[music-dsp] Iterative decomposition of an arbitrary frequency response by biquad IIR

2014-03-03 Thread Uli Brueggemann
Hello music-dsp,

I like to decompose an arbitrary frequency response by biquads. So I'm
searching for an algorithm or paper on how to run an iterative
decomposition. In my imagination it should be possible to
a) find a first set of biquad parameters with a best fit frequency response
in comparison to the given response
b) create a IIR filter with inverse gain
c) apply the filter to the given response to get a new one
d) repeat a)-d) until some end criteria is reached

a) should include the different filter types like peaking filter, lowpass,
highpass, lowshelf, highshelf...

Is there any good information for such an approach around? Is there a
downside for such an approach?

Uli
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Re: [music-dsp] Iterative decomposition of an arbitrary frequency response by biquad IIR

2014-03-03 Thread Stefan Sullivan
For matching just the magnitude response, MATLAB has a built-in function for it:
http://www.mathworks.com/help/signal/ref/yulewalk.html

And maybehaps some more parametric modelling techniques will be useful for you
http://www.mathworks.com/help/signal/ug/parametric-modeling.html

-Stefan

On Mon, Mar 3, 2014 at 12:00 PM, Uli Brueggemann
uli.brueggem...@gmail.com wrote:
 Hello music-dsp,

 I like to decompose an arbitrary frequency response by biquads. So I'm
 searching for an algorithm or paper on how to run an iterative
 decomposition. In my imagination it should be possible to
 a) find a first set of biquad parameters with a best fit frequency response
 in comparison to the given response
 b) create a IIR filter with inverse gain
 c) apply the filter to the given response to get a new one
 d) repeat a)-d) until some end criteria is reached

 a) should include the different filter types like peaking filter, lowpass,
 highpass, lowshelf, highshelf...

 Is there any good information for such an approach around? Is there a
 downside for such an approach?

 Uli
 --
 dupswapdrop -- the music-dsp mailing list and website:
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 links
 http://music.columbia.edu/cmc/music-dsp
 http://music.columbia.edu/mailman/listinfo/music-dsp
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[music-dsp] R: Iterative decomposition of an arbitrary frequency response by biquad IIR

2014-03-03 Thread Marco Lo Monaco
Thanks Peter, that sounds interesting, have you ever tried DE on filter
estimation?

M.

-Messaggio originale-
Da: music-dsp-boun...@music.columbia.edu
[mailto:music-dsp-boun...@music.columbia.edu] Per conto di Peter S
Inviato: lunedì 3 marzo 2014 13:04
A: A discussion list for music-related DSP
Oggetto: Re: [music-dsp] Iterative decomposition of an arbitrary frequency
response by biquad IIR

On 03/03/2014, Uli Brueggemann uli.brueggem...@gmail.com wrote:
 I like to decompose an arbitrary frequency response by biquads. So I'm 
 searching for an algorithm or paper on how to run an iterative 
 decomposition.
...
 Is there any good information for such an approach around? Is there a 
 downside for such an approach?

I guess you can always use a general iterative problem solving approach like
differential evolution if you apply that to audio
filters:

https://www.google.com/search?q=differential+evolution

Such genetic algorithms are great to iteratively find a 'best fit' set of
parameters.

- Peter
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[music-dsp] R: Iterative decomposition of an arbitrary frequency response by biquad IIR

2014-03-03 Thread Marco Lo Monaco
Stefan/Uli
I use Scilab and there is not so much as in MAtlab for filter estimation.
My experience with Scilab's invfreqz (which is based on the Levi paper I
think) was always deluding for practical analog filter identification, but I
bet I was unlucky or I didnt get the point of how to use it effectively.
What I don’t personally like in these methods is that they also need a
weight vector on the data set, which adds a new degree of freedom that you
must guess (not only the order).

Let me know how is/was your experience then...

Marco

-Messaggio originale-
Da: music-dsp-boun...@music.columbia.edu
[mailto:music-dsp-boun...@music.columbia.edu] Per conto di Stefan Sullivan
Inviato: lunedì 3 marzo 2014 12:17
A: A discussion list for music-related DSP
Oggetto: Re: [music-dsp] Iterative decomposition of an arbitrary frequency
response by biquad IIR

For matching just the magnitude response, MATLAB has a built-in function for
it:
http://www.mathworks.com/help/signal/ref/yulewalk.html

And maybehaps some more parametric modelling techniques will be useful for
you http://www.mathworks.com/help/signal/ug/parametric-modeling.html

-Stefan

On Mon, Mar 3, 2014 at 12:00 PM, Uli Brueggemann uli.brueggem...@gmail.com
wrote:
 Hello music-dsp,

 I like to decompose an arbitrary frequency response by biquads. So I'm 
 searching for an algorithm or paper on how to run an iterative 
 decomposition. In my imagination it should be possible to
 a) find a first set of biquad parameters with a best fit frequency 
 response in comparison to the given response
 b) create a IIR filter with inverse gain
 c) apply the filter to the given response to get a new one
 d) repeat a)-d) until some end criteria is reached

 a) should include the different filter types like peaking filter, 
 lowpass, highpass, lowshelf, highshelf...

 Is there any good information for such an approach around? Is there a 
 downside for such an approach?

 Uli
 --
 dupswapdrop -- the music-dsp mailing list and website:
 subscription info, FAQ, source code archive, list archive, book 
 reviews, dsp links http://music.columbia.edu/cmc/music-dsp
 http://music.columbia.edu/mailman/listinfo/music-dsp
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Re: [music-dsp] R: Iterative decomposition of an arbitrary frequency response by biquad IIR

2014-03-03 Thread Stefan Sullivan
Marco,

No, I've never actually had much luck with those kinds of filter
design approaches. Matlab is not my favorite tool in the world. I
usually find it a bit annoying that matlab seems more interested in
matching some sort of brick-wall filter than designing a good-sounding
audio filter. Although, this one doesn't seem to have weights, so it
may be a little easier to use. That said, for OP's question, I think
the paper referenced in that page is at least a step towards a good
(better than matlab's?) solution.

-Stefan

On Mon, Mar 3, 2014 at 4:24 PM, Marco Lo Monaco
marco.lomon...@teletu.it wrote:
 Stefan/Uli
 I use Scilab and there is not so much as in MAtlab for filter estimation.
 My experience with Scilab's invfreqz (which is based on the Levi paper I
 think) was always deluding for practical analog filter identification, but I
 bet I was unlucky or I didnt get the point of how to use it effectively.
 What I don't personally like in these methods is that they also need a
 weight vector on the data set, which adds a new degree of freedom that you
 must guess (not only the order).

 Let me know how is/was your experience then...

 Marco

 -Messaggio originale-
 Da: music-dsp-boun...@music.columbia.edu
 [mailto:music-dsp-boun...@music.columbia.edu] Per conto di Stefan Sullivan
 Inviato: lunedì 3 marzo 2014 12:17
 A: A discussion list for music-related DSP
 Oggetto: Re: [music-dsp] Iterative decomposition of an arbitrary frequency
 response by biquad IIR

 For matching just the magnitude response, MATLAB has a built-in function for
 it:
 http://www.mathworks.com/help/signal/ref/yulewalk.html

 And maybehaps some more parametric modelling techniques will be useful for
 you http://www.mathworks.com/help/signal/ug/parametric-modeling.html

 -Stefan

 On Mon, Mar 3, 2014 at 12:00 PM, Uli Brueggemann uli.brueggem...@gmail.com
 wrote:
 Hello music-dsp,

 I like to decompose an arbitrary frequency response by biquads. So I'm
 searching for an algorithm or paper on how to run an iterative
 decomposition. In my imagination it should be possible to
 a) find a first set of biquad parameters with a best fit frequency
 response in comparison to the given response
 b) create a IIR filter with inverse gain
 c) apply the filter to the given response to get a new one
 d) repeat a)-d) until some end criteria is reached

 a) should include the different filter types like peaking filter,
 lowpass, highpass, lowshelf, highshelf...

 Is there any good information for such an approach around? Is there a
 downside for such an approach?

 Uli
 --
 dupswapdrop -- the music-dsp mailing list and website:
 subscription info, FAQ, source code archive, list archive, book
 reviews, dsp links http://music.columbia.edu/cmc/music-dsp
 http://music.columbia.edu/mailman/listinfo/music-dsp
 --
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