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MultiWay Conventional loudspeakers with crossovers 
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3rd October 2004, 06:45 PM  #11  
diyAudio Member
Join Date: Mar 2003
Location: Denmark, Copenhagen

It depends on the variables
Quote:
Please don't forget the subject for this thread: "What's your favorite crossover slope?" It could be interesting to hear what DIY people has as thier favorite crossover slope. Be free to post what you don't like, but also remember to be constructive! Hence what about posting your own favorite crossover slope... Which slope is your favorite crossover slope? And Why? Regards, Ask 

3rd October 2004, 09:08 PM  #12  
diyAudio Member
Join Date: Aug 2003
Location: Santa Cruz, California

Re: It depends on the variables
Quote:
My favorite crossover slope starts at about 6 dB/oct, then gradually increases slope until it ends up at 90 dB/oct. It's a Finite Impulse Response (FIR) filter, and the other band is derived by subtracting the filtered signal from a delayed version of the input. I use these for $DAYJOB, and they work marvelously well, especially if you try to keep the filter as short as possible. Longer filters tend to have preringing, which on axis isn't a problem since everything adds up to 1.0 if your drivers are matched at xover frequency, but off axis the preringing creeps in again. The shorter FIRs I design allow about an octave of overlap between the drivers from 1dB to 20dB, but with decent drivers that isn't really an issue. Typically it isn't what your driver does within half an octave of the crossover which kills you, it's what the things do an octave or two into their rolloffs which trash the passband. For example, a 2500 Hz FIR crossover lowpass filter could be 1dB at 1800 Hz, 6dB at 2500 Hz, 20dB at 3200 Hz, and greater than 60dB at more than 4000Hz. Its HPF dual is 1dB at 3200 Hz, 6dB at 2500 Hz, 20dB at 1800 Hz, and greater than 60dB at less than 1000 Hz. Even though the filters allow some overlap between drivers, they're essentially out of the picture less than an octave into their stopband. For woofers, you avoid exciting code breakup modes, and excursion is considerably reduced for tweeters. The implications for metalcone woofers and ribbon tweeters are obvious. You also get phase linearity as a side benefit; it literally drops out of the equation. Cheers, Francois. 

3rd October 2004, 09:43 PM  #13 
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Join Date: Apr 2003
Location: Tampere Finland Europe

Francois,
That sounds interesting, how do you calculate the coefficients for a FIR filter with that kind of slope? Mikko 
4th October 2004, 01:34 AM  #14  
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Join Date: Aug 2003
Location: Santa Cruz, California

Quote:
Word length isn't quite as critical for FIRs as it is for IIRs, but you still want 24 bit math at least for nice stopbands. That should keep you busy for a while.... Cheers, Francois. 

4th October 2004, 08:24 AM  #15 
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Join Date: May 2002
Location: Switzerland

Mikko
If you intentionally design a FIR filter with short length you will automatically end up with a filter that is less steep and that will have shorter preringing (in terms of time). Regards Charles 
4th October 2004, 10:36 AM  #16 
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Join Date: Apr 2003
Location: Tampere Finland Europe

Thanks,
If I design a FIR filter using the window method there will be ripple in stopband. If I then use the subtractive method to calculate the HPF the ripple will be in passband, right (or not, is the subtraced response mirrored vertically or horizontally after all)? Using the Remez method there will be ripple in both pass and stop band, is it any better then? Also which windowing method is best (Kaiser, Hamming or Blackmann, the other are not good I think, Kaiser was recommended somewhere, how about Chebyshev, I think it also looks nice)? 
4th October 2004, 12:16 PM  #17 
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Join Date: May 2002
Location: Switzerland

I don't remember Francois mentioning that he hasn't any ripple in the stopband !?
I for myself would definitely go for a filter derived by using the windowing method because I want a flat passband. The stopband ripple looks large on a diagram showing the response in dB but measuerd in volts or what ever it is not much. It will therefore cause a passband ripple in the derived branch that would be best expressed in millidB ! The usual way to get FIR higpass parameters using the windowing method is the subtraction of the coefficients of a lowpass from the coefficients of an allpass anyway ! Not much different than the subtractive method mentioned by Francois. I have to admit thopugh that I am a fan of analogue solutions and I would therefore go for an analog subtractive crossover anyway. Regards Charles 
4th October 2004, 01:08 PM  #18 
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Join Date: Apr 2003
Location: Tampere Finland Europe

So the ripple is there in "subtracted" high pass but it is in the stopband, not in the passband.
So I really need to use convolution only for the LPF, and the HPF output is got by subtracting the delayed input sample from the LPF output sample? In case of 71 point LP kernel it's 71 multiplications and 72 additions (71 MAC's + one add in DSP). 
4th October 2004, 06:39 PM  #19  
diyAudio Member
Join Date: Aug 2003
Location: Santa Cruz, California

Quote:
Of course the LPF stopband ripple also turns into HPF passband ripple, but I wouldn't worry about 0.01 dB ripple in either case. And yes, convolving for the LPF and subtracting to generate the HPF is exactly right. You can also do it the other way, if you want, convolving to generate the HPF and subtracting to generate the LPF. That one might be a bit easier if your tool doesn't allow you to plot response from inputted coefficients; you need to do this with the subtracted set to make sure the stopband is behaving, and HPF stopband behaviour is more of an issue if you wish to avoid blowing up tweeters. Francois. 

5th October 2004, 12:10 AM  #20  
diyAudio Member
Join Date: Sep 2002
Location: deep south

Re: It depends on the variables
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The Neville Thiele Method slopes are pretty new so I'll post a couple of links http://www.bss.co.uk/includes/produc...2_include.aspx In the first one above you'll notice they incorporate a notched response for steeper rolloff. In this next link, the graphics indicate that a 4th order NTM has steeper rolloff than 4th order LR. http://www.fmsystems.net/pdf/cutsheet/fds334t.pdf while I know that I prefer the NTM 52db slope to the LR 48 DB slope in my current application, I haven't actually done any listening tests to compare the NTM 48DB slope to the LR 48DB. Regards Ken L
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