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<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>I think this is just defaulting to your HW capabilities. Use <b>&#8211;v</b>
to see whats happening.<o:p></o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>examples on my machine, <o:p></o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>gst-launch <b>-v </b>alsasrc device=hw:1 ! audioconvert !
alsasink device=hw:1<o:p></o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>gst-launch <b>-v</b> alsasrc device=hw:0 ! audioconvert !
alsasink device=hw:0<o:p></o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>produce differing numbers of source channels. I suspect your
older computer only has one channel.<o:p></o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>&gt;LAUNCH 1<o:p></o:p></span></p>

<p class=MsoNormal><i><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>/pipeline0/alsasrc0.src: caps = audio/x-raw-int,
endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16,
rate=(int)48000, channels=(int)1<o:p></o:p></span></i></p>

<p class=MsoNormal><i><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>&#8230;&#8230;.&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <o:p></o:p></span></i></p>

<p class=MsoNormal><i><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>/pipeline0/audioconvert0.src: caps = audio/x-raw-int,
endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16,
rate=(int)48000, channels=(int)2<o:p></o:p></span></i></p>

<p class=MsoNormal><i><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>/pipeline0/audioconvert0.sink: caps = audio/x-raw-int,
endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16,
rate=(int)48000, channels=(int)1<o:p></o:p></span></i></p>

<p class=MsoNormal><i><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>/pipeline0/alsasink0.sink: caps = audio/x-raw-int,
endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16,
rate=(int)48000, channels=(int)2<o:p></o:p></span></i></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>&gt;LAUNCH 2<o:p></o:p></span></p>

<p class=MsoNormal><i><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>pipeline0/alsasrc0.src: caps = audio/x-raw-int,
endianness=(int)1234, signed=(boolean)true, width=(int)32, depth=(int)32,
rate=(int)44100, channels=(int)2<o:p></o:p></span></i></p>

<p class=MsoNormal><i><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>&#8230;.<o:p></o:p></span></i></p>

<p class=MsoNormal><i><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>/pipeline0/audioconvert0.src: caps = audio/x-raw-int,
endianness=(int)1234, signed=(boolean)true, width=(int)32, depth=(int)32,
rate=(int)44100, channels=(int)2<o:p></o:p></span></i></p>

<p class=MsoNormal><i><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>/pipeline0/audioconvert0.sink: caps = audio/x-raw-int,
endianness=(int)1234, signed=(boolean)true, width=(int)32, depth=(int)32,
rate=(int)44100, channels=(int)2<o:p></o:p></span></i></p>

<p class=MsoNormal><i><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>New clock: GstAudioSrcClock<o:p></o:p></span></i></p>

<p class=MsoNormal><i><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>/pipeline0/alsasink0.sink: caps = audio/x-raw-int,
endianness=(int)1234, signed=(boolean)true, width=(int)32, depth=(int)32,
rate=(int)44100, channels=(int)2</span></i><span style='font-size:11.0pt;
font-family:"Calibri","sans-serif";color:#1F497D'>/<o:p></o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>I can do <o:p></o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>gst-launch <b>-v</b> alsasrc device=hw:0 ! alsasink device=hw:0<o:p></o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>but not<o:p></o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>gst-launch <b>-v </b>alsasrc device=hw:1 ! alsasink device=hw:1<o:p></o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>as I need to modify the number of channels, I need the
audioconvert for this one!<o:p></o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

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<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> Viraj Karandikar
[mailto:virajk@gmail.com] <br>
<b>Sent:</b> 29 September 2009 14:13<br>
<b>To:</b> Discussion of the development of GStreamer<br>
<b>Subject:</b> Re: [gst-devel] One microphone, recording mono fails<o:p></o:p></span></p>

</div>

<p class=MsoNormal><o:p>&nbsp;</o:p></p>

<div>

<p class=MsoNormal>Hi,<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>I dont see any issue in using audioconvert.<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>Did you see the log by setting debug level to 5?<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>When exactly &quot;Could not negotiate format&quot; error is
coming? it might give some clue on whats happening.<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal style='margin-bottom:12.0pt'>-Viraj<o:p></o:p></p>

</div>

<div>

<p class=MsoNormal>On Tue, Sep 29, 2009 at 5:55 PM, Jan Martinek &lt;<a
href="mailto:honza@dp.fce.vutbr.cz">honza@dp.fce.vutbr.cz</a>&gt; wrote:<o:p></o:p></p>

<p class=MsoNormal>Hi,<br>
<br>
this is interesting, thank you. But is it a recommended, clean solution<br>
or just workaround?<br>
<br>
I have tried mono recording on a different (older) computer and it works<br>
fine there. Relevant line from &quot;lspci&quot; is this:<br>
<br>
00:11.5 Multimedia audio controller: VIA Technologies, Inc.<br>
VT8233/A/8235/8237 AC97 Audio Controller (rev 50)<br>
<br>
On the other hand, one channel recording fails on my computer with<br>
00:14.2 Audio device: ATI Technologies Inc SBx00 Azalia (Intel HDA)<br>
<br>
Does it depend on hardware? Do capabilities differ? I would be<br>
surprised, because this seems to me as very basic functionality.<br>
<br>
Thank you,<br>
Jan Martinek<o:p></o:p></p>

<div>

<p class=MsoNormal><br>
On 09/29/2009 01:57 PM, Viraj Karandikar wrote:<br>
&gt; Hi,<br>
&gt; You can capture 2 channels and then use audioconvert to convert it to<br>
&gt; mono channel.<br>
&gt; gst-launch gconfaudiosrc ! capsfilter \<br>
&gt; caps=audio/x-raw-int,rate=44100,channels=2,depth=16 ! \<br>
&gt; audioconvert !<br>
&gt; audio/x-raw-int,rate=8000,depth=16,channels=1,width=16,signed=\(boolean\)TRUE,endianness=\(int\)1234<br>
&gt; ! wavenc ! filesink location=test.wav<br>
&gt; regards,<br>
&gt; Viraj<br>
&gt;<br>
&gt; On Tue, Sep 29, 2009 at 4:53 PM, Jan Martinek &lt;<a
href="mailto:honza@dp.fce.vutbr.cz">honza@dp.fce.vutbr.cz</a><o:p></o:p></p>

</div>

<div>

<div>

<p class=MsoNormal>&gt; &lt;mailto:<a href="mailto:honza@dp.fce.vutbr.cz">honza@dp.fce.vutbr.cz</a>&gt;&gt;
wrote:<br>
&gt;<br>
&gt; &nbsp; &nbsp; Hello,<br>
&gt;<br>
&gt; &nbsp; &nbsp; I am unable to record only one channel (mono). I have only
one<br>
&gt; &nbsp; &nbsp; microphone attached to sound card. But if I run this<br>
&gt;<br>
&gt; &nbsp; &nbsp; gst-launch gconfaudiosrc ! capsfilter \<br>
&gt; &nbsp; &nbsp; caps=audio/x-raw-int,rate=44100,channels=1,depth=16 ! \<br>
&gt; &nbsp; &nbsp; wavenc ! filesink location=test.wav<br>
&gt;<br>
&gt; &nbsp; &nbsp; then I get this error message:<br>
&gt;<br>
&gt; &nbsp; &nbsp; Setting pipeline to PAUSED ...<br>
&gt; &nbsp; &nbsp; ERROR: Pipeline doesn't want to pause.<br>
&gt; &nbsp; &nbsp; ERROR: from element<br>
&gt; &nbsp; &nbsp;
/GstPipeline:pipeline0/GstGConfAudioSrc:gconfaudiosrc0/GstBin:bin0/GstAlsaSrc:alsasrc0:<br>
&gt; &nbsp; &nbsp; Could not negotiate format<br>
&gt; &nbsp; &nbsp; Additional debug info:<br>
&gt; &nbsp; &nbsp; gstbasesrc.c(2584): gst_base_src_start ():<br>
&gt; &nbsp; &nbsp; /GstPipeline:pipeline0/GstGConfAudioSrc:gconfaudiosrc0/GstBin:bin0/GstAlsaSrc:alsasrc0:<br>
&gt; &nbsp; &nbsp; Check your filtered caps, if any<br>
&gt; &nbsp; &nbsp; Setting pipeline to NULL ...<br>
&gt; &nbsp; &nbsp; Freeing pipeline ...<br>
&gt;<br>
&gt;<br>
&gt; &nbsp; &nbsp; It can be solved if I use two channels for recording, so
this<br>
&gt;<br>
&gt; &nbsp; &nbsp; gst-launch gconfaudiosrc ! capsfilter \<br>
&gt; &nbsp; &nbsp; caps=audio/x-raw-int,rate=44100,channels=2,depth=16 ! \<br>
&gt; &nbsp; &nbsp; wavenc ! filesink location=test.wav<br>
&gt;<br>
&gt; &nbsp; &nbsp; works fine. But I would like to record one channel because
further data<br>
&gt; &nbsp; &nbsp; processing is little easier.<br>
&gt;<br>
&gt; &nbsp; &nbsp; Thank you<br>
&gt; &nbsp; &nbsp; Jan Martinek<br>
&gt;<br>
&gt; &nbsp; &nbsp;
------------------------------------------------------------------------------<br>
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<p class=MsoNormal>&gt; &nbsp; &nbsp; &lt;mailto:<a
href="mailto:gstreamer-devel@lists.sourceforge.net">gstreamer-devel@lists.sourceforge.net</a>&gt;<o:p></o:p></p>

<div>

<p class=MsoNormal>&gt; &nbsp; &nbsp; <a
href="https://lists.sourceforge.net/lists/listinfo/gstreamer-devel"
target="_blank">https://lists.sourceforge.net/lists/listinfo/gstreamer-devel</a><br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; --<br>
&gt; - Viraj<br>
&gt; Reality is merely an illusion, albeit a very persistent one.<br>
&gt;<br>
&gt;<o:p></o:p></p>

</div>

<p class=MsoNormal>&gt;
------------------------------------------------------------------------<o:p></o:p></p>

<div>

<div>

<p class=MsoNormal>&gt;<br>
&gt;
------------------------------------------------------------------------------<br>
&gt; Come build with us! The BlackBerry&amp;reg; Developer Conference in SF, CA<br>
&gt; is the only developer event you need to attend this year. Jumpstart your<br>
&gt; developing skills, take BlackBerry mobile applications to market and stay<br>
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target="_blank">https://lists.sourceforge.net/lists/listinfo/gstreamer-devel</a><br>
<br>
------------------------------------------------------------------------------<br>
Come build with us! The BlackBerry&amp;reg; Developer Conference in SF, CA<br>
is the only developer event you need to attend this year. Jumpstart your<br>
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target="_blank">https://lists.sourceforge.net/lists/listinfo/gstreamer-devel</a><o:p></o:p></p>

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</div>

<p class=MsoNormal><br>
<br clear=all>
<o:p></o:p></p>

<p class=MsoNormal><br>
-- <br>
- Viraj<br>
Reality is merely an illusion, albeit a very persistent one.<o:p></o:p></p>

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