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rePhase, a loudspeaker phase linearization, EQ and FIR filtering tool
rePhase, a loudspeaker phase linearization, EQ and FIR filtering tool
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Old 2nd March 2018, 06:48 AM   #2461
BYRTT is offline BYRTT  Denmark
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rePhase, a loudspeaker phase linearization, EQ and FIR filtering tool
Sounds right Rajapruk.
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Old 2nd March 2018, 08:17 AM   #2462
pos is offline pos  Europe
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But don't forget to remove it before generating the FIR, as that protection filter will not be part of the final system.
I will try to clarify this use of the compensate filter by adding a special version in the measurement tab that only shows on the measurement...
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Old 2nd March 2018, 11:26 AM   #2463
Rajapruk is offline Rajapruk  Sweden
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Sounds like a good idea.
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Old 3rd March 2018, 08:38 PM   #2464
pos is offline pos  Europe
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Quote:
Originally Posted by thierry38 View Post
Missing a frequency domain coeffs export.
Hi Thierry,

As a follow up, I think the most logical way of implementing that feature would be to add a "frequency response (.txt)" output format that would be a straight FFT of the generated impulse.
Would that suit your needs?
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Old 4th March 2018, 05:35 PM   #2465
torgeirs is offline torgeirs  Norway
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Manipulating many drivers can be timeconsuming and prone to human error.
Is it any posibility to open up the format of the .rephase settingsfile?
Then such files could be made with scripting tools.

If you made a comand line version of rephase the filtergeneraton could also be scripted and even "realtime" changing of parameters could be possible.

I completely understand if the answer is a big NO!

I ask of course because I think rePhase is a really nice, easy and well proven tool for FIR generating. So thank you, POS!

Last edited by torgeirs; 4th March 2018 at 05:38 PM.
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Old 4th March 2018, 07:05 PM   #2466
diyaudnut is offline diyaudnut  United States
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Quote:
Originally Posted by pos View Post
The measurement you import into rephase already has (or should have) some sort of travel delay compensation to make it close to its minimum-phase behavior.

The good news is that you don't need to worry about all this until the very last part of the crossover process, ie when actually attempting to sum the drivers
The needed delays will not be the same with a corrected and uncorrected driver, so no need to worry at this stage.

When analyzing a full range measurement, most of the time you want to ignore the low pass (20kHz-ish), as you will not (and should not!) attempt to correct its phase.
So in this situation the best approach is to center the measurement impulse on the first positive peak (and of course you want this peak to be positive, or you need to switch polarity), which results in a phase curve reaching 0° near the low pass (and possibly falling apart afterward)
This is what HOLM will do automatically for example, but you can also adjust this in the measurement tab in rephase, possibly after some minimum-phase EQ, to still get that "lazy" 0° target.

Now when dealing with an active crossover the situation is different, as you cannot ignore the phase behavior of the low pass of a driver that you will have to integrate with another one, ie integrate that "natural" low pass in a crossover.
If the low pass is well defined you can sometime "guess" what the proper offset should be to get the textbook phase behavior, but this can be complicated.
In this situation, the easiest approach is to start with the "positive peak" centering trick (as above) and then apply the compensation (and minimum-phase EQ) so that you get a flat magnitude response deep into the low pass stop band (up to the noise floor "wall"), and then play with the measurement time offset to get a flat phase curve in that same area (note that if you then undo your compensation correction you will see what the real minimum-phase response of your measurement looks like).
You can then build your correction, including linear-phase filters, and you should not need any phase EQ to get a flat phase (in fact if you need some this means that you failed at the above step).

Now, once everything is corrected (ie acoustical crossover are well matched across drivers), all you need to do is enter the appropriate delay into your crossover (including possible differences in FIR length and delays) to get a proper summation.
If everything is done correctly your offset differences should be very close to the geometrical distance differences from the drivers' emitting surfaces.
So you start with that, and adjust carefully with measurements (at a proper distance, ideally the listening position(s)), with one driver in reversed polarity, seeking for the deepest null (well known trick that gets the job done!)...

Hope this helps
Thank you Pos for explaining elaborately.
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Old 5th March 2018, 08:33 AM   #2467
pos is offline pos  Europe
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Quote:
Originally Posted by torgeirs View Post
Manipulating many drivers can be timeconsuming and prone to human error.
Is it any posibility to open up the format of the .rephase settingsfile?
Then such files could be made with scripting tools.

If you made a comand line version of rephase the filtergeneraton could also be scripted and even "realtime" changing of parameters could be possible.

I completely understand if the answer is a big NO!

I ask of course because I think rePhase is a really nice, easy and well proven tool for FIR generating. So thank you, POS!
The format changes from one version to another, when features are added or modified, and retro compatibility is handled within rephase with a conversion layer for each successive version. That is a not a good candidate for this kind of use

At one point I considered doing an API version, letting an external client connect to rephase with TCP and emit commands, but that is out of the scope for now.
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Old 5th March 2018, 10:49 AM   #2468
arcgotic is offline arcgotic  Romania
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Hello,
I did some tests and reading, I have 2 questions:
1) @Pos, A few posts ago you said that the number of taps should be as few as possible, and that If target response is reached, it is OK. On other web pages (I don't saved them..) it is said that more taps is OK, because ripples in the impulse response become much flatter. I don't hear difference in sound. The best way to go is as little number of taps, correct?


2) The exported impulse response. If opened in REW > Import Impulse response. Should the impulse response spike be as close to ideal? Why I am asking. If I flatten the phase with Paragraphic Phase EQ, for the range the speaker will play, the impulse response looks much better in REW than the one created with Minimum Phase Box linearization. The question is, should I use the impulse response that looks good in REW? The measurement for that speaker is the same with either impulse response file used.

Thanks all!

And 3) (just thought of it). My Asus card can play at 192khz sample rate. But 99% of my music is 44.1Khz. Now the sound in Windows is set at 192, and all impulse files are created at 192. Is it better to go to 44.1khz with audio, impulse then can have 4times less taps for the same response target? What would you do?
Thanks again!

Last edited by arcgotic; 5th March 2018 at 11:06 AM.
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Old 5th March 2018, 01:57 PM   #2469
thierry38 is offline thierry38  France
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Quote:
Originally Posted by pos View Post
Hi Thierry,

As a follow up, I think the most logical way of implementing that feature would be to add a "frequency response (.txt)" output format that would be a straight FFT of the generated impulse.
Would that suit your needs?
Hi Thomas,
Perfect !
I was thinking taking the last itération (FFT <--->iFFT) in the frequency domain of the generating process.
about 2~3 hundreds points.(or raw coeffs)
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Old 5th March 2018, 06:01 PM   #2470
pos is offline pos  Europe
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Quote:
Originally Posted by thierry38 View Post
Hi Thomas,
Perfect !
I was thinking taking the last itération (FFT <--->iFFT) in the frequency domain of the generating process.
about 2~3 hundreds points.(or raw coeffs)
Alright then
The number of points will be dictated by the FFT length.
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