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22nd April 2013, 02:29 PM  #971 
diyAudio Member
Join Date: Aug 2003
Location: Santa Cruz, California

Why a FIR block for driver compensation, as opposed to a nice set of biquads?

22nd April 2013, 03:28 PM  #972  
diyAudio Member
Join Date: Feb 2013

Quote:
The figures I remember are 8500 FIR taps or 120 biquads. 120 biquads equals 480 poles/zeros. This might be totally wrong of course and I'm happy to stand corrected! /K 

22nd April 2013, 06:42 PM  #973  
diyAudio Member
Join Date: Aug 2003
Location: Santa Cruz, California

Quote:
Now the FIR can correct an entire spectrum within its limitations, of course, whereas you'd need an IIR for each anomaly, but the point remains. There's a more subtle point  speakers (NOT rooms) are minimumphase systems in the physical world, so their impulse responses will be infinite (within the limitations of measurement noise), whereas FIR correction will truncate the correction impulse so the two won't match exactly in the time domain. You can get arbitrarily close with longer FIRs, but then we're back to the starting point in the above paragraph. BTW, a biquad would have two poles (denominator) and two zeros (numerator). 

23rd April 2013, 08:12 AM  #974 
diyAudio Member
Join Date: Feb 2013

Good points, DSP_Geek!
Right now I'm using only biquads in my optimization routine. I have a desired frequency response and then I generate an amplitude response from sets of 5 coefficients: H(z) = ( b0 + b1*z1 + b2*z2 )/( 1 + a1*z1 + a2*z2 ) where z1 = z^(1) and so on. The filter is a cascade of biquads so if I choose 4 biquads I will have 20 coefficients. Then I compute an error function in the LSQ sense and minimize this error. Results are good, I can get great accuracy. As you noted speakers are minimumphase so therefore I do not care about the phase response. It is simply the Hilbert transform of the amplitude response. See Wikipedia Of course there is a delay between the elements as the physical distance between the phase centers are different but this will be compensated with a delay in the DSP. My reasoning behind the FIR block was to keep the zeros inside the unit circle to still have a minimum phase system but give the optimization routine some more degrees of freedom. I have not tested this so can't say if it will work. Since the code is already written it's pretty easy to test though. In short: Poles inside unit circle for stability. Two per biquad. Zeros inside unit circle for minimum delay. Two per biquad plus one per FIR tap. Added in conjugate pairs to keep system real. As for digital room correction I will abstain from this and use bass management instead to avoid exciting room modes. A couple of notches will suffice I guess. This is a much later experiment though as I have to actually finish my speakers first in order to put them in a room... /K 
23rd April 2013, 05:54 PM  #975  
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Join Date: Nov 2005
Location: Norway

Quote:
That was good news. So now you just have to wait for a Najda software update For now I have just pulled out the usb and started listening best, Paal 

23rd April 2013, 06:20 PM  #976  
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Join Date: Apr 2010
Location: Luxembourg

Quote:
How do you sample the desired frequency response? On a log scale? Otherwise I suppose it's an iterative process (?). So how do you move poles and zeros? Are you trying all possible combinations and keep the best one, or are you gradually approaching the optimal placement based on some rule? Quote:
Quote:
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Najda DSP 

23rd April 2013, 11:04 PM  #977 
diyAudio Member
Join Date: Apr 2010
Location: Luxembourg

I'm looking now at custom biquads.
Can someone post here a set of coefficients so that I can see how the formatting is looking like? What I have in mind for now is as exemplified below: Code:
N ; Number of biquads b0 ; First biquad coef set b1 b2 a1 a2 b0 ; Second biquad ... ... b0 ; Nth biquad b1 b2 a1 a2 Would that be compatible with the files you're otherwise dealing with?
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Najda DSP 
24th April 2013, 01:54 AM  #978 
diyAudio Member
Join Date: Aug 2006
Location: Montreal

Hi Nick,
Glad your working on custom biquad, But if I can make a little sugestion on the interface of your software, It would be nice if you can undock the graphs windows so that you can fine tune more easily without having to change the tabs back and forwards... Thank you 
24th April 2013, 06:09 AM  #979  
diyAudio Member
Join Date: Feb 2013

Quote:
I use a genetic algorithm: Wikipedia /K 

24th April 2013, 06:45 AM  #980 
diyAudio Member
Join Date: Sep 2011
Location: Luxembourg

So now DSP stands for DNA Sound Processing. You and CharpaK are running for the Nobel Prize ? (;)

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