stereo stream and then shove them into the vorbis encoder. You
need an audio mixer, or something like that.





                </pre>
              </blockquote>
              <pre wrap="">I would like to encode separate channels.

For example, this runs

gst-launch-0.10 -v interleave name=i ! queue ! \
vorbisenc ! vorbisdec ! \
jackaudiosink connect=none \
jackaudiosrc ! audioconvert ! queue ! i. \
jackaudiosrc ! audioconvert ! queue ! i.

but this does not:

gst-launch-0.10 -v interleave name=i ! queue ! \
vorbisenc ! vorbisdec ! \
jackaudiosink connect=none \
jackaudiosrc ! audioconvert ! queue ! i. \
jackaudiosrc ! audioconvert ! queue ! i. \
jackaudiosrc ! audioconvert ! queue ! i. \
jackaudiosrc ! audioconvert ! queue ! i.

Do you know why? The vorbis spec allows for 255 channels and I simply
would like to run n channels through the vorbis encoder ...

I really could use some help.

Thanks in advance, Dirk


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              </pre>
            </blockquote>
            <pre wrap="">

            </pre>
          </blockquote>
          <pre wrap="">------------------------------------------------------------------------

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          </pre>
        </blockquote>
        <pre wrap="">
        </pre>
      </blockquote>
      <pre wrap="">

      </pre>
    </blockquote>
    <pre wrap="">------------------------------------------------------------------------

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    </pre>
  </blockquote>
  <pre wrap=""><!---->

--
Tristan Matthews
Soci&eacute;t&eacute; des arts technologiques [SAT]
email: <a class="moz-txt-link-abbreviated" href="mailto:tristan@sat.qc.ca">tristan@sat.qc.ca</a>
web: <a class="moz-txt-link-freetext" href="http://www.music.mcgill.ca/~tmatthews">http://www.music.mcgill.ca/~tmatthews</a>


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</blockquote>
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