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Old 20th January 2013, 11:35 AM   #321
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Originally Posted by kgrlee View Post
You are right of course. It wa a typo cos i kunt reed en rite.

I should have said, "At LF, when room dimensions are similar or less then the wavelength, the room is a 'point' and very simple EQ gives good results."

Mea maxima culpa!
I can only respond to what is written, not what is thought.
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Old 20th January 2013, 02:48 PM   #322
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Originally Posted by kgrlee View Post
Actually at LF, it is simple to get pressure response flat. The smaller the room, the easier it is.

There are 3 regions for EQ of speaker in rooms.
  • At HF, the speaker is sufficiently far from boundaries to enable EQ of the 'anechoic' response. This has been the focus of most Digital EQ efforts in the last decade or so. It is known to give good results.
  • At LF, when room dimensions are the same or greater then the wavelength, the room is a 'point' and very simple EQ gives good results.
  • The difficult part is the 'midrange' where there are significant reflections which can't be distinguished from the direct response but cause complicated response changes.

It is the 3rd region where I believe room & speaker EQ still needs a lot of research.

We know that speakers in anechoics sound terrible. So what are we trying to achieve? With respect to Bohdan and other workers in this field, I think the jury is still out on this question.

I have not done serious work on this for well over a decade so am interested in what others have found. This millenium, the DSP power available to us is far greater than the early 90's. But what should we be using it for?

Barley, how different are the EQs for the 2 channels on your final system? Can you post a curve(s)?

You mean you can get linear phase interconnects?

All the interconnects I've measured have been Minimum Phase. They satisfy one of the necessary AND sufficient criteria.
How do you measure your interconnects?

Even when room is point phase rolls off. Room acts as large closed coupler; when doing calibration work coupler is chosen with dimensions <1/6th wavelengths studied for flattening phase response.

Above sub as band pass system passes fundamental and three next terms of square wave at 15Hz resulting in recognizable square wave:

sq 15Hz.gif

Going down to 8Hz more terms are passed and square form is still fairly good:

sq 8Hz.gif

Phases of output components remain largely coherent to source stimulus.

Above is obtainable at >2m. For listener, sweet spot at these frequencies is big.

I think Bohdan and my goals are fairly clear: Improving temporal fidelity.



Integrated with two-way that does square waves <60Hz, result is phase coherent system all the way to HF cut off. Midrange difficulty averted; this of course is by thinking along lines of JohnK concerning listener and source transfer in producing measurements for correction, and manner of integrating components together.

I've no idea how long is would take to get this result with rePhase, if possible at all. Prior to using Kirkeby I've done lots of iterative tweaking with individual filters as found in rePhase, and typical of boxes such as DCX2496, flattening responses and optimizing delays, and know it is easy to blow a lot of time, and get nowhere the results with mathematical inversion.
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Old 20th January 2013, 03:21 PM   #323
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Originally Posted by Barleywater View Post
Prior to using Kirkeby I've done lots of iterative tweaking with individual filters as found in rePhase, and typical of boxes such as DCX2496, flattening responses and optimizing delays, and know it is easy to blow a lot of time, and get nowhere the results with mathematical inversion.
This I'm very interested in. In this document,

http://pcfarina.eng.unipr.it/Public/...n-workshop.pdf

the Kirkeby transform is described as:

Quote:
The IR to be inverted is FFT transformed to frequency domain:H(f) = FFT [h(f)]2) The computation of the inverse filter is done in frequency domain:Where ε(f) is a small, frequency-dependent regularization parameter3) Finally, an IFFT brings back the inverse filter to time domain:c(t) = IFFT [C(f)]
(plus a couple of formulae that haven't copied across)

It is supposedly an automatic way to invert the impulse response without excessive gain at the driver's frequency extremities. Does this kind of thing work even if the system is non-minimum phase?

(I ask, because complete inversion of the impulse response does actual magic, doesn't it? A time domain echo can be cancelled out with an appropriately-timed further impulse from the speaker, and then the first cancellation impulse is cancelled out with another impulse, and then that impulse is cancelled out with a further impulse, and so on. In a non-minimum phase system it is possible for this to go unstable if the impulses are not diminishing - I think that's the general idea.)

@Barleywater, when using the Kirkeby transform, what do you personally use for ε(f) for the various drivers you're testing?
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Old 20th January 2013, 03:38 PM   #324
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Originally Posted by Barleywater View Post
I've no idea how long is would take to get this result with rePhase, if possible at all. Prior to using Kirkeby I've done lots of iterative tweaking with individual filters as found in rePhase, and typical of boxes such as DCX2496, flattening responses and optimizing delays, and know it is easy to blow a lot of time, and get nowhere the results with mathematical inversion.
It will take some time with current version of rePhase because you will have a lot of correct/convolve/measure iterations to do (or simpler correct/convolve if using the C=A*B thingy in HOLM or REW), but it will soon be easier when loading measurements will be available in a next version (soon to be released... ).

If doing a measurement inversion is your goal then DRC-FIR is probably the best tool to use (with a given target curve) as it will automatically take care of a lot of potential pitfalls related to frequency variable windowing and correction limits (or PORC for MP correction and then textbook FIR filters...).
But even with these precautions this kind of automated correction requires a lot of care and knowledge in the measurement procedure.
In all cases, inversion based on a single measurement will likely not be valid for all frequencies, nor for all listening positions, and can cause problems greater than the ones it corrects in some situations...

I am sure you know how to do it properly of course (even without DRC-FIR) but for most users I find that these automated approaches often promise a lot of things but give disappointing results to the unsuspecting user.
In this regard I find manual corrections to be a much more coherent and safer approach for most situations. It also forces the user to really ponder each correction and know exactly what he is doing, and I think it is a good thing...

Anyway, rePhase is focused on manual corrections, so these inversion discussion are quite OT
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Old 20th January 2013, 03:55 PM   #325
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Originally Posted by pos View Post
In all cases, inversion based on a single measurement will likely not be valid for all frequencies, nor for all listening positions, and can cause problems greater than the ones it corrects in some situations...
Could you elaborate on that further? I would be most interested in correcting a driver from a nearfield measurement, I think. Would this be so close to a minimum phase system, that complete impulse response inversion would be equivalent to hand-tailored rePhase EQ, anyway?

Yes, DRC seems to be the way for room correction (but I have never yet got good results with it - my own shortcomings for sure, not the program's).
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Old 20th January 2013, 04:19 PM   #326
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Just to stay a bit longer OT. I never got good filters from DRC (measurements problem?) but Audiolense always worked for me even if it is completely automated. A good software can do it.
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Old 20th January 2013, 05:58 PM   #327
pos is offline pos  Europe
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Originally Posted by CopperTop View Post
Could you elaborate on that further? I would be most interested in correcting a driver from a nearfield measurement, I think. Would this be so close to a minimum phase system, that complete impulse response inversion would be equivalent to hand-tailored rePhase EQ, anyway?
For a narrow bands it should be quite easy, as it would just be a matter of finding the good measurement technique (including impulse post processing) and distance.
But some frequency bands are more difficult than others like midbass, where room starts to dominate, and baffle also typically starts not to be large enough to restrict the radiation to <180...
In such situations the only practical solution is to average measurements taken at several positions in the listening "window", as anechoic measurement would not really help even if they could be achieved.
Low frequencies are much easier (close mic), but then it is quite meaningless in a room anyway...
The high frequency driver can be accurately dealt with with an anechoic measurement (windowed), but then if it is not a directive device (and even then) you will have to deal with diffraction artifact that are not stable with measurement position.

A brutal impulse response inversion, without any processing done to the impulse, would be the worst case.
When you see perfectly linear amplitude (and phase) responses you know that it was measured at exactly the same position as the correction was calculated from. Move the mic from XX centimeters and guess what happen...

Quote:
Yes, DRC seems to be the way for room correction (but I have never yet got good results with it - my own shortcomings for sure, not the program's).
Room is part of the system anyway, and automated frequency-dependent windows are the best way to deal with its effect.
Jean-Luc Ohl, the author of Align2 (Jlo on this forum) is working on a multi-point version of Align2 that will use DRC-FIR. I bet it will give very nice results.

Last edited by pos; 20th January 2013 at 06:01 PM.
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Old 21st January 2013, 12:04 AM   #328
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Originally Posted by pos View Post
A brutal impulse response inversion, without any processing done to the impulse, would be the worst case.
When you see perfectly linear amplitude (and phase) responses you know that it was measured at exactly the same position as the correction was calculated from. Move the mic from XX centimeters and guess what happen...
... In this regard I find manual corrections to be a much more coherent and safer approach for most situations. It also forces the user to really ponder each correction and know exactly what he is doing, and I think it is a good thing...
I second all that

pos, you are right that the midrange is the most difficult and I'm still not sure what is the best strategy for it. Averaging over an area is somewhat naive but I haven't got a better plan.

What I DO know is that you don't want to achieve 'anechoic' results.

Kirkeby

I've never used Kirkeby regularisation, Angelo Farina's favourite method, for speakers or rooms. But I have a lot of experience of it for microphones.

It's really a method to avoid trying to EQ 30dB dips in response. I was at the previous millenium IoA conference where it was first proposed.

Prof. Farina has an AGM/DPA4 soundfield microphone. He has never been able to get good EQ for it using Kirkeby regularisation. (A soundfield microphone has substantial EQ). In 2008, he visited me in Cooktown and brought his DPA4.

I used old fashion techniques, slightly updated with 21st century digits to devise an IIR EQ for him and for the first time, he got good sound from the mike.

He also thinks IIRs are evil while I think FIRs are evil. So I devised some evil FIRs for him based on my evil IIRs so everyone was happy.
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Coppertop, you are right about the cancelling of echoes. I first did this in the late 70's Use of Tapped Delay Lines in Speaker Work well before supa dupa digits were easily available.

The stability criteria is a bit complicated. IIRC, the best explanation is "Invertibility of a Room Impulse Response - Neely" or something. It's not in AES.
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Barley, I'm still interested in the Left & Right EQs for your system. Can you post a curve or two?

Last edited by kgrlee; 21st January 2013 at 12:24 AM.
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Old 21st January 2013, 12:21 AM   #329
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Originally Posted by CopperTop View Post
This I'm very interested in. In this document,

http://pcfarina.eng.unipr.it/Public/...n-workshop.pdf

the Kirkeby transform is described as:



(plus a couple of formulae that haven't copied across)

It is supposedly an automatic way to invert the impulse response without excessive gain at the driver's frequency extremities. Does this kind of thing work even if the system is non-minimum phase?

(I ask, because complete inversion of the impulse response does actual magic, doesn't it? A time domain echo can be cancelled out with an appropriately-timed further impulse from the speaker, and then the first cancellation impulse is cancelled out with another impulse, and then that impulse is cancelled out with a further impulse, and so on. In a non-minimum phase system it is possible for this to go unstable if the impulses are not diminishing - I think that's the general idea.)

@Barleywater, when using the Kirkeby transform, what do you personally use for ε(f) for the various drivers you're testing?
It is best to experiment and learn.

Pos: Sourceforge DRC is clogged up with author's prescriptions for all the over bandied words concerning linear phase: pre-ringinging. Years ago I started using computer as many; looking to assist in making passive crossovers behave better. Along the way Rod Elliot, Linkwitz, and Smith's dspguide.com got me going with DSP, after having already explored the more technical side of Cool Edit Pro to point of using own band pass bursts for speaker measurements. This lead to DCX2496, and continued study. I waded into muck of DRC, and already knew enough that the various published results, and all the tweaks and iterations of practitioners was fishy. I finally brushed up with Farina's various papers, and got to bending head around swept sine measurements, and saw Kirkeby in:

http://pcfarina.eng.unipr.it/Public/...226-AES122.pdf

From here I explored Farina's Auro plugins, and also based on Farina's descriptions built swept sine pairs for measurements using his time domain approach. This always leads to measurement system with windowed sinc function representing the bandwidth, and pre-ringing as viewed in time domain. Faina's plugs also provided my first access to MLS, and puzzled over how it does DC to Nyquist.

DRC package uses measure like Farina's, and authors awareness failed to address this. They are not alone; other packages use this. Holm, Audiolense, REW, and others use better method, by building swept sine inverse in frequency domain, and using FFT to get time domain, but this is like Kirkeby.

Examination of DRC source code show a section with Kirkeby name. It's the nuts and bolts behind the curtain. Author's insisting on listening position solutions without full understanding of reciprocity lead to lots of kludges to get around krglee's fore mentioned "midrange" problem.

The whole correct the listing position thing lead to variety of "frequency dependent windowing" approaches, making some ease of application, and lots of sacrifice to potential.

Bodzio UE describes merging nearfield far field measurements to get around the midrange problem, this is JohnK's baby. Still, in terms or reciprocity something doesn't jibe quite right in my mind.

And you point to it again here with near field for woofer, and gated far field for >200Hz.

I go other way, near as possible for driver array integration of mains, and listening position for woofer, because as mentioned in recent posts, room becomes like point with decreasing frequency, and near field/far field distinction becomes lost with the narrowing phase margin.

All modes for room have origin at speaker. When measurement microphone placed outside of basis mode about speaker, locally applicable solution is obtained, often with "head in vise sweet spot". With microphone within 1/4 wave of speaker for all frequencies of interest, inverse is correct solution for modal peaks of given frequency throughout room... With microphone within inches of speaker, sonogram still shows distinct reflections withing room, and often from within speaker enclosure.

Mantra: Linear time invariant system is periodic and has an inverse.

Kirkeby may be used to generate inverse for swept sine measurements that lets swept sine effectively measure DC to Nyquist. Kirkeby may be applied to MLS to generate inverse too. Extremely powerful....solution of millions of floating point samples. 128bit encryption pales in comparison.

Convolve a piece of music with MLS signal and it sounds like noise. Convolve again with time reversed MLS, and out comes music, with lead in and lead out, not magic, information science.

In a transmission system with periodic IR, if it is sufficiently linear, then the mantra applies. For sampled system with given time bandwidth, all samples define arbitrary impulse response, thus possibility of IR recovery of same result using swept sine, MLS, and potentially any broad band signal.
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Old 21st January 2013, 12:32 AM   #330
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Quote:
Originally Posted by kgrlee View Post
I second all that

pos, you are right that the midrange is the most difficult and I'm still not sure what is the best strategy for it. Averaging over an area is somewhat naive but I haven't got a better plan.

What I DO know is that you don't want to achieve 'anechoic' results.

Kirkeby

I've never used Kirkeby regularisation, Angelo Farina's favourite method, for speakers or rooms. But I have a lot of experience of it for microphones.

It's really a method to avoid trying to EQ 30dB dips in response. I was at the previous millenium IoA conference where it was first proposed.

Prof. Farina has an AGM/DPA4 soundfield microphone. He has never been able to get good EQ for it using Kirkeby regularisation. (A soundfield microphone has substantial EQ). In 2008, he visited me in Cooktown and brought his DPA4.

I used old fashion techniques, slightly updated with 21st century digits to devise an IIR EQ for him and for the first time, he got good sound from the mike.

He also thinks IIRs are evil while I think FIRs are evil. So I devised some evil FIRs for him based on my evil IIRs so everyone was happy.
________________

Coppertop, you are right about the cancelling of echoes. I first did this in the late 70's Use of Tapped Delay Lines in Speaker Work well before supa dupa digits were available.

The stability criteria is a bit complicated. IIRC, the best explanation is "Invertibility of a Room Impulse Response - Neely" or something. It's not in AES.
________________

Barley, I'm still interested in the Left & Right EQs for your system. Can you post a curve or two?

My room, as most has issues, and to keep perspective, I use same correction for left and right, and usually sum for sub. My goal is getting original signals into room, and letting it behave as such. Good room placement of corrected speaker yields excellent sound stage, and spectral content of reflections correlates well, leaving mind to concentrate on direct sound.
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