I find their approach is very interesting.
http://www.grimmaudio.com/whitepapers/speakers.pdf
1. EQ the drivers beyond operating range to flat
2. Apply the intended crossover, eg. LR4.
3. Apply time delay.
So far so good. But what about this 4th step, quote ..
Finally, the icing on the cake. The sum of an ideal LR4
system is a second order all-pass with a Q of 0.7. In order
to avoid the problems associated with correcting
phase exactly, build an inverse all-pass filter based on
the theoretical ideal. This filter will be non-causal so
there’s a good reason for using FIR.
Questions:
- What do they mean by " The sum of an ideal LR4
system is a second order all-pass with a Q of 0.7" ?
- What do you think " an inverse all-pass filter based on
the theoretical ideal" ?
If this can be realised using MiniDSP it would be great I think.
http://www.grimmaudio.com/whitepapers/speakers.pdf
1. EQ the drivers beyond operating range to flat
2. Apply the intended crossover, eg. LR4.
3. Apply time delay.
So far so good. But what about this 4th step, quote ..
Finally, the icing on the cake. The sum of an ideal LR4
system is a second order all-pass with a Q of 0.7. In order
to avoid the problems associated with correcting
phase exactly, build an inverse all-pass filter based on
the theoretical ideal. This filter will be non-causal so
there’s a good reason for using FIR.
Questions:
- What do they mean by " The sum of an ideal LR4
system is a second order all-pass with a Q of 0.7" ?
- What do you think " an inverse all-pass filter based on
the theoretical ideal" ?
If this can be realised using MiniDSP it would be great I think.
In my understanding:
you
1º LR24 filters are non phase linear filters which happen to have same (or similar) group delay pattern as a 2º order all pass filter.
2º Consequently, if you reverse this all pass filter, the product ( convolution) of the new filter (reversed all pass) and the LR24 will give you a "linear phse" LR24.
Hypex promised to make this feature available in their DSp products some time ago, but it seems that Grimm is their only user... But you can do this thanks to other providers.
You can generate a FIR filter like this (reversed allpass) with a soft as Acourate and use Convolver to eq your LR24 into a linear phase filter.
Or you can use Thuneau Arbitrator which doeas somerhing similar the IIR way.
you
1º LR24 filters are non phase linear filters which happen to have same (or similar) group delay pattern as a 2º order all pass filter.
2º Consequently, if you reverse this all pass filter, the product ( convolution) of the new filter (reversed all pass) and the LR24 will give you a "linear phse" LR24.
Hypex promised to make this feature available in their DSp products some time ago, but it seems that Grimm is their only user... But you can do this thanks to other providers.
You can generate a FIR filter like this (reversed allpass) with a soft as Acourate and use Convolver to eq your LR24 into a linear phase filter.
Or you can use Thuneau Arbitrator which doeas somerhing similar the IIR way.
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from interview with Siegfried Linkwitz
Audioxpress 7/2011
"JD: Looking at your Orions, I can’t
but wonder whether that purposedesigned
analog crossover cannot be
replaced by a DSP unit?
SL: In principle, I am convinced that
it can be done. But it’s not trivial. You
need to measure and adjust the whole
chain, analog in to sound out. Now
someone may come up with an FIR filter
that makes everything linear phase,
but you get pre-ringing from the filter
and the sound may combine correctly
at one point in space but what about
off-axis?
You could use IIR filters to mimic the
analog filters, and that would probably
be my approach. Then I would first
make the amplitude response correct
and then you can do an overall phase
response correction of the complete
system. That’s how Bruno Putzeys
does it in his Grimm Audio two-way.
And I would not buy a standard DSP
box off the market. The devil is in the
details. But it’s hard, and it’s not on my
to-do list!"
Audioxpress 7/2011
"JD: Looking at your Orions, I can’t
but wonder whether that purposedesigned
analog crossover cannot be
replaced by a DSP unit?
SL: In principle, I am convinced that
it can be done. But it’s not trivial. You
need to measure and adjust the whole
chain, analog in to sound out. Now
someone may come up with an FIR filter
that makes everything linear phase,
but you get pre-ringing from the filter
and the sound may combine correctly
at one point in space but what about
off-axis?
You could use IIR filters to mimic the
analog filters, and that would probably
be my approach. Then I would first
make the amplitude response correct
and then you can do an overall phase
response correction of the complete
system. That’s how Bruno Putzeys
does it in his Grimm Audio two-way.
And I would not buy a standard DSP
box off the market. The devil is in the
details. But it’s hard, and it’s not on my
to-do list!"
Go to Active Filters
What is called allpass or reversed allpass is also called delay correction by S. Linkwitz. See his analog implementations. It "fixes" the phase error of the 4th order LR filter and provides the phase correction needed. Used with success in the Orion active XO. To me a changing of phase is all about delay.
What is called allpass or reversed allpass is also called delay correction by S. Linkwitz. See his analog implementations. It "fixes" the phase error of the 4th order LR filter and provides the phase correction needed. Used with success in the Orion active XO. To me a changing of phase is all about delay.
Take an analog textbook LR4 crossover and sum the two outputs together and look at the transfer function relative to the input and you'll see the second-order all-pass function.
An inverse all-pass filter would yield an opposite phase response to what you just observed. Then, if you apply it to your textbook LR4 crossover the final phase response would be linear.
Regarding the a MiniDSP solution......I'm afraid not. The minidsp can add pure delay or create an all-pass function, but it can't create an inverse all-pass function. You need something more advanced to create that correction. A FIR based system or some sort of reverse-processed IIR filtering like used in the Thuneau.
Cheers,
Dave.
Thanks all, that clears up a lot of things! So in summary
- It is a good (smart) idea, even supported by SL
- It is a phase correction of IIR filter, using FIR
- It has the benefit of IIR filter and FIR, without the drawbacks of FIR.
- It cannot be implemented with MiniDSP (IIR).
So there is no way of creating the inverted allpas using Biquad for example?
- It is a good (smart) idea, even supported by SL
- It is a phase correction of IIR filter, using FIR
- It has the benefit of IIR filter and FIR, without the drawbacks of FIR.
- It cannot be implemented with MiniDSP (IIR).
So there is no way of creating the inverted allpas using Biquad for example?
Imho, the only reason to accept an iir based solution is to avoid a computer in our set up. Linear phase means delay which means memory. So this is the limit for the hardware, and a computer has practically no limits.
And with a computer you can do much more than eq the phase of your cross over. Like plug your TD124 direct to a good ADC, use FIR RIAA correction, remove clicks and crackle to your old LPs while playing... What else ?
And with a computer you can do much more than eq the phase of your cross over. Like plug your TD124 direct to a good ADC, use FIR RIAA correction, remove clicks and crackle to your old LPs while playing... What else ?
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I still don't want one in my stereo !!! :-(
Back to on-topic:
This paper here is agood description of a practical implementation:
http://downloads.goldmund.com/downl...kZg/goldmund_proteus-leonardo_white_paper.pdf
Regards
Charles
Back to on-topic:
This paper here is agood description of a practical implementation:
http://downloads.goldmund.com/downl...kZg/goldmund_proteus-leonardo_white_paper.pdf
Regards
Charles
Interesting paper. But i don't want a solution that binds me to a hifi manufacturer hardware propiretary solution.
Though i understand that these might be the only possible salvation for that industry, keeping up to date the old classical ( and a bit boring too...) 2 channels setups, with good old passive xovers and may be not so good caps and coils. But hey, there are plenty of expensive hiend esoteric ones around...
Though i understand that these might be the only possible salvation for that industry, keeping up to date the old classical ( and a bit boring too...) 2 channels setups, with good old passive xovers and may be not so good caps and coils. But hey, there are plenty of expensive hiend esoteric ones around...
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A "proprietary" solution is what was called for at the beginning. I.e. how can it be implemented (specially on small DSP crossover hardware) ?
If these boards can do 2nd order allpass filters (the ADI SigmaStudio environment for instance can do it) a classic allpass group-delay EQ could be "built". One has to take care however that a group-delay EQ dimensioned for the analog domain may probably not exactly work as intended in the digital domain due to the frequency warp.
Regards
Charles
If these boards can do 2nd order allpass filters (the ADI SigmaStudio environment for instance can do it) a classic allpass group-delay EQ could be "built". One has to take care however that a group-delay EQ dimensioned for the analog domain may probably not exactly work as intended in the digital domain due to the frequency warp.
Regards
Charles
Absolutely right, and this is what Hypex anounced a couple of years ago with DLCP.
I said "That's what i want!", but 2 years after it is still in beta ... or in the Grimm stuff...
No problem. In the end i think it's better to get different providers for the hard and for the soft.
The rest is too often "not yet availabe" or " already obsolete".
I have never looked at the Hypex xover so far but I think it uses a DSP from the Sigma processor family. If the programming GUI allows one could implement the Lipshiz-Vanderkooy type crossovers with ease.
I have tried it with SigmaStudio once and it worked as proposed but the built-in simulator has its problems displaying the proper frequency response. See post #4 here:
EngineerZone: Problem with subtractor block
Regards
Charles
I have tried it with SigmaStudio once and it worked as proposed but the built-in simulator has its problems displaying the proper frequency response. See post #4 here:
EngineerZone: Problem with subtractor block
Regards
Charles
I'm going to build LS1 Clones. W22EX001 and DXT are already here, as well as Hypex AS2.100.
Finally, the icing on the cake......looks really impressive in the step and phase response.....but, are we discussing now the audibility of a group delay caused by an LR4@1500Hz?
I think the wide baffle and steps:
are the essential and relevant design features.
Christoph- Status
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