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Which is the best DSP option: DIRAC vs Acourate vs Audiolense vs RePhase vs ?

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JRS

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Nothing in Room Correction software is quite as complicated as that. Dirac is different in this regard as the multiple measurements are separate and information from each measurement can be processed separately, they do have some papers I have read before describing parts of it.

These are the steps performed by DRC_FIR
  1. Initial windowing and normalization of the input impulse response.
  2. Optional microphone compensation.
  3. Initial dip limiting to prevent numerical instabilities during homomorphic deconvolution.
  4. Decomposition into minimum phase and excess phase components using homomorphic deconvolution.
  5. Prefiltering of the minimum phase component with frequency dependent windowing.
  6. Frequency response dip limiting of the minimum phase component to prevent numerical instabilities during the inversion step.
  7. Prefiltering of the excess phase component with frequency dependent windowing.
  8. Normalization and convolution of the pre-processed minimum phase and excess phase components (optional starting from version 2.0.0).
  9. Impulse response inversion through least square techniques or fast deconvolution.
  10. Optional computation of a psychoacoustic target response based on the magnitude response envelope of the corrected impulse response.
  11. Frequency response peak limiting to prevent speaker and amplification overload.
  12. Ringing truncation with frequency dependent windowing to remove any unwanted excessive ringing caused by the inversion stage and the peak limiting stage.
  13. Postfiltering to remove uncorrectable (subsonic and ultrasonic) bands and to provide the final target frequency response.
  14. Optional generation of a minimum phase version of the correction filter.
  15. Final optional test convolution of the correction filter with the input impulse response.
Frequency dependent windowing is the main function that assists with separating the speaker from the room. In REW the window is quite basic and is in cycles or octaves for the whole range. This still means that the window is short at the top and long at the bottom.
So you get progressively more room in the measurement as frequency goes down.

DRC_FIR goes further with a sliding window where the length of the window can be set different at both the low frequency and high frequency independently and there is also an exponent parameter to change how the two interact in the middle. Changing the window exponent is one factor that has a big impact on the way the correction sounds. Differences of 0.01 are audible as it strongly affects the midrange.

View attachment 183777

This really lets you tailor just how much speaker correction vs room correction is being done. I see very few speakers where the direct sound still couldn't be smoother but whether messing with this has a positive effect depends a lot on how even the directivity of the speaker being corrected is.
Interesting. I think your explanation has helped me gain a deeper appreciation of the problem. For one, it leaves me thinking that there is much to recommend high DI speakers away from the walls with separate ambience channels.

I just wonder how well the Apple Music Atmos remastered titles will sound. Personally, the films I enjoy the most don't depend a lit on gee whiz sound effects. It's not that I don't enjoy the enhanced realism. Just been never enough of a draw to invest tge time and equipment into a multichannel set up. One of those two channel purists (or Luddites depending on ones perspective) I guess, as done fight, there is great satisfaction to be had.
Of course Toole and others hzvd been saying this for a long time. Always been a matter of software, and perhaps we are at the dawn of a new era when there is a plethora of music in MCh formats, and having 4 or 5 really good speakers will be far more important than having two gobsmackers, cable lifters, along with Shaman and quantum mechanics to get everything just so.

I really want to thank you for breaking down the DRC flow for me. That exponential windowing sensitivity reminds me a bit of tinkering with gamma in photoshop type software. I can imagine it doesn't take much tinkering to ruin an otherwise good image. Gonna have to mull all of this over a bit.
 

fluid

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For one, it leaves me thinking that there is much to recommend high DI speakers away from the walls with separate ambience channels.
I think is a valid way of trying to have your cake and eat it too. This is the route I am pursuing myself.
I really want to thank you for breaking down the DRC flow for me. That exponential windowing sensitivity reminds me a bit of tinkering with gamma in photoshop type software. I can imagine it doesn't take much tinkering to ruin an otherwise good image. Gonna have to mull all of this over a bit.
You are welcome. A WE of 1.0 is a straight line, going less results in a shorter window for midband correction, this is my preference and I use 0.93, no other changes have made me want to deviate from this very far.
 

Holmz

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Interesting. I think your explanation has helped me gain a deeper appreciation of the problem. For one, it leaves me thinking that there is much to recommend high DI speakers away from the walls with separate ambience channels.

Probably the opposite. High directivity would have a better chance of minimal interaction for an early reflection.



I really want to thank you for breaking down the DRC flow for me. That exponential windowing sensitivity reminds me a bit of tinkering with gamma in photoshop type software. I can imagine it doesn't take much tinkering to ruin an otherwise good image. Gonna have to mull all of this over a bit.

There are probably equivalencies in image processing… (but I dunnno them.)
 
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Probably the opposite. High directivity would have a better chance of minimal interaction for an early reflection.




There are probably equivalencies in image processing… (but I dunnno them.)
Thats what I meant, Inalways think of some nice 6 x 2' electrostatic panels coming right at you.

As for gamma, it's similar in ghe sense that's an exponential function which has a powerful effect on tonal balance, but with imaging tone being gray scale or red blue balancd if I'm not mistaken. It's actually more akin to audio compression/expansion to maxamize dynamic range with minimal bit space. All I really know is small tweaks can have huge impact on pic quality. Compare the three photos to see the difference if so inclined.

 

Holmz

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Thats what I meant, Inalways think of some nice 6 x 2' electrostatic panels coming right at you.

As for gamma, it's similar in ghe sense that's an exponential function which has a powerful effect on tonal balance, but with imaging tone being gray scale or red blue balancd if I'm not mistaken. It's actually more akin to audio compression/expansion to maxamize dynamic range with minimal bit space. All I really know is small tweaks can have huge impact on pic quality. Compare the three photos to see the difference if so inclined.


Dirac is more like the processing called “Maximum Entropy”, which is used as a sharpening algorithm.
Gamma is probably more analogous with frequency-response / magnitude-EQ (??).

Note that in @fluid ’s post, the steps 4-14 end up becoming a single set of taps for FIR, where Dirac (Impulse-response/phase-response), Any amplitude EQ, and any room EQ… all gets put into a single stage of filtering.
 
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Dirac is more like the processing called “Maximum Entropy”, which is used as a sharpening algorithm.
Gamma is probably more analogous with frequency-response / magnitude-EQ (??).

Note that in @fluid ’s post, the steps 4-14 end up becoming a single set of taps for FIR, where Dirac (Impulse-response/phase-response), Any amplitude EQ, and any room EQ… all gets put into a single stage of filtering.
Exactly, and there in no mathematical evidence that it makes any difference, my opinion is that it is best to separately tune the speakers for the best possible quasi-anechoic response (which is a set of filters you can take anywhere), and then try by ear, or msmt to get the best in room response. That's when you ice the cake with the DRC. I have no proof, but I believe this approach helps to minimize artifact and excessive efforts at correction. I believe many of us have heard that DSP when overdone can sound absolutely dreadful, with comments such as "sucking the life out of the system." Seems to me the above is your best shot--you have optimized the speakers, they sound wonderful, but the room is getting in the way. Then and only then do we try to improve the sound beyond that.
 

kthulhutu

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Anyone know how to use ASIO with Audiolense? I have an RME Babyface Pro FS for my mic (plugged in via USB) and a USB soundcard that sends SPDIF to my speakers.
 

Holmz

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Exactly, and there in no mathematical evidence that it makes any difference, my opinion is that it is best to separately tune the speakers for the best possible quasi-anechoic response (which is a set of filters you can take anywhere), and then try by ear, or msmt to get the best in room response. That's when you ice the cake with the DRC. I have no proof, but I believe this approach helps to minimize artifact and excessive efforts at correction. I believe many of us have heard that DSP when overdone can sound absolutely dreadful, with comments such as "sucking the life out of the system." Seems to me the above is your best shot--you have optimized the speakers, they sound wonderful, but the room is getting in the way. Then and only then do we try to improve the sound beyond that.

The other school of thought is:
1) That the speakers should be time and phase (impulse response) correct, and the speakers out in the room for that the early reflections are >=10 msec, (and room treatments).
2) Then only use amplitude EQ for the < ~400 Hz. Pretty much what Lyngdorf does.

I am still trying to find what s/w solution should go between an RME ADI-Pro, and an Octo DAC-8 for largely FIR based EQ (but IIR can used used for cross overs.)
 

Sal1950

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I am still trying to find what s/w solution should go between an RME ADI-Pro, and an Octo DAC-8 for largely FIR based EQ (but IIR can used used for cross overs.)
I would contact our member @mitchco and get his input on your questions.
AFAIK He has about the deepest understanding of DSP related subjects.
 

JStewart

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Anyone know how to use ASIO with Audiolense? I have an RME Babyface Pro FS for my mic (plugged in via USB) and a USB soundcard that sends SPDIF to my speakers.

If you post on the audiolense forum the author, Bernt, screen name juicehifi, will usually reply.
 

dualazmak

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Just for your reference, I use EKIO on Windows 11 PC feeding crossover-ed/EQ-ed 8 channels into OKTO DAC8PRO in my multichannel multi-driver multi-way multi-amplifier stereo project. You may find here the latest system configuration of my setup including the details of EKIO parameters.

The paid EKIO supports non-limited numbers of output channels, and IN and OUT routings are very much flexible through all the ASIO drivers available in the PC; simple straightforward and well designed GUI for easy and stable operation. Almost all of the parameters can be flexibly changed/adjusted "on the fly".

EKIO uses IIR filters. The processing is done using a cascade of second order transposed direct form II biquad sections. Every calculation is done using 64 bit floating point numbers. EKIO works very fast and stable, small usage of CPU power of my completely silent audio dedicated PCs.

I confirmed that EKIO gives no audible post-ringing nor pre-ringing at all in my rather sensitive test environment.

I recently established 0.1 msec precision time alignment (proved by objective measurement of room air sound, summary here) over all the SP drivers using EKIO's reliable (and fully validated) group delay capabilities.
 
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