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Old 29th December 2012, 11:15 PM   #131
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I guess I'm missing something becuase I don't understand the need to iterate.

If T(f) is the target and A(f) is what you have, in the frequency domain, is

T(f) = A(f) * E(f)

Were E(f) is the error correction, and

E = T / A which is really easy to do in the frequency domain. Then take the IFFT of E(f) to get e(t), the correction impulse. T(f) can be anything you like, linear phase, minimum phase, what ever.
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Old 30th December 2012, 12:02 AM   #132
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???

Last edited by thierry38efd; 30th December 2012 at 12:11 AM.
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Old 30th December 2012, 12:07 AM   #133
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Originally Posted by Pano View Post
When I do this, I get a perfectly flat phase plot, but no amplitude! Never seen a HOLM plot with no amplitude trace. What am I doing wrong?
it's normal.
i think you're opening measurement ?
export slot A as .txt file and import directly the file.(as impulse.txt or frequency.txt)

in slot B import IR.txt from rephase.

HOLM AxB or A/B is working with .txt files (or two measurement file ) but not the mix of the two extensions.
format of measurement are not .txt.

Last edited by thierry38efd; 30th December 2012 at 12:17 AM.
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Old 30th December 2012, 12:10 AM   #134
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Originally Posted by john k... View Post
I guess I'm missing something becuase I don't understand the need to iterate. ... loadsa textbook stuff ...
The problem is that unless your FIR is zillion point, you get something like pos's top curves in #84. This is caused by the finite length, windowing bla bla.

So you iterate by tweaking the 'target' ... I mean use conjugate search bla bla to get a better approximation to your target .. the bottom curves.

Remez exchange is a hi' falutin' name for one such method with loadsa double integrals to get into IEEE Trans.

An important caveat with some methods is the resultant FIR develops nasty 'edges' at the start & end but good windowing normally deals with this.

If all this sounds airy fairy and hand waving that's cos it IS!
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Old 30th December 2012, 12:34 AM   #135
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Ok, I understand all that as we did it in the UE. I just didn't follow what was being iterated on. The iteration just goes back to what I was telling dlr about errors at low frequency and to change the target to compensate. Now I get it that Pos's code is doing it in an automated, iterative fashion.

We used 8192 taps in the UE with 48k sampling. But you really don't need a zillion points 8192 works very well down to about 100 Hz before errors are significant.


Here(link) is an example of what the UE can do for a 200Hz to 2k Hz band pass. I would not call it hand waving. It's all quite rigorous.
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Old 30th December 2012, 01:03 AM   #136
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Originally Posted by john k... View Post
We used 8192 taps in the UE with 48k sampling. But you really don't need a zillion points 8192 works very well down to about 100 Hz before errors are significant.

Here(link) is an example of what the UE can do for a 200Hz to 2k Hz band pass. I would not call it hand waving. It's all quite rigorous.
This beach bum apologises for taking a light-hearted view of FIR optimisation.

It's just that there are many ways to do this and even very crude methods work well.

When I started playing with DSP, a 200 pt FIR was SOTA so I'm in awe at the zillion (8192) pts done as a matter of course these days.

Thanks for your most interesting link. If I may ask some naive questions ..
  • Were the measurements taken in an anechoic?
  • Was the EQ in the demo, Minimum Phase?
  • What is Burst Decay? CSDs are from Fincham & Berman (1978?) which we used to call KEFplots.
  • What are the 'periods' in a Burst Decay plot.
  • Is ARTA your favourite acoustic measurement package?

Last edited by kgrlee; 30th December 2012 at 01:07 AM.
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Old 30th December 2012, 02:26 AM   #137
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Originally Posted by thierry38efd View Post
i think you're opening measurement ?
export slot A as .txt file and import directly the file.(as impulse.txt or frequency.txt)
Ca Marche! Thanks Thierry, I had forgotten about using text instead of wav files.
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Old 30th December 2012, 11:55 AM   #138
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Originally Posted by kgrlee View Post
This beach bum apologises for taking a light-hearted view of FIR optimisation.

It's just that there are many ways to do this and even very crude methods work well.

When I started playing with DSP, a 200 pt FIR was SOTA so I'm in awe at the zillion (8192) pts done as a matter of course these days.

Thanks for your most interesting link. If I may ask some naive questions ..
  • Were the measurements taken in an anechoic?
  • Was the EQ in the demo, Minimum Phase?
  • What is Burst Decay? CSDs are from Fincham & Berman (1978?) which we used to call KEFplots.
  • What are the 'periods' in a Burst Decay plot.
  • Is ARTA your favourite acoustic measurement package?
To answer your questions in order:

1) Yes, no, sort of. The measurements were taken with the mic about 6", as I recall, from the driver so any room reflections would be in the noise.

2) The way the UE works is that minimum phase EQ is applied to the driver to make the response flat over some user specified frequency range. This can be basically DC to Nyquist. Then the target for the acoustic output is defined as a series on minimum phase filters, HP, LP, Q boost/notch, shelfs, what ever you like. Then all this is combined in to a single FIR impulse which "equalizes" the driver's acoustic output to the specified target. At that point the user has the choice to linearize phase or not. If he chooses to do so, the impulse is again modified. So in the end, a single FIR impulse does it all. If phase linearization is not selected the result is minimum phase.

3) Burst decay is the response of the system to a raised cosine burst. Each burst consists of 3 to 5 cycles at the frequency being tested. So, its a X cycle sine wave burst with a cosine shaped envelope.

4) Rather than plotting on a time axis, the axis is in periods so it scales with frequency.

5) No, it is just more convenient for things like this and generates a very nice data presentation.

Going back to 2 for the moment, this is where FIR is very powerful since trying to even approximate this degree of EQ with IIR would require many stages of biquads and you would have to worry about gain staging. With FIR we end up with the impulse for a single transfer function (TF) and all that needs to be considered is if the dsp engine has sufficient head room for the max gain of that TF. Of course the question is, do you really need to EQ to the degree the UE is capable of? That's something only time and listening will tell, along with the quality of the dsp engines. Right now the dsp engine for the UE is a PC and multichannel sound card. When the project was started and I was involved we wanted to develop hardware to take it off the PC, like what miniDSP is doing. But it was cost prohibitive. As you can see, miniDSP is charging $300 for a 2 in, 2 out board that can handle 6144 taps at 48k. Something like 2 in x 8 out with 8192 taps (or more) just wasn't in the cards. Cost of the product would have been too high let alone the cost of development.
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Old 30th December 2012, 08:31 PM   #139
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Originally Posted by kgrlee View Post
It's just that there are many ways to do this and even very crude methods work well.
This is the impression I get, too. I want to do it 'text book', but what sort of difference in quality are we talking about, between an out-of-the-box crossover filter (say 4th or 8th order) with no further correction at all, and with various stages of correction e.g.

- Basic baffle step correction - I think we should notice this, but perhaps we could achieve this by pretty crude 'analogue' methods

- Phase alignment through crossover region (if it's a steepish crossover are we going to notice it?)

- Playing with delays between drivers

- Flattening of amplitude response

- Flattening of phase response

- Room correction

Basically, a three way speaker with bog standard linear phase 8th order crossover filters and judicious selection of crossover points and amplifier volume control settings, already sounds pretty damn good - and is kind of 'natural' i.e. every phase bump is probably accompanied by an amplitude bump and is caused by physical factors that our ears possibly hear past anyway. Should we be expecting a clearly better sound with the further tweaking (that a non-audiophile would notice, for example), or is it much more subtle than that?

Edit: (I know that wasn't what you meant exactly, kgrlee, but the audibility of differences between various DSP methods.)

Aside from the maths there are other variables such as mic position (obviously), degree of averaging and smoothing of measurements that the corrections are based on etc. There seems to be no definitive right answer, and one is always left with the worry that if it only sounds different, but not better, that it may, in fact, be worse. Not that it's going to stop me from doing it, obviously!

Last edited by CopperTop; 30th December 2012 at 08:57 PM.
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Old 30th December 2012, 10:13 PM   #140
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Originally Posted by john k... View Post
To answer your questions in order: ...

3) Burst decay is the response of the system to a raised cosine burst. Each burst consists of 3 to 5 cycles at the frequency being tested. So, its a X cycle sine wave burst with a cosine shaped envelope.

4) Rather than plotting on a time axis, the axis is in periods so it scales with frequency.
Thanks for this John.

It looks like Burst Decay is an extension of Analogue Loudspeaker Measurement with 3-D Display We called those PAFplots after Peter A Fryer.

IMHO, this gives the best representation of the most important speaker 'distortions' if you consider it as a 1D transmission chain.

The theoretical difference between PAFplots & Burst Decay is PAFplots use rectangular windowing.

Quote:
Going back to 2 for the moment, this is where FIR is very powerful since trying to even approximate this degree of EQ with IIR would require many stages of biquads and you would have to worry about gain staging.
Simple Arbitary IIRs shows how you can use Direct Form I IIRs instead and gain loadsa efficiency. The design of such beasts is slightly more complicated than FIRs.

After all, a FIR is just a Direct Form I IIR with feedback disabled.

Gotta get my finger out and finish the JAES version.
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John, are those stand-alone DSP boxes like the miniDSP integer only?

This is important if you are using FFT convolution cos you really need to dither at every butterfly stage.
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CopperTop, high order xovers are evil for most speakers; whether analogue, digital, Min. Phase or Linear Phase. Richard Greenfield has an AES paper on the subject. IMHO, 4th order LR is already beyond the pale. I prefer 3rd order Arthur-Smythe xovers for the usual treble xover.

Last edited by kgrlee; 30th December 2012 at 10:28 PM.
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