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Old 13th November 2003, 08:04 AM   #11
Henckel is offline Henckel  Denmark
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Hello

The request to include more options in the cross over section including channel volumen, equalizer and delay has already been relayed to the author.

However to some extend this is not so important since the application is to be used togehter with DRC which also will compensate for frequency and time deviations in relation to the total response from the speaker.
So just use the cross over functiosn available to do a coarse ajustment of the system - DRC will do the rest.


The use of FIR filters and "phase coherent" filters has also been relayed to the author.

However i great doubt that a Phase coherent FIR filter will be of any use apart for a theoretical study. The reason for this is that a phase coherent filter will have an symmetrical Impulse response and thus give rise to pre-ringing off axsis especially if steep filters are deployed.

I belive that the best compromise is between phase coherent and minimum phase filters - but since FIR filters can accomodate both. However a good application for providing the filter coeeficients are not available to my knowledge - nor a multi channel convolver for the Windows platform
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Old 13th November 2003, 11:59 AM   #12
dc is offline dc
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jwb,

what program do you use to implement the filters you design in MATLAB? brutefir?

btw, my understanding is the same is yours -- phase coherent = linear phase = constant group delay across the frequency band. So, while the original signal is delayed by the filter, the phase of the original signal is preserved.

ASIOXO builds on another program called DRC, which it uses for room correction. DRC works in the time and frequency domains, correcting phase and impulse response. IIRC, ASIOXO runs the signal through DRC first, and then through the IIR xo filters. I'm not a DSP engineer (though I'd like to play one in my spare time), but, my understanding is that correcting the impulse response does not necessarily lead to linear phase, particularly if you're correcting the impulse response first, and then running the signal through an IIR filter, which will distort the group delay.

There are some good papers available on this topic in the AES Loudspeakers Anthology Vols. 3 & 4, written by Vanderkooy & Lipshitz. There is also some information available on the Web (Google "Vanderkooy filter").
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Old 13th November 2003, 03:17 PM   #13
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Quote:
Originally posted by Henckel
Hello

However i great doubt that a Phase coherent FIR filter will be of any use apart for a theoretical study. The reason for this is that a phase coherent filter will have an symmetrical Impulse response and thus give rise to pre-ringing off axsis especially if steep filters are deployed.
I'm not sure what you mean by "phase coherent", the objective of the filtering should be that the system has linear magnitude and "linear phase". If in addition all the drivers are "in phase" you minimize off axis lobing problems. You can certainly do this without any ringing (I'm doing it) and I love it.
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Old 13th November 2003, 03:21 PM   #14
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Does anybody know of a plug-in for Windows Media Player 9 that implements arbitrary IIR/FIR crossovers?

I'm working on it now and have gotten as far as filtering a stereo input signal and outputing the filtered stereo signal, but am working towards an arbitrary number of input and output signals, eg. 6 in 10 out.

However, if someone has already done this I'd really like to save my time.

Thanks, John
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Old 13th November 2003, 04:59 PM   #15
Thunau is offline Thunau  United States
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Quote:
Originally posted by jwb


Simply choose your crossover frequency and find two same-order FIR filters, one high-pass and one low-pass, with 3dB corners at that frequency. As I said, I let the Matlab Filter Deisng Toolbox do the heavy lifting for me. I don't know how to pick filter coefficients. And, not having used this particular program, I really don't know how you would feed the coefficients into it.
.
Yes this would work with two or more perfect drivers. But, loudspeakers are bandpass devices, mostly non- minimum phase especially in their enclosures. Their phase rolls just like that of IIR filters. It is possible to compensate for it and arrive at a transient perfect loudspeaker using delay and IIR filters (or good old analog components for that matter).
So, just because you apply a textbook FIR filter to your drivers, it doesn't mean you end up with a transient perfect loudspeaker.
Or one that has flat frequency response. You have to know what the individual drivers are doing and take into account when designing a crossover.

Quote:
Originally posted by Henckel
Hello

The request to include more options in the cross over section including channel volumen, equalizer and delay has already been relayed to the author.

However to some extend this is not so important since the application is to be used togehter with DRC which also will compensate for frequency and time deviations in relation to the total response from the speaker.
So just use the cross over functiosn available to do a coarse ajustment of the system - DRC will do the rest.
Yes and no, You can measure the impulse response with Asio XO in place and correct for frequency response errors, but that could be just an excersise in equalizing a potentially bad loudspeaker.
You can't turn a Bose into an Avalon by applying one FIR filter.
There are consequences of sloppy crossover design. Lobing and power response are what can make or break a speaker with flat frequency response on axis.
You should have the tools to make the best conventional loudspeaker out of the components you have and then use the room correction algorithm to make that speaker shine.

Quote:
Originally posted by Henckel
The use of FIR filters and "phase coherent" filters has also been relayed to the author.

However i great doubt that a Phase coherent FIR filter will be of any use apart for a theoretical study. The reason for this is that a phase coherent filter will have an symmetrical Impulse response and thus give rise to pre-ringing off axsis especially if steep filters are deployed.

I belive that the best compromise is between phase coherent and minimum phase filters - but since FIR filters can accomodate both. However a good application for providing the filter coeeficients are not available to my knowledge - nor a multi channel convolver for the Windows platform
You are right about the FIR filters. Unless the user has the right tools to design crossovers using them it's pointless and a waiste of DSP horse power to offer them.

The author should stick with IIR filters for individual drivers. It would be nice if some kind of high precission math was used (like 32 bit FP or straight 48 bit computting with dither as opposed to truncation for going back to 16 bit). A lot of sound cards accept 24 bit input and have 24 bit converters. That should be an output option as well.
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Old 13th November 2003, 05:18 PM   #16
jwb is offline jwb  United States
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Quote:
Originally posted by Thunau


Yes this would work with two or more perfect drivers. But, loudspeakers are bandpass devices, mostly non- minimum phase especially in their enclosures. Their phase rolls just like that of IIR filters. It is possible to compensate for it and arrive at a transient perfect loudspeaker using delay and IIR filters (or good old analog components for that matter).
Ah, I see your point better now. To do that job properly you'd need a calibrated microphone, I would think. Luckily with software it's easy enough to experiment with different filters.
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