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Re: audio problems
From: |
Paul Kienzle |
Subject: |
Re: audio problems |
Date: |
Tue, 11 Dec 2001 11:10:22 -0500 |
When you are downsampling, you will want to low-pass filter
your input first. E.g., to downsample by a factor of q
[b, a] = cheby1(n, 0.05, 1/q);
y = filter(b,a,x);
y = y(1:q:length(x));
The function cheby1 is available from octave-forge at octave.sf.net
Also, /dev/dsp may be expecting mulaw encoded data.
In that case you will need to do playaudio(lin2mu(y)).
Paul Kienzle
address@hidden
On Tue, Dec 11, 2001 at 06:50:55PM +1100, Matthew Flax wrote:
> Hi there,
>
> I am just writing to let you know that the audio output is alwayse @ 8
> bit, 8 KHz
> The way to use this is to make sure your signal (x) is :
> %# scale x to max 8 bit levels
> x=x/max(x)*128;
> %# resample x to the correct sampling freq.
> fs=44100
> playaudio(x(1:(fs/x):length(x)))
>
> that should work I believe
> --
> Matt
>
> For electronic musicians ...
> Vector Bass : http://mffmvectorbass.sourceforge.net/
> For developers ...
> TimeScale Audio Mod : http://mffmtimescale.sourceforge.net/
> Multimedia Time Code : http://mffmtimecode.sourceforge.net/
> 3D Audio Library : http://mffm3daudiolib.sourceforge.net/
>
>
>
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>
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Octave is freely available under the terms of the GNU GPL.
Octave's home on the web: http://www.octave.org
How to fund new projects: http://www.octave.org/funding.html
Subscription information: http://www.octave.org/archive.html
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