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Old 25th January 2013, 12:25 AM   #341
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Default Kirkeby

Quote:
Now, back to Kirkeby. I do not own a device, that is based on this algorithm, therefore I am struggling to understand the implications of their method.
  • When you create a system (loudspeaker+Kirkeby equalizer) to equalize the loudspeaker between F1 and F2 – is the system between F1 and F2 “linear phase”, or “minimum phase”.
  • As I understand, such system has transition regions below F1 and above F2. So my next question is: within the transition regions, is the system “minimum-phase”, or “linear-phase” or “undetermined-phase”.
  • Do the transition regions fall into the audio bandwidth for any driver?. If yes are they audible?. How do you deal with this issue?.
  • Can you run Kirkeby equalizer in minimum-phase mode, or it imposes straight away linear-phase mode?.
Can anybody answer these questions?.
Kirkeby is simply an elegant way to avoid EQing -40dB dips. It was developed for Room + Speaker EQ so has to deal with HUGE dips.

You measure the IR in a room and take the FFT. You can invert (in freq domain) this directly but then the -40dB dips translate to +40dB peaks in the EQ.

Inverting a measured (complex freq) response is the most naive way to use supa dupa digits and gives very poor results including blown speakers & amps.

What Kirkeby does is to add a 'Re' error band to the original FFT so they are no -40dB dips to EQ. You have to excuse Prof. Farina for obfuscating this in loadsa maths but that's all Kirkeby does.

Kirkeby is just a simple way to get less naive & yucky results. It usually results in sorta Linear Phase for the 'pass band'. The smaller the error band, the closer it is to LP but it doesn't really get there. It NEVER gets close to Minimum Phase except for trivial examples.

The main problem with Kirkeby isn't with the technique itself. But that it encourages people to stop thinking about what they are actually measuring. Even Angelo was/is guilty of that.

One BIG problem is that ALL measurements have NOISE. Using a straight inversion technique like Kirkeby means this measurement noise is convolved with your signal. When you apply this to eg a sine wave, you get noise sidebands.

Then there are all the peaks & dips from poor measurement technique rather than your speaker or microphone. I can claim to have taught Prof Farina something about that

pos is right that it's important to think carefully when you EQ. In ALL EQ, what you DON'T EQ is as, if not more important, than what you EQ.
________________

Barley, how do you measure the "anechoic" response that you EQ? In a room?
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Old 25th January 2013, 12:46 AM   #342
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Hi kgrlee,

Thank you for your response.

I think we can safely leave equalizing the -40dB dips and also measurement noise out of this discussion.

I am specifically interested in what characteristics are exhibited by equalized system. For simplicity, we can call it loudspeaker+equalizer (no crossover).

Your comment is:

“..It usually results in sorta Linear Phase for the 'pass band'. The smaller the error band, the closer it is to LP but it doesn't really get there. It NEVER gets close to Minimum Phase except for trivial examples….”

OK, I’ll take this for an answer, due to lack of others. So, my understanding is, that the Kirkeby equalizer does not offer minimum-phase equalization.

Secondly, I am specifically interested in transition regions, their characteristics and the consequences of introducing them. You have not commented on these?.

Best Regards,
Bohdan
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Old 25th January 2013, 01:52 AM   #343
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Quote:
Originally Posted by john k... View Post
POS, while I understand what you are saying about equalizing flat based on a single measurement, what happens off axis is largely dependent on the design of the speaker. If constant directivity is maintained off axis then the single point reference for eq is actually pretty good.
I am not talking about the directivity behavior per se, but on the artifacts you will get with a single point measurement (diffraction for example).

Quote:
Originally Posted by john k... View Post
If you don't want to EQ every small ripple you can start with a smoothed response, like 1/3 octave smoothing applied to the reference measurement.

In any event it is still possible to take any number of measurements and average them with the UE. The danger with that approach is that such an averaged response can actually result in a reference response that is worse than the measurement at any of the points contributing to the average. This is because you are not dealing with a simple average of amplitude but of a vector sum. This becomes very sensitive at higher frequency where the averaged response may exhibit comb-filtering effects.
Assuming the offset is always made correct averaging several measurement should work (HOLM for example is able to use fractional offsets for the impulse, and it automated offset detection is quite reproducible).
But anyway, spatial averaging (multiple measurements) or frequency averaging (smoothing) is not the solution as it will only smooth the problems and blur their effects, as you pointed out. So ultimately even that solution is not optimal, and automated correction base on one measurement (even averaged) is not that good a solution. And it is also not as simple as it was advertised anymore ("measure, set a target curve, and you are done").

On the other hand, taking multiple measurements and looking at the effect of a given (manual) correction on each of them alternatively (and adjusting it) looks like a good solution to me, as you are then able to really ponder every and each correction you make.
It takes some time to do, but at least there is no surprise at the end. And in fact it is not that much longer to do than a proper averaging/smoothing strategy...

For now to do that with rePhase you have to use an external convolution engine of the measurement (using HOLM or REW for example), but I will soon implement it in rePhase. It takes some time because I want to do it right, with the possibility to load several measurements and switch from one to another easily...
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Old 25th January 2013, 01:58 AM   #344
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Quote:
Originally Posted by bohdan1232000 View Post
I think we can safely leave equalizing the -40dB dips and also measurement noise out of this discussion.
Actually you can't. Measurement noise is ALWAYS there. If your EQ method doesn't deal with it, the resulting EQ will convolve noise sidebands onto sine waves.

Quote:
I am specifically interested in what characteristics are exhibited by equalized system. For simplicity, we can call it loudspeaker+equalizer (no crossover).

Your comment is:

“..It usually results in sorta Linear Phase for the 'pass band'. The smaller the error band, the closer it is to LP but it doesn't really get there. It NEVER gets close to Minimum Phase except for trivial examples….”

OK, I’ll take this for an answer, due to lack of others. So, my understanding is, that the Kirkeby equalizer does not offer minimum-phase equalization.
Not in any of the forms I've seen including Angelo's.

Quote:
Secondly, I am specifically interested in transition regions, their characteristics and the consequences of introducing them. You have not commented on these?.
Usually, the 'transition' regions simply have bigger 'error bands' so less EQ is done. So less close to Linear Phase. Results vary from example to example.

Remember, Kirkeby is simply an elegant method to avoid EQing big dips. It doesn't do anything more. But it does this very well if a lot of reflections make the original VERY non-Minimum Phase.

For 'anechoic' or 'near anechoic' EQ of speakers, IMHO there are better methods. From what I can see, this is what most people here are doing here. DRC has good discussions on the topic though I don't agree with everything Dennis says.

I do points 1-4 with my favourite methods which involve medium size IIRs and address noise modulation & measurement artifacts.

I don't think points 5-6 are worth doing cos my Blind Listening Tests bla bla including Greenfield etc. But this was all 20 yrs ago and we really didn't have the computing power to do 5-6 properly in da old days. I'd like to re-visit this but I'm really a beach bum these days.

Even then, I believe there are far more important things to correct than excess phase. Just my $0.02

BTW, the original Kirkeby paper was presented to the Institute of Acoustics, NOT AES. The co-author (who's name I ashamed to have forgotten) is now head of the ISVR, Southampton. Steve Elliot

Last edited by kgrlee; 25th January 2013 at 02:06 AM.
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Old 25th January 2013, 02:05 AM   #345
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Guys, those discussions are really interesting (I mean it), but I would prefer to keep that thread a little bit more focused on rePhase if possible. As it is now a user of rePhase would have difficulties to find any relevant information on the use of that software in that thread.
Comparisons between automated and manual corrections are interesting, as it allow the user to choose its path, but then for the sake of clarity I feel discussions really specific to automated correction and UE should be kept out of that thread if possible (there is a dedicated thread).

Did you guys actually *try* rePhase?

Last edited by pos; 25th January 2013 at 02:12 AM.
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Old 25th January 2013, 03:14 AM   #346
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Hi kgrlee,

Many thanks for the comments.

Perhaps I am the one confused here. This is because I was lead to believe, that Kirkeby algorithm was used to equalize loudspeakers, or in general acoustic systems.
For example, Angelo Farina’s “Advancements in impulse response measurements by sine sweeps”, where he uses the Kirkeby filter to equalize his acoustic measurement system. Interestingly, even though Farina calculated the Kirkeby equalizer filter from 10Hz to11kHz, the filter did not equalize below 200Hz, prompting Farina to make a comment about it.

My concern about the transition regions was, that introducing “manually shaped” transition function at low-end of the woofer response and high-end of the tweeter response, creates something completely artificial, and inserts it into the frequency response being immediately adjacent to the 3dB pass band. Were the effects ever evaluated/measured?. Is it still Hi-Fi?.

Anyway, thank you for your insight, as I have now developed a firm conviction to stay away from the Kirkeby algorithm in relation to loudspeakers EQ.




Pos,

I did not mean to highjack the thread – sorry. The thread was dying anyway, and people were visibly loosing interest. Since Kirkeby was mentioned in the thread, I wanted to pick the brains of people-in-the-know. This was not discussion about UE.


Best Regards,
Bohdan
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Old 25th January 2013, 04:09 AM   #347
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Quote:
Originally Posted by bohdan1232000 View Post
... lead to believe, that Kirkeby algorithm was used to equalize loudspeakers, or in general acoustic systems.
I think it is only Angelo (and his disciples) who use the word "Kirkeby" in this way. You need to look up the original IoA paper to confirm this.

Quote:
For example, Angelo Farina’s “Advancements in impulse response measurements by sine sweeps”, where he uses the Kirkeby filter to equalize his acoustic measurement system. Interestingly, even though Farina calculated the Kirkeby equalizer filter from 10Hz to11kHz, the filter did not equalize below 200Hz, prompting Farina to make a comment about it.
Angelo & I have been discussing this for a long time, as is Fons Adreaensen who is a true Linux DSP guru. (I'm only a pseudo one ) Fons & I like IIRs while Angelo prefers FIRs, hence his liking for "Kirkeby". Fons now works for Angelo so perhaps we can turn him from the Dark Side.

Quote:
My concern about the transition regions was, that introducing “manually shaped” transition function at low-end of the woofer response and high-end of the tweeter response, creates something completely artificial, and inserts it into the frequency response being immediately adjacent to the 3dB pass band. Were the effects ever evaluated/measured?. Is it still Hi-Fi?.
Angelo doesn't really have transition bands as such. He just lets the 'error' become greater there.

There's many ways to treat the 'transition'. One method is the "Target Function" approach first published by KEF. In some ways, all the evil LR xovers are adaptations of this.

I used to work for Calrec who designed mixers mainly for Broadcast organisations like the BBC. From that, I have a strong liking for band limited signals cos they sound better. There is much BBC research into this and I have done so too.

Quote:
.. as I have now developed a firm conviction to stay away from the Kirkeby algorithm in relation to loudspeakers EQ.
Don't dismiss Angelo's "Kirkeby" too quickly. When you start doing room EQ, you may find it useful. Especially as you use evil FIRs

pos, much of what we discussed here is of relevance to any Digital EQ method.
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Old 25th January 2013, 05:49 AM   #348
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Quote:
Originally Posted by bohdan1232000 View Post
Pos,

I did not mean to highjack the thread – sorry. The thread was dying anyway, and people were visibly loosing interest. Since Kirkeby was mentioned in the thread, I wanted to pick the brains of people-in-the-know. This was not discussion about UE.


Best Regards,
Bohdan
This thread was not meant to be ultra active, it was meant to be the reference thread for rephase (as there is no documentation), and it is linked in the rephase sourceforge page. It is meant for people (hopefully users) to ask questions and find answers about rephase without skimming hundreds of OT messages.

If you look at the first post as well as the first few pages you will find that there was an effort to consolidate something around this idea (changelog, new version announcement, feature requests, bug reports, etc.).

If the thread is not active for XX days, weeks, or months, so be it: it will remain accessible (from google or the sourceforge page, etc.) for people to ask questions and find examples and links. I prefer this to an overactive thread with none of the participants being actual users of rephase, even if the discussion are interesting...
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Old 25th January 2013, 05:57 AM   #349
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Quote:
Originally Posted by kgrlee View Post
pos, much of what we discussed here is of relevance to any Digital EQ method.
so why did it end up in this particular thread?

Seriously, I am not against some general EQ/FIR/convolution discussion, especially when there are so knowledgeable people like you involved, but it would be more on topic if it could be related to some degree to rephase ("from time to time" as CopperTop said). Why not open a more general thread about those topics?
I would be very interested in your opinion and evolution ideas about rephase if you would try it.
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Old 25th January 2013, 09:20 AM   #350
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Quote:
Originally Posted by kgrlee View Post
BTW, the original Kirkeby paper was presented to the Institute of Acoustics, NOT AES. The co-author (who's name I ashamed to have forgotten) is now head of the ISVR, Southampton. Steve Elliot
Du.uuh! It were Phil Nelson before he became Prof Nelson. The original was an IoA paper but there is a JAES version.

[16] Kirkeby, 0. & Nelson, P.A, "Digital Filter Design for Inversion Problems in Sound Reproduction", JAES Vol 47, jul/aug99
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