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Michael Hellwig | 1 Jun 2007 12:08
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can a soundcard generate pure DC output?

I need this info for a project (physics) I'm working on and you bunch
here are the ones who know most about sound hardware and how to drive
it, at least of the people I could think of.

So. Given some higher-level soundcard that supports 24bit/96kHz or even
192kHz (the DACs for that seem to be easily available at analog.com), is
it possible with an existing audio driver (i.e. alsa) to tell such a
card to output a certain DC voltage (at that 24bit precision) on a channel?

Because that would be really cool in that it would mean we wouldn't have
to buy chips or evaluation bords and do a lot of development ourselves
.. using existing stuff is of course always better.

I'm just a little doubtful since of course in audio work, DC signals are
always to be avoided ...

thanks for any hints or thoughts!

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Julien Claassen | 1 Jun 2007 14:03
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Re: can a soundcard generate pure DC output?

Hi!
  I think you could tell a soundcard to generate a pure DC signal. You might 
want to use csound for this.
  There are a lot of oscilators and other sound-generators, that might do the 
task easily.
  There are probably other synthesizers or audio-editors, that could create 
such a signal, but I'm not aware of them.
  HTH.
  Kindest regards
       Julien

--------
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======== FIND MY WEB-PROJECT AT: ========
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René Bastian | 1 Jun 2007 15:35
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Re: can a soundcard generate pure DC output?

Hi,

with Python, numarray, pythoneon :

import profils as _p
import live

audio = live.Audio()

time = 15*60.0 # seconds == 15 minutes

w = _p.profil([[_p.recta, time, 1.0, 1.0]])
# in this form 1.0 == maximum voltage

audio.play(w, w) # output on 2 channels

audio.close()

You will hear nothing else than a click at the end
(when the voltage goes to zero).

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Sergei Steshenko | 1 Jun 2007 18:21
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Re: can a soundcard generate pure DC output?


--- Michael Hellwig <michael.hellwig <at> uni-ulm.de> wrote:

> I need this info for a project (physics) I'm working on and you bunch
> here are the ones who know most about sound hardware and how to drive
> it, at least of the people I could think of.
> 
> So. Given some higher-level soundcard that supports 24bit/96kHz or even
> 192kHz (the DACs for that seem to be easily available at analog.com), is
> it possible with an existing audio driver (i.e. alsa) to tell such a
> card to output a certain DC voltage (at that 24bit precision) on a channel?
> 
> Because that would be really cool in that it would mean we wouldn't have
> to buy chips or evaluation bords and do a lot of development ourselves
> .. using existing stuff is of course always better.
> 
> I'm just a little doubtful since of course in audio work, DC signals are
> always to be avoided ...
> 
> thanks for any hints or thoughts!
> 
> -------------------------------------------------------------------------
> This SF.net email is sponsored by DB2 Express
> Download DB2 Express C - the FREE version of DB2 express and take
> control of your XML. No limits. Just data. Click to get it now.
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plutek | 7 Jun 2007 23:53
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multichannel confusion

greetings!

i am trying to combine 2 stereo files into one 4-channel file, such that channels 1/2 of the first file become
channels 1/2 of the quad file, and channels 1/2 of the second file become channels 3/4 of the quad file.
ecasound is barfing on my chain-setups.

help!

here's what i've tried:

ecasound -a:1,2 -i stereofile1.aiff -a:3,4 -i stereofile2.aiff -a:1,2,3,4 -f:16,4,48000 -o
quadfile.aiff -chmove:1,1 -chmove:2,2 -chmove:3,3 -chmove:4,4

and the same thing with -chcopy.

i always get a "Invalid argument, unable to parse: '-chmove:1,1'" (or '-chcopy:1,1') error.

thanks in advance for any help!

--

-- 
.pltk.

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Julien Claassen | 8 Jun 2007 00:35
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Re: multichannel confusion

Hi!
  I'd try something like this:
ecasound -f:16,2,48000 -a:1,2 -i file1.aiff -a:3,4 -i file2.aiff 
-f:16,4,48000 -a:1,2,3,4 -o quad.aiff 
-a:1 -chmove:1,1 -a:2 -chmove:2,2 -a:3 -chmove:1,3 -chmove:2,4
  If that doesn't work, I don't know. But I think you could optimise this to 
two chains and use two -chmove options for each chain.
  Kindest regards
      Julien

--------
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======== FIND MY WEB-PROJECT AT: ========
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the Linux TextBased Studio guide
======= AND MY PERSONAL PAGES AT: =======
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plutek | 8 Jun 2007 05:18
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Re: multichannel confusion

>From: Julien Claassen <julien <at> c-lab.de>

>  I'd try something like this:
>ecasound -f:16,2,48000 -a:1,2 -i file1.aiff -a:3,4 -i file2.aiff 
>-f:16,4,48000 -a:1,2,3,4 -o quad.aiff 
>-a:1 -chmove:1,1 -a:2 -chmove:2,2 -a:3 -chmove:1,3 -chmove:2,4
>  If that doesn't work, I don't know. But I think you could optimise this to 
>two chains and use two -chmove options for each chain.

yes, you're right - we can use two stereo chains (auto-detected on input):

ecasound -a:1 -i file1.aiff -a:2 -i file2.aiff -f:16,4,48000 -a:1,2 -o quad.aiff -a:1 -chmove:1,1
-chmove:2,2 -a:2 -chmove:1,3 -chmove:2,4

this puts all the input channels in the correct output channels, but the strange thing is that all
amplitudes are reduced to half of what they were in the input files.

???

--

-- 
.pltk.

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Julien Claassen | 8 Jun 2007 10:04
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Re: multichannel confusion

Hi!
  Half the amplitude? No problem: try adding: -z:mixmode,sum at the end or 
beginning of your options. b default it uses the -z:mixmode,avg option, which 
maybe useful if you mixdown a lot of x-channel files to one x-channel file. If 
you'd add all the amps, it'd be too much. So ecasound can average the amps and 
add them then. It divides each files amplitude by the number of chains, thus 
even in the worst case you'd have no clipping.
  Kindest regards
        Julien

--------
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======== FIND MY WEB-PROJECT AT: ========
http://ltsb.sourceforge.net
the Linux TextBased Studio guide
======= AND MY PERSONAL PAGES AT: =======
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Jan Weil | 8 Jun 2007 10:08
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Re: multichannel confusion

On Thu, 2007-06-07 at 23:18 -0400, plutek <at> infinity.net wrote:
> >From: Julien Claassen <julien <at> c-lab.de>
> 
> >  I'd try something like this:
> >ecasound -f:16,2,48000 -a:1,2 -i file1.aiff -a:3,4 -i file2.aiff 
> >-f:16,4,48000 -a:1,2,3,4 -o quad.aiff 
> >-a:1 -chmove:1,1 -a:2 -chmove:2,2 -a:3 -chmove:1,3 -chmove:2,4
> >  If that doesn't work, I don't know. But I think you could optimise this to 
> >two chains and use two -chmove options for each chain.
> 
> yes, you're right - we can use two stereo chains (auto-detected on input):
> 
> ecasound -a:1 -i file1.aiff -a:2 -i file2.aiff -f:16,4,48000 -a:1,2 -o quad.aiff -a:1 -chmove:1,1
-chmove:2,2 -a:2 -chmove:1,3 -chmove:2,4
> 
> this puts all the input channels in the correct output channels, but the strange thing is that all
amplitudes are reduced to half of what they were in the input files.

Try adding '-z:mixmode,sum' to your command line. Default is
'-z:mixmode,avg' due to backwards compatibility reasons, IIRC.

See ecasound(1).

Hope that helps

Jan

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plutek | 8 Jun 2007 14:32
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Re: multichannel confusion

>From: Julien Claassen <julien <at> c-lab.de>

>  Half the amplitude? No problem: try adding: -z:mixmode,sum at the end or 
>beginning of your options. b default it uses the -z:mixmode,avg option, which 
>maybe useful if you mixdown a lot of x-channel files to one x-channel file. If 
>you'd add all the amps, it'd be too much. So ecasound can average the amps and 
>add them then. It divides each files amplitude by the number of chains, thus 
>even in the worst case you'd have no clipping.

ok, that works. it just seems counter-intuitive that chcopy is a mix operation with a default of averaging a
signal and null, in this sort of situation. however, i guess if one was chcopying a number of channels to one
destination channel, that would be necessary.

thanks for the help -- i've learned a little more about how ecasound's (or kai's) brain works!

--

-- 
.pltk.

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Gmane