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Multi-Channel, Multi-Amplifier Audio System Using Software Crossover and Multichannel-DAC

Lbstyling

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Hello Lbstyling and friends,

Guillaume of LUPISOFT kindly responded to me informing;
"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."

Since I am not an expert in mathematical and theoretical design of audio crossover, the theoretical comparison between EKIO and other XO solutions would be far beyond my knowledge and capabilities.

I would like to share, therefore, only these two articles;
http://www.differencebetween.net/science/difference-between-iir-and-fir-filters/
and
https://www.advsolned.com/difference-between-iir-and-fir-filters-a-practical-design-guide/
I like the "Advantages, Disadvantages" descriptions in this article, especially "Analog equivalent advantage for IIR";
in this article, this portion would well fit for what Guillaume of LUPISFT informed me;
View attachment 65496

I would really welcome some of you may join this thread to describe about the theoretical, mathematical and physical basics of EKIO in comparison with other software XO solutions, even though the actual algorithm of EKIO's internal processing is not open for public.

In any way, I am always much impressed by EKIO in terms of functionality, simplicity, GUI design, unlimited I/O channel numbers, flexible I/O in full ASIO routing, fast and light CPU processing, very small physical memory consumption, and most importantly its wonderful sound quality including phase features.


IIR induces post ringing (less audable as the sound masks the ringing) FIR minimum causes preringing (ringing sound comes before the actual sound event- more audable), FIR linear phase (perfect). As far as i'm aware, that's basically it. Given the choice, FIR linear is the one you want, but it is only possible with reasonable amounts of computing power. Some MiniDSP units can do a limited amount of it. Most PCs can do all you wish.

As to how audible this is and what the threshold is, I personally have tried all 3 with the same crossover and EQ settings and to me, I am comfortable it is clearly audible, but evidence for this conclusion (double blind tests) are not conclusive as far as I am aware.

To use it, I would recommend JRiver and the 'convolution' filters exported from Pos's RePhase software. (he is a member here). If you use JRiver's EQ filters, they are FIR minimum phase, so I get a 'softening' of the sound, and some strange effects when too much EQ is applied. Subjectively, the effect is much worse with narrow Q filters particularly when I go over 7.
 
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dualazmak

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IIR induces post ringing (less audable as the sound masks the ringing) FIR minimum causes preringing (ringing sound comes before the actual sound event- more audable), FIR linear phase (perfect). As far as i'm aware, that's basically it. Given the choice, FIR linear is the one you want, but it is only possible with reasonable amounts of computing power. Some MiniDSP units can do a limited amount of it. Most PCs can do all you wish.

As to how audible this is and what the threshold is, I personally have tried all 3 with the same crossover and EQ settings and to me, I am comfortable it is clearly audible, but evidence for this conclusion (double blind tests) are not conclusive as far as I am aware.

To use it, I would recommend JRiver and the 'convolution' filters exported from Pos's RePhase software. (he is a member here). If you use JRiver's EQ filters, they are FIR minimum phase, so I get a 'softening' of the sound, and some strange effects when too much EQ is applied.

Hello Lbstyling,

Thank you indeed for your clear-cut comments and suggestions which I understood well.

OK, after I would complete the first stage of this project having multi-channel amplifier(s) and fully eliminating LC-network+attenuators, hopefully in late June, I will carefully test the "post ringing; audible or not?" by using useful tracks of the "Super Audio Check CD", especially track-18 "Transient Check" which is extremely sharp transient sound of a hard-wooden clapper. In case if I would find it audible and better to be eliminated, then I will try JRiver and the 'convolution' filters you kindly suggested.

Thank you again for your nice post which should be also very useful for many people visiting this thread, I believe.

Edited to add:
I just quickly tested with one crossover configuration of EKIO for "post-ringing or pre-ringing; audible or not?", and I found EKIO gives no audible "post-ringing nor pre-ringing" at all in my rather sensitive testing environment. The details of the test will be shared in my next post, hopefully within 6 hours...
 
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dualazmak

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Software crossover EKIO (using IIR filters) gives audible post-ringing or pre-ringing, or not?

After having nice discussion in posts #138, #139, #140, #141 and #142 on filter algorithm of EKIO, today I quickly tested with one crossover configuration of EKIO for "post-ringing or pre-ringing; audible or not?"

Today's Conclusion:
EKIO gives no audible post-ringing nor pre-ringing at all in my rather sensitive test environment.


Test conditions and environment:
I used an EKIO crossover configuration of simple 12dB/OCT LR filter stereo 5-way 10-channel in 192 kHz 24 bit processing which shared in my post #129, shown here again;
index.php

And, the I/O routing is;
WS000568.JPG


As you may aware very well, the catalog spec of DAC8PRO is:
0.000032% / -130 dB full-scale THD (Total Harmonic Distortion)
0.00011% / -119 dB full-scale THD+N (Total Harmonic Distortion + Noise) or 119dB SINAD
125dB dynamic range (20Hz - 20kHz)

For today's "post-ringing or pre-ringing" test, I use my single-amplifier, ACCUPHASE E-460, LC-network+attenuators, to drive YAMAHA NS-1000 + Super Tweeter FOSTEX T915A and Sub-Woofer YAMAHA YST-SW1000 (L & R) as shown in schematic diagram shown in my post #137.

The catalog S/N spec of ACCUPHASE E-460 is:
BALANCED INPUT: 100 dB, -114 dBV (at 180W/ch, 8 Ohm), EIA S/N 93 dB

For the test signal of sharp transient sound, track-18 "transient check" of "Super Audio Check CD" by CBS/Sony released in 1983 was used. Also please refer to my post #26 for the contents of this Auio CHeck CD. This "transient check" sound is a very sharp, high gain, transient sound of hard-wooden clapper. If you are really interested in the contents of this Audio Check CD, please PM me.

I did careful repeated ear-listening and "peak/level meter watching" (of EKIO and E-460) tests in high volume (gain) (of course I was careful enough not to break my SP units) by playing the transient check sound in each of the 10 channels, each pair of the 5 stereo channels, and also all the 10 channels together.

Throughout today's quick but intensive test with one EKIO configuration (10-channel crossover, all LR-type 12dB/Oct slope filters), I found EKIO gives no audible, visible (by peak meters) post-ringing nor pre-ringing at all in my test conditions and environment.

Today's Conclusion is:
EKIO gives no audible post-ringing nor pre-ringing at all in my rather sensitive test environment as shown above.

I assume and believe that EKIO's internal IIR filter algorithm is well designed and carefully tuned/optimized to eliminate or minimize (to under audible level) any possible post-ringing (and pre-ringing).
 
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Lbstyling

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Do you intend to use any peak type filters?

I personally find that the best way to identify the sound was to get the target spl curve as flat as possible using a crossover filter and around 10 peaking filters. Narrow filters are much more audible, so when all this was applied and measured using fIR min filtering I could clearly hear pre ringing. Once I had heard it, I knew what I was listening for more, it was then a case of moving the filters one by one into RePhase, and testing when I could not hear it. Narrow filters were clearly different, but It got debatable when I got to reducing the crossover order (as a LR 2nd order phase corrects as part of the design, this along with a 1st offer filter would be the 2 least audiable filter types in terms of pre/post ringing.) The filters needed adjusting from one system to the next as IIR does not work exactly as software predicts, so needs resweeping and tweeking. FIR measures as close as makes no difference to software prediction once the filter is applied.

Using peaking filters to get the target curve Vs shelf type also made a difference. Shelf filters were better imo.

Basically, I'm saying, if you are only using a LR 12db filter, you may well have no audiable effect. But be careful not to make the conclusion that it does not matter altogether, especially if you later start applying lots of EQ.

I also do not understand how you have tested FIR Vs IIR without using different software? If I remember correct, FIR min phase is the same effect as passive crossovers, so all other things being equal, they should sound the same.
 
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dualazmak

dualazmak

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Do you intend to use any peak type filters?
At this moment, I have no plan to use any peak type filiters.

Basically, I'm saying, if you are only using a LR 12db filter, you may well have no audiable effect. But be careful not to make the conclusion that it does not matter altogether, especially if you later start applying lots of EQ.
Yes, I understand well. You would please read carefully my post #129 entitled " 'First-to-try' EKIO Configuration: the simpler, the better ". My approach and policy in this step-by-step project is first to mimic the LC(coils and capacitors)-Network by EKIO crossover. This is the main reason for I am now using only LR 12dB filter, as described in posts #129 and #130.

The "phase issue" is another important reason for the use of 12dB/Oct filter.
Please refer to my post #31"Phase issues...-1-", #33 "Phase issues...-2-", #37 "Phase issues...-3-" and #39 "Phase issues...-4-" .

I really like and love current sound of NS-1000+T925A+YST-SW1000, and would like to preserve basic "XO structure and feeling" of the total sound after completion of multi-channel system, of course with improved cleanliness, sonority, 3D-perspectives, etc. due to the elimination of LC-network and attenuators.

It would be highly possible that I will not step-up further by applying lots of EQ or using steeper LR filters, if I could establish satisfactory total SQ with such "the simplest is the best" type EKIO XO configuration. I may stay in the status of "the simpler, the better", as far as I would fully satisfy the total SQ.

I also do not understand how you have tested FIR Vs IIR without using different software? If I remember correct, FIR min phase is the same effect as passive crossovers, so all other things being equal, they should sound the same.
So far, I have not yet compared EKIO vs other software. Kees Huizinga suggested me to try DePhonica (see his recent post for me) since using DePhonica we can hear IIR or FIR. He said, however, that "To be honest, I find it difficult to determine the differences. I have not been able to detect the dreaded ringing at FIR with 4th order filters. You have a license for Dephonica, maybe you have a better hearing than me."

I have one license of DePhonica, but had little time to try it, and I also stumbled on the multi-channel output routing from Dephocia to DAC8PRO, as I wrote here. Then I suspended the trial with DePhonica, and decided first to go with EKIO to get my first goal of the project. It may be possible I will try DePhonica later-on after the completion of my first goal with EKIO, DAC8PRO and multi-channel apmlifier(s).

Until yesterday, I have been thinking that EKIO is min-phase FIR, and thanks to your inquiry, I could contacted with Guillaume of LUPISOFT and found that EKIO is IIR filter.

I do not know the reason for LUPISOFT is not describing the "IIR filter within EKIO" in their home page and User Manual; I assume their policy would be not to go into technical details of their product EKIO, but they are, Guillaume is, much confident in EKIO's total functional excellence and sound quality. Maybe, their target customers would be rather middle level audio lovers trying multi-channel XO (just like myself!), and not the high-level enthusiastic, semi-professional(?) experts in audio multi-channel crossover system.
 
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Kees Huizinga

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Hello Dualazmak,
As promised, here's a description of what I did.
Before you start working with Dephonica, it is wise to first watch the three videos about Dephonica on YouTube.
In JRiver choose in Tools, "thePhonica Asio Sink x64 driver" as the audio device. In dePhonica choose DIYINHK ASIO Driver as the output device.
I measure with a Focusrite Scarlett 2i2 and, just like you, with an ECM8000 microphone. (unfortunately not calibrated)
After I have set the filters for all speakers (270Hz, 2100Hz, LR4) in Dephonica(in the video's explained) , I will measure with Holmimpulse. Through an open loop measurement I generate a frequency curve of these filters, so that I know whether what I will measure with the microphone will correspond to the desired filter curves. With the equalizer and shelving filters in Dephonica I try to make the measured frequency curve coincide as much as possible with the desired filters. Working with 3 programs at the same time does not always go well. In any case, do not choose Asio with the measurement programs. JRiver must be set as the default device. A capacitor in series with the tweeter is recommended. For convenience, I measure everything first with 1 sound card. This is to prevent problems with different clocks
With the VituixCAD program, you can calculate the delay of the drivers after measuring with Arta.
I also determine the bafflestep with VituixCAD. You can open saved text files from Measurements with ARTA and Holmimpulse in an equalizer window in dePhonica and you can press the EQ invert button if you want a counter-curve. To expand the bass I use a Linkwitz Transform. I determine the values for this with the WinISD program. I enter the calculated values at the parametric equalizer in JRiver. By means of a closed loop measurement in Holmimpulse I convert those values into a frequency curve. I can enter the resulting text file in DePhonica in an equalizer window. For this I use the Setup pre-processing option of Dephonica (for both channels). I also enter a target curve there. (slightly descending in the high)
To get reliable measurements, it is wise to keep the taps in Dephonica at or below 8000 and not to use FIR, but only IIR, phase corrected if necessary. I also do not apply a resample. This is to keep the latency low.
For serious listening after all measurements I have set all filters to FIR and the output to straight FIR and I use 18000 taps. These taps require a lot of computing power.
In JRiver I use SoX for resampling to 192kHz.
The open loop measurements, so without a microphone, clearly show with Holmimpulse what the influence of the different settings is on phase behavior . (FIR, IIR, IIR with phase correction, FFT and Staight FIR)
With REW I check whether there is still something to be corrected at the listening position.
Hopefully I was able to clarify a bit what I did?
Dephonica offers more options than I wrote down here. As you will understand, I am not an expert, but so far I am very satisfied with the result achieved.
Good luck.

Regards, Kees
 
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dualazmak

dualazmak

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Hello Kees,

Welcome to this thread, and thank you so much for your kind descriptions of your experiences with DePhoinca. I believe your info is also much valuable for many people visiting here.

First, please allow me to ask rather very naive question;
How you can establish multi-channel output from DePhonica into 8 channels of DAC8PRO?

As you may find in this screen capture, I can allocate any of the EKIO's 10 output channels to any of the 8 channels of DAC8PRO;
WS000541.JPG

How can you achieve this kind of multi-channel routing with DePhonica into DAC8PRO? I have first stumbled on this multi-channel output routing with DePhonica, and after that I could not find my free time to try DePhonica. I will highly apprecicate if you could simply paste some screen capture image, with simple explanations, of DePhonica's multi-channel output routing into DAC8PRO

Sorry for my naive inquiry, at first.
 
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dualazmak

dualazmak

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Hello again, Kees,

You wrote; "In dePhonica choose DIYINHK ASIO Driver as the output device. "

OK, this maybe the key to my inquiry, is't it?

I first tried DePhonica before the arrival of my DAC8PRO and its DIYNHK ASIO driver. At that time, I tried to allocate multi-chiannel into OPPO SONICA DAC and ONKYO DAC-1000S usuing ASIO4ALL together with ASIO drivers of SONICA DAC and DAC-1000S, but failed to establish the multiple output.

You mean that after I choose DIYNHK ASIO driver in DePhonica, then I will be able to allocate multi-channel output from DePhonica into DAC8PRO, right?
 
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dualazmak

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Hello Kees, Good morning from Japan,

>That's right.
>And in JRiver the asiosink driver.

OK thanks, I understood well.

My second question is,,,
After you have completed the high-level tuning and configuration in DePhonica, how could you check and/or measure the total phase feature of the sound, especially in comparison with previous non-DePhonica LC-crossover sound?
I am carefully reading the last portion of your post on your phase tuning especially;

>The open loop measurements, so without a microphone, clearly show with Holmimpulse what the influence of the different settings is on phase behavior . (FIR, IIR, IIR with phase correction, FFT and Staight FIR)

In addition to these measurements, do you have your routine ear-listening method and suitable test sound for total phase optimization?
 
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Kees Huizinga

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Hello Dualazmak,
I don't know if I completely understand your question. I measure the phase and frequency curve with programs such as REW, Arta and Holmimpulse. In these programs you choose Jriver as the output device. (Dephonica is also active) In these programs do NOT select Asio, that always gives me problems. Because I measure with a two-channel sound card, I can only measure two speakers at the same time, but then you see how the phase goes. (I choose minimum phase) You can also measure at the same time with the dac8pro (output), but because it concerns two devices, I do not know whether the measurement is very reliable in terms of phase. (different clocks) Frequency curve of course is ok. I measure indoors, so anything below 250Hz is questionable.
In summary, three programs were therefore linked during the measurement: Holmimpulse, JRiver and Dephonica. I keep the number of taps low, because of the latency. In Holmimpulse you can increase the measuring time.
Regards, Kees
 

Nabussan

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Hello Nabussan,
I am still carefully gathering info on multi-channel amplifiers, and in addition to your suggested [email protected] IMG Stageline STA-2000D, Apollon Audio's NCMP8350 is now in my candidate list. It has 4 x Hypex Ncore NC502MP, so 350 W @ 8 ohm for each channel.
As I wrote in my post #104, I will take enough review and investigation time before to finally decide my multi-channel amplifier, and I am very much looking forward to hearing your experiences with STA-2000D.
I will of course share my free trials with rather expensive DENTEC DP-NC400-4 (XLR inputs) which should be a nice lead to my decision on Hypex-based multi-channel amplifier. Looks two of DENTEC DP-NC400-4 (XLR inputs) will arrive at my home in late June.
Hi Dualazmak,

unfortunately, my order of the B-stock 2000D has somehow been canceled, so I could not check the S-Pro2's noise level with my 95dB speakers and report my experiences. Fortunately, TabCam, a fellow multi-channel enthusiast and DIY Audio user, has promised to step in (https://www.diyaudio.com/forums/class-d/350143-img-stageline-sta-800d-post6223565.html).

I was misled anyway because the primary noise source are my miniDSPs, less so my amps. So I have to fix this issue first.

A cost-efficient option for me to do so would be to implement digital DSP plus 8-channel DACs via the miniDSL 4x10HD. But since one of my design goals is to make visually disappear the electronic components, I don't want/need the extra box, so I think I will go for the Auroa freeDSP board (https://www.diyaudio.com/forums/dig...usb-audio-dif-adat-bluetooth-wifi-contro.html) to mount it in on the back of one of my speakers. As you know, it even allows us to play around with FIR filters to a certain extent (though the computational power is if of course very limited). And it has some extra DIY appeal ...

I think the psychological effect of the Dentec NC400 amps option is what cognitive psychologists Daniel Kahneman and Amos Tversky have called Anchoring (https://en.wikipedia.org/wiki/Anchoring_(cognitive_bias). The Dentec seems to be so grossly overpriced that every other option must seem like a bargain. So my bet is you will finally skip the competently designed NCxxxMP-based options and rather opt for something like Audiophonics's Purifi-based 4-channel amps (https://www.audiophonics.fr/en/powe...oies-class-d-purifi-4x400w-4-ohm-p-14452.html). At least this is an option I would consider if I strived for the pleasure of optimum performance. Even four of the highly acclaimed Benchmark amps are cheaper than two of the Dentecs, if I see correctly.

For those of us who choose to fly economy, I have compiled information about S-PRO2 implementations (links that other users have provided in various forum posts).
Anselm Goertz's company FourAudio implements the SPR-2s in their "high-quality DSP-powered" PPA series amps (http://www.fouraudio.com/de/produkte/ppa-serie.html). The only difference with regard to the 1000D/2000D might be an improved input stage.

The 1000D/2000Ds use NJM4558LD OpAmps (see https://www.diyaudio.com/forums/class-d/350143-img-stageline-sta-800d-post6160597.html).
It might be worthwhile to replace them with better ones (http://www.ac-vogel.de/Download/Download/Audio Opamps, fact, no myths and THD measurement results.pdf). But Amrir's JR535 measurements seem to show that such improvements might be futile.
 
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dualazmak

dualazmak

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Hello Dualazmak,
I don't know if I completely understand your question. I measure the phase and frequency curve with programs such as REW, Arta and Holmimpulse. In these programs you choose Jriver as the output device. (Dephonica is also active) In these programs do NOT select Asio, that always gives me problems. Because I measure with a two-channel sound card, I can only measure two speakers at the same time, but then you see how the phase goes. (I choose minimum phase) You can also measure at the same time with the dac8pro (output), but because it concerns two devices, I do not know whether the measurement is very reliable in terms of phase. (different clocks) Frequency curve of course is ok. I measure indoors, so anything below 250Hz is questionable.
In summary, three programs were therefore linked during the measurement: Holmimpulse, JRiver and Dephonica. I keep the number of taps low, because of the latency. In Holmimpulse you can increase the measuring time.
Regards, Kees

Hello Kees,

Thank you again fore your further in-depth info on phase tuning. I would like to test and try DePhonica using some of the measurement tools you kindly described, after I will soon get the first goal of my project with EKIO, DAC8PRO and my own not yet decided multi-channel amplifier(s). Although I am afraid that I will not be able to go up near your high level of the tuning skills, I should learn great lessons from your descriptions. It may be possible I will get back to you for your help and suggestions during such near-future progress.

I believe your recent two posts here in this thread should be very valuable info for many people visiting here.

with my very best regards,

dualazmak
 

pos

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IIR induces post ringing (less audable as the sound masks the ringing) FIR minimum causes preringing (ringing sound comes before the actual sound event- more audable), FIR linear phase (perfect). As far as i'm aware, that's basically it.
Minimum-phase IIR or FIR will have the exact same time response (ie only post ringing).
Linear-phase FIR will cause preringing if a filter is not matched with a complementary filter t form a proper crossover.

@dualazmak, I did not read all the posts here, but what matters it to consider the acoustical response, that is the "natural" response of the driver in addition to the electrical filter.
Getting an acoustical LR filter is typically not simply a matter of using electrical LR filter. To be clear, using electrical LR filters is very unlikely to produce acoustical LR filters. You need to measure what you get, and adjust using EQ and filters to obtain your acoustical target. Simulation software can be used for this.
When FIR can be used then it become easier as targeting a linear phase response is much easier than trying to match phase shifts between drivers, especially with a 3+ way system.
 

Kees Huizinga

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Here is an example of an openloop measurement with Dephonica. (input connected to output)
Blue is FIR an green is IIR. The difference is clear, but if it is audible.....
I wish I was an expert.
 

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dualazmak

dualazmak

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Hi Dualazmak,

unfortunately, my order of the B-stock 2000D has somehow been canceled, so I could not check the S-Pro2's noise level with my 95dB speakers and report my experiences. Fortunately, TabCam, a fellow multi-channel enthusiast and DIY Audio user, has promised to step in (https://www.diyaudio.com/forums/class-d/350143-img-stageline-sta-800d-post6223565.html).

I was misled anyway because the primary noise source are my miniDSPs, less so my amps. So I have to fix this issue first.

A cost-efficient option for me to do so would be to implement digital DSP plus 8-channel DACs via the miniDSL 4x10HD. But since one of my design goals is to make visually disappear the electronic components, I don't want/need the extra box, so I think I will go for the Auroa freeDSP board (https://www.diyaudio.com/forums/dig...usb-audio-dif-adat-bluetooth-wifi-contro.html) to mount it in on the back of one of my speakers. As you know, it even allows us to play around with FIR filters to a certain extent (though the computational power is if of course very limited). And it has some extra DIY appeal ...

I think the psychological effect of the Dentec NC400 amps option is what cognitive psychologists Daniel Kahneman and Amos Tversky have called Anchoring (https://en.wikipedia.org/wiki/Anchoring_(cognitive_bias). The Dentec seems to be so grossly overpriced that every other option must seem like a bargain. So my bet is you will finally skip the competently designed NCxxxMP-based options and rather opt for something like Audiophonics's Purifi-based 4-channel amps (https://www.audiophonics.fr/en/powe...oies-class-d-purifi-4x400w-4-ohm-p-14452.html). At least this is an option I would consider if I strived for the pleasure of optimum performance. Even four of the highly acclaimed Benchmark amps are cheaper than two of the Dentecs, if I see correctly.

For those of us who choose to fly economy, I have compiled information about S-PRO2 implementations (links that other users have provided in various forum posts).
Anselm Goertz's company FourAudio implements the SPR-2s in their "high-quality DSP-powered" PPA series amps (http://www.fouraudio.com/de/produkte/ppa-serie.html). The only difference with regard to the 1000D/2000D might be an improved input stage.

The 1000D/2000Ds use NJM4558LD OpAmps (see https://www.diyaudio.com/forums/class-d/350143-img-stageline-sta-800d-post6160597.html).
It might be worthwhile to replace them with better ones (http://www.ac-vogel.de/Download/Download/Audio Opamps, fact, no myths and THD measurement results.pdf). But Amrir's JR535 measurements seem to show that such improvements might be futile.

Hello Nabussan,

Thank you for your very nice info and comments on the selection of multi-channel amplifier(s) not only for my project but also for any type of multi-channel projects; your info should be, therefore, much valuable for many people visiting here.

I fully agree with your feeling on Dentec NC400 amps especially the rather psychological over decorations and negative cost performances. I would like to just try and hear, however, the sound of NC400 modules as a representative of Hypex D-class amp modules.

I am also very much interested in the discussions in the thread entitled "Are pre-amps necessary"; Although I do not have "scientific evidences", I feel warm sympathy with this post given by MattHooper.
He wrote;
I've tried passive pre-amps before, and also some DACs that offered volume control (e.g. an old Meitner DAC and my current Benchmark DAC).
Whenever I ran a DAC directly in to my amps (monoblock tube amps) instead of whatever pre-amp I had on hand, I was always struck with the sense of clarity and smoothness...the impression of very "direct, uncolored" sound going straight in to the amps to my speakers. Problem is I always gravitated back to an active pre-amp because for whatever reason the direct signal to my amp from a DAC seemed to lack the same body/density of sound and I found it unsatisfying after a while.
Problem is that just remains subjective anecdote and there was no easy way to blind test between the DAC using volume control directly in to the amps vs using my pre-amp.
I am now using a nice integrated amp ACCUPHASE E-460, and I just like and enjoy the total SQ given by E-460.

I have still not yet fully excluded, therefore, the possibility of using four Hi-Fi stereo integrated amps with XLR balance input (fortunately we have several nice products in Japan).

Yes, the selection of multi-channel amplifier(s) should be very difficult process (of course also from budget point of view), but also it is the enjoyable final step to get my first goal in this project. Having your kind inputs and above thoughts, I will carefully and cheerfully try and test several amplifiers before to make my final decision.
 
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dualazmak

dualazmak

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Hello again, Nabussan,

I really thank you for your info on;
https://www.audiophonics.fr/en/powe...oies-class-d-purifi-4x400w-4-ohm-p-14452.html

I also looked at amirm's nice report on PURIFI 1ET400A;
https://www.audiosciencereview.com/...easurements-of-purifi-1et400a-amplifier.7984/
looks very nice and really attractive.

Yesterday, I discussed with one of my audio-enthu friends in Japan on PURIFI 1ET400A, and he kindly informed that Apollon already released 3-channel amp with PURIFI 1ET400A;
https://www.apollonaudio.com/3-channel-purifi-1et400a-apollon-audio/
which looks extremely nice, and the price is 2690 EURO,,, but for my multi-channel project, I need 8 channels, i.e. 3 sets of this product!

I am now seriously considering to contact with Apollon for the possibility of 4 channel unit, or even 8 channel unit, after the current still ongoing COVID-19 status in Japan and Europe would be fully calm down.

In any way, thanks you so much again for your kind info on multi-channel amplifiers.
 
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dualazmak

dualazmak

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Hello friends,

Just for your info, an interesting discussion entitled "What to trust ear or measurement?" is ongoing at;
https://www.audiosciencereview.com/forum/index.php?threads/what-to-trust-ear-or-measurement.13696/

The thread was started by referring;

Although I believe the same kind of discussions have been repeatedly held in this ASR Forum in the past, I still find the new thread is interesting and worthwhile, at least for myself.

Please avoid any discussion on this topics here in this thread; if you like to join the discussion, please visit the specific thread....
 
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Sal1950

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EWWWW not Gutenberry again. :eek:
 
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