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Old 25th January 2013, 08:30 AM   #351
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pos, I tried re-reading this thread but gave up at about page 20.

I was trying to find out if you actually implemented IMPORT of a WAV file impulse or other measurement. This is certainly worth doing if just to make it easier for users to see what improvement they get.

Another worthwhile addition is Minimum Phase EQ.

You might like to have a look at

[25] Ramos G & Lopez J, "Filter Design Method for Loudspeaker Equalization Based on IIR Parametric Filters", JAES 54 no12 dec06

It's an auto method for IIRs but you might find it useful for your dreaming up a target which you can implement with your FIR. In "Simple Arbitrary IIRs", I say of this method ..

"The largest discrepancy between the current response and the target is EQd with the first parametric, then the largest remaining discrepancy with the second parametric and so on. When the parametrics are used up, go back to the first parametric and see if it can be adjusted for further improvements and so on. Continue iterating until some criteria is met."

Just my $0.02 and trying not to let my prejudices show too much ..
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Old 25th January 2013, 10:24 AM   #352
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my little brain has understood why it's needed tu use IIR EQ.

because FIR EQ do not modify the imaginary part,only the real part.

i've checked it with REW.(spl correction only )
-uncorrected curve
-FIR corrected curve ,group delay is not corrected,only spl...
-IIR corrected curve (the nicer group delay)

that allows to obtain a textbook group delay if speaker is corrected till transition band with IIR EQ.
..all work has to be redone at home.
it's very fast with the REW automatic IIR EQ.
we can export the impulse filter from REW and convolve it with a phase linearization IR (manualy with HOLM or REW).
so what the linear phase EQ is done for ?


Click the image to open in full size.

Last edited by thierry38efd; 25th January 2013 at 10:51 AM.
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Old 25th January 2013, 10:28 AM   #353
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Quote:
Originally Posted by thierry38efd View Post
my little brain has understood why it's needed tu use IIR EQ.

because FIR EQ do not modify the imaginary part,only the real part.
???? FIR can do anything IIR can. If it doesn't in your example it is because of the way FIR has been implimented.
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Old 25th January 2013, 10:37 AM   #354
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Quote:
Originally Posted by kgrlee View Post
pos, I tried re-reading this thread but gave up at about page 20.

I was trying to find out if you actually implemented IMPORT of a WAV file impulse or other measurement. This is certainly worth doing if just to make it easier for users to see what improvement they get.
You see, that is the problem
No there is not measurement importing functionality for now. It was said in post #5 already, and several times along the thread. Last time was in port #343, 8 posts above yours
I plan to add this functionality in a next version.

Quote:
Another worthwhile addition is Minimum Phase EQ.
Yes a minimal phase EQ section would indeed be nice to have.
As it is now you are supposed to correct amplitude and phase independently, but an additional minimum phase section would simplify most inband corrections.
Phase-only correction remain the prefered method for correcting phase shifts in the lower band of an horn for example, as minimal-phase phase linearization would require *lots* of EQ down low (Linkwitz transform).

That said, depending on the way the convolution is implemented, minimal phase EQ can most of the time be implemented in the filter (DCX2496, etc.).

Quote:
Originally Posted by kgrlee View Post
You might like to have a look at
thanks for your link

Last edited by pos; 25th January 2013 at 10:44 AM.
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Old 25th January 2013, 10:49 AM   #355
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Originally Posted by john k... View Post
???? FIR can do anything IIR can. If it doesn't in your example it is because of the way FIR has been implimented.
yes,i meant rePhase linear phase EQ .
i gonna use REW minimum phase EQ in a FIR filter right now.
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Old 25th January 2013, 02:16 PM   #356
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Originally Posted by kgrlee View Post
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.

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.

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.

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.
Kirkeby inverse is much more than an EQ.

Here is FR of Peerless 2" full range measured at 23cm with FR of inverse generated using Kirkeby plugin with small regularization parameters from 2Hz to 24kHz:

1Peerless 2 23cm and raw kirk.gif

Horrible above 21kHz; but low end shows exactly EQ for bass extension.

A zoomed in view:

2Peerless 2 23cm and raw kirk zoom.gif

The above shows a little bit more of inverse's detail in mirroring response.


Convolution of IR and inverse yield ruler smooth spectrum.

Resultant IR:

3IR sample view.gif

Above view highlights perfect pulse type response.

Zooming out time to 1k samples and in on amplitude:

4IR 1k sample -54db.gif

The dreaded pre-ringing that so many people don't understand. Here it most definitely can't be heard. All this is related to low pass behavior just below Nyquist. To highlight this, a 256 point 18kHz low pass filter with Blackman-Harris windowing is applied:

5IR LP18k 262sample -54db.png

Once again, no way, no how is this filtering ringing audible when applied to program source.

Next view is IR 82k samples:

6IR 82k sample -72db.png

A super low frequency ringing is seen below -66dB. Noise is not an issue. Function is still trying to get 2" driver performing bass duties.

This is where high pass choice is made. Here is IR with Linkwitz-Riley 24dB/oct 250Hz high pass filter:

7IR LR4 250HP 1ksamples.gif

Above filter destroys square waves, and pleases people stuck in IIR paradigm do to not understanding basics of waveform propagation and information transmission systems.

Here is above with linear phase, that will pass square waves:

8IR LR4 250HP 1ksamples linear.gif

Once again a pre-ringing is seen that is completely inaudible when applied to continuous source signal. Judging filter performance based on looking at response is pointless without understanding how a wave function in time works as a filter via convolution. Likewise, listening to the filter played as source is completely pointless.

kgrlee: Farina's sweep pair generator calculates inverse for exponential swept sine in time domain. Convolution of these sweep pairs always results in sync function that fouls up results.
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Old 25th January 2013, 06:36 PM   #357
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Quote:
I was trying to find out if you actually implemented IMPORT of a WAV file impulse or other measurement. This is certainly worth doing

if just to make it easier for users to see what improvement they get.
Click the image to open in full size.

open HOLM software (same work can be done with REW with .wav file)
HOLM hates .wav file

Click the image to open in full size.

REW

using rePhase.

open software and create a Hpf and a Lpf filter
advices for setting IR are done in the window

Click the image to open in full size.

generate IR (16 or 24 bits )
.wav for convolver and REW simulation
.txt for HOLM simulation

open REW-->file-->import impulse response-->choose IR generated by rephase

import lpf.wav and import hpf.wav
all spl-->controls tab-->generate A+B (lpf.wav+hpf.wav)

Click the image to open in full size.

you can apply the filter IMPULSE to speaker measurement.convolution are LPF x woofer and HPF x tweeter in this case.
generate the sum filtered_woofer + filtered_tweeter

Click the image to open in full size.

you can do any simple math operation.

Making your own house target and create the FIR filter equalization file

open rePhase

create with filters and EQ the house curve you need.(slopes,low frequency boost ,subsonic filter )

Click the image to open in full size.

export the target with target.txt for holm
export the target with target.wav for REW.
open HOLM or REW

A slot import the target.txt you've created with rephase.
B slot import the measurement (right or left at the listenning spot ).

change offset to merge the two curves,it will avoid high positive corr.

C=A/B
it gives the global equalization.

Click the image to open in full size.

the green curve (C) is the result of difference between measurement and target.

the limitation of positive correction.
C=A x B
create with rephase an IR with negative EQ to reduce the too positive ones

Click the image to open in full size.

export as .WAV,and make a stereo file with an another measurement.

Last edited by thierry38efd; 25th January 2013 at 06:46 PM.
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Old 25th January 2013, 06:48 PM   #358
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Originally Posted by Barleywater View Post
... Once again, no way, no how is this filtering ringing audible when applied to program source. ..... Above filter destroys square waves, and pleases people stuck in IIR paradigm do to not understanding basics of waveform propagation and information transmission systems. ...

Once again a pre-ringing is seen that is completely inaudible when applied to continuous source signal. Judging filter performance based on looking at response is pointless without understanding how a wave function in time works as a filter via convolution. Likewise, listening to the filter played as source is completely pointless.
Barley, thank you for your learned and constructive comments on pre-ring, waveform propagation, info tranmission, convolution and other stuff.

If you don't mind, I'll wait until I can carry out audibility tests on pre-ring to confirm or disprove my own experiments and trials. (in the 80's & 90's using primitive DSP chips).

BTW, it was possible to make speakers that reproduced good square waves over a "large bandwidth" using purely analogue means for a long time. I first did it in the late 70's. I'd quote some AES papers but its obvious you have no need to read them.
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Quote:
Farina's sweep pair generator calculates inverse for exponential swept sine in time domain. Convolution of these sweep pairs always results in sync function that fouls up results.
Errh! Angelo's method is for measuring impulse responses. You can get it to have imperceptible 'sync functions' for any fs. If you have "sync function that fouls up results", you are probably using it wrong.

BTW, ALL sampled methods of impulse response measurement have a 'sync' nature. If you don't use it properly, this will become visible.

I did the theory for Angelo's method circa 1990. I wanted it for measuring speaker response & distortion in the shortest theoretically possible time for prodution testing. But at that time, the computing power and particularly good DACs were too expensive.

But I'm quite happy for Angelo to take the credit cos he has done loadsa work to popularize it.
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Barley, how do you measure the "anechoic" response that you EQ? In a room?

Last edited by kgrlee; 25th January 2013 at 06:55 PM.
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Old 25th January 2013, 07:48 PM   #359
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Quote:
Originally Posted by kgrlee View Post
Errh! Angelo's method is for measuring impulse responses. You can get it to have imperceptible 'sync functions' for any fs. If you have "sync function that fouls up results", you are probably using it wrong.

BTW, ALL sampled methods of impulse response measurement have a 'sync' nature. If you don't use it properly, this will become visible.

Barley, how do you measure the "anechoic" response that you EQ? In a room?
For tweeter in previous post paired with upward firing woofer of Pluto type speaker above 23cm measurement is used for making both IR measurements.

Angelo's gensweep module creates sweeps which add artifacts which become garbage when used with his Kirkeby inverse module that shows up as mixed phase IR.

MLS as sampled measurement system doesn't return a sync function. Pure correlation. Swept sine measurement over full band can also return near perfect correlation when at least partially constructed in the frequency domain.

kgrlee: So what sweep pair would you generate with Angelo's gensweep to measure the above Peerless 2" driver?
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Old 25th January 2013, 08:10 PM   #360
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can we have a look to screenshot measurement ?

please,around listening spot,1,2 and 3 feets up and down.
looking to a impulse response is not very confortable.

it will be very usefull for everybody.

Last edited by thierry38efd; 25th January 2013 at 08:26 PM.
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