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

QMuse

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Hello QMuse,

http://linea-research.co.uk/wp-content/uploads/LR Download Assets/Tech Docs/CrossoverFilters White Paper -C.pdf

and it is well known in many text books that with 12 dB/Oct BW or LR filter, the phase rotation at the cross-point is 180 degree. So if you use series of 12dB/Oct filerts, e.g. series of five ( 5) 12dB/Oct filters to divide the 15 Hz - 25 kHz into five regions (including 15 Hz low-cut), the theoretical phase status in each of the five regions, in general, are;

(SL) reverse - (LO) normal - (MD) reverse - (HI) normal - (SH) reverse

Sure. I missunderstood that you're going to invert 4 out of 5 instead of 3 out of 5.

I still think you should measure phase and see what you're getting. At the end doing that is much easier and quicker than writing such long posts and you will have your measurement as a fact instead of an assumption that it is good enough. :D
 
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dualazmak

dualazmak

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Sure. I missunderstood that you're going to invert 4 out of 5 instead of 3 out of 5.

I still think you should measure phase and see what you're getting. At the end doing that is much easier and quicker than writing such long posts and you will have your measurement as a fact instead of an assumption that it is good enough. :D

OK, noted, thank you.

Please stay safe, and let's keep contacts in this thread, and/or other threads of mutual interests!
 
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dualazmak

dualazmak

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Hello friends,

Just one point about DAC8PRO to share with you... Warming-up needed?

My OPPO SONICA DAC needs about 30 min warming-up before giving its best SQ operation, maybe due to its power supply unit. After my 3-day experience with DAC8PRO, on the other hand, I found it requires no (or less than 10 min?) worming up period before giving best SQ; I believe this would be also very nice and important for professional usage, e.g. at recording studios.

Do you, who is using DAC8PRO, also feel the same?
 

Nabussan

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Hi Nabussan,

I see in your setup you use Bridge Tied Load (BTL) for the subwoofer to get 500W/8ohm instead 2x125W/8ohm from that plate amp. I probably also will use BTL in my system. I ordered a Nord 8-channel amp with several nCore modules. From the Hypex datasheet I learned that the voltage gain will be 6dB in this mode (a rather high 32 dB instead of 26 dB). That's to be expected of course.

I wonder if I have to compensate for that in the DSP (taking the BTL-channel 6 dB down relative to the other channels), besides the phase inversion of one channel. Do you know about that? Or maybe someone else?
Thx

John
Yes, in my understanding everything in modal region (where you use the amps in BTL mode for the Subs) is about EQ, so you do it in the DSP anyway if required.
 

Nabussan

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As described in my post #14, Nord Acoustics’ configurable 8-channel amplifier Nord One MP NCXXX 4-8 122-500W;
https://www.nordacoustics.co.uk/pro...w-custom-configurable-channel-amplifier-black
with;
Channels 1-2: NC252MP (2x250W 4Ohm, 2x200W 8Ohm)
Channels 3-4: NC122MP (2x125W 4Ohm, 2x75W 8Ohm)
Channels 5-6: NC122MP (2x125W 4Ohm, 2x75W 8Ohm)
Channels 7-8: NC122MP (2x125W 4Ohm, 2x75W 8Ohm)
is my present top candidate, and I am gathering various info on Hypex Ncore amp modules, NCxxxMP series.

Prior to place my purchase order for such a configurable 8-channel amplifier, however, I would like to have one careful preparatory step to fully check and confirm the possible multi-channel multi-amplifier system using exactly same 8 (eight) amplifiers which are capable of balanced XLR input from DAC8PRO, and preferably having gain/volume controller, and of course usual protection functions. Since this should be my preparatory step before the final decision on Hi-Fi multichannel amplifier, I need to go through this step within reasonable budget.

After intensive search for possible suitable amplifiers on this step, I am now considering to use two of YAMAHA XM4080, a PA type 4-channel amplifier with balanced XLR input and gain/volume;
https://usa.yamaha.com/products/proaudio/power_amps/xm_series/specs.html#product-tabs
https://usa.yamaha.com/files/download/other_assets/5/334565/xm4180_en_om_e0.pdf
View attachment 58889

With XM4080, the 4 amplifiers can be used completely independently, and if I would use two of XM4080, I may flexibly try to drive my SPs using these independent 8 amplifiers.

Of course I will use DAC8PRO’s volume/gain controller for the “Master Volume” in my whole system and also for relative gain/volume and balance control for each of the 8 channels. The 8 volume/gain controllers of XM4080 in my possible preparatory trials should be also very nice to protect my SPs, and also to simulate my current SP level attenuator settings for SQ, TW and ST.

If I would use two of XM4080, I can draw two options for their utilization, as shown below;
Option-1:
View attachment 59050

or Option-2:
View attachment 59051


I assume that Option-1 would be better choice since two XM4080s would work in almost same level of power consumption.

Just for your info, the sub-woofer (SW) YAMAHA YST-SW1000 has both line level input and speaker level input, and therefore, the branching from WO's speaker terminals into SW's speaker level input is quite OK;
View attachment 58900


Does anyone here have experiences with YAMAHA XM4080?

I could find only one user comment on XM4080 from Germany in 2015;
https://www.thomann.de/gb/yamaha_xm_4080.htm#bewertung
where Christoph442 reported a little concern about the fan noise of XM4080.

On March 25, I contacted with YAMAHA on this, and they quickly responded as; “With recent model/production of XM4080, the fan is quieter than before, and the fan rotation speed is system temperature dependent. Even though it always rotates, in home audio usage, the rotation should be in low rpm, and therefore the fan noise would be almost negligible.”

Furthermore, in case if the fan would be still rather noisy for me, I may replace the fans with low noise, low rpm ones, and/or I may change the setting or sensor for system temperature monitoring.

As for the specs of XM4080, I have a little bit negative feeling about the S/N of 103 dB and Residual Noise of max 73 dBu, while all the other specs, including the Damping Factor of min.100 (8Ohm 1 kHz), looks acceptable for my trial step, I think.

As shown in my first post #1, current single-DAC+single-amp+LC network system is preserved for DSD bit perfect listening, and also as my reference sound system. I may easily switch (go back) to this system by changing the SP cable connections at the "Speaker Cabling Boards";
View attachment 59052
Hi Dualazmak,

another amp option you might want to consider is the 4x500W@4R IMG Stageline STA-2000D, based on Pascal S-PRO2 modules. They are moderately priced.

With a dynamic range >119dB they are rated only slightly below your Dac8-Pro and apparently better than the Yamaha. As per user hege the Pascal modules are, for example, used in the not so crappy D&D 8c speakers (https://www.audiosciencereview.com/...mplifiers-to-test-and-review.2238/post-325598). Rouge Audio in Italy offers a 2-channel Pascal S-PRO2-based amp at >€1100 (the corresponding Stageline 1000D sells at € 380)!

The STA-2000Ds seem to be available only in Europe. But hege was able to purchase it at amazon.de here in Germany and get it shipped to the US ex-VAT (which is 19%) (see https://www.audiosciencereview.com/...mplifiers-to-test-and-review.2238/post-325598). So the same could be possible to get them to Japan (or perhaps get one of them dropshipped to Amir for testing before).

I have ordered a B-stock STA-2000D today (@ € 620 incl. VAT, the regular price is about € 730, see https://www.amazon.de/IMG-STAGELINE...jbGlja1JlZGlyZWN0JmRvTm90TG9nQ2xpY2s9dHJ1ZQ==).

If it works as expected, I will probably buy a second unit plus the minidsp 4x10HD (https://neurochrome.com/pages/minidsp-4x10hd). To minimize the visual Impact of these ugly devices I will build a case and place it vertically next to one of my subs (it's ca. 48 cm/19" deep, as the width of the units). With this solution, I get slightly more power than with the Ncore FA502 modules at comparable dynamic range for less bucks. And I can simply reuse my minidsp 2x4 filter settings without having to bother with new software.
 
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dualazmak

dualazmak

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Hi Dualazmak,

another amp option you might want to consider is the 4x500W@4R IMG Stageline STA-2000D, based on Pascal S-PRO2 modules. They are moderately priced.

With a dynamic range >119dB they are rated only slightly below your Dac8-Pro and apparently better than the Yamaha. As per user hege the Pascal modules are, for example, used in the not so crappy D&D 8c speakers (https://www.audiosciencereview.com/...mplifiers-to-test-and-review.2238/post-325598). Rouge Audio in Italy offers a 2-channel Pascal S-PRO2-based amp at >€1100 (the corresponding Stageline 1000D sells at € 380)!

The STA-2000Ds seem to be available only in Europe. But hege was able to purchase it at amazon.de here in Germany and get it shipped to the US ex-VAT (which is 19%) (see https://www.audiosciencereview.com/...mplifiers-to-test-and-review.2238/post-325598). So the same could be possible to get them to Japan (or perhaps get one of them dropshipped to Amir for testing before).

I have ordered a B-stock STA-2000D today (@ € 620 incl. VAT, the regular price is about € 730, see https://www.amazon.de/IMG-STAGELINE-STA-2000D-4-Kanal-PA-Digital-Verstärker/dp/B00J8LCC8A/ref=sr_1_2_sspa?__mk_de_DE=ÅMÅŽÕÑ&dchild=1&keywords=sta-2000d&qid=1589559033&sr=8-2-spons&psc=1&spLa=ZW5jcnlwdGVkUXVhbGlmaWVyPUFHN0dEOEpTN0UxNzUmZW5jcnlwdGVkSWQ9QTA2NTA0MTYxRzlaMTNKUFJRQ1RIJmVuY3J5cHRlZEFkSWQ9QTA4MjEyOTIyMDNLSlg0R1JPT1lQJndpZGdldE5hbWU9c3BfYXRmJmFjdGlvbj1jbGlja1JlZGlyZWN0JmRvTm90TG9nQ2xpY2s9dHJ1ZQ==).

If it works as expected, I will probably buy a second unit plus the minidsp 4x10HD (https://neurochrome.com/pages/minidsp-4x10hd). To minimize the visual Impact of these ugly devices I will build a case and place it vertically next to one of my subs (it's ca. 48 cm/19" deep, as the width of the units). With this solution, I get slightly more power than with the Ncore FA502 modules at comparable dynamic range for less bucks. And I can simply reuse my minidsp 2x4 filter settings without having to bother with new software.

Hello Nabussan,

Thank you for your very nice info on possible multi-channel amplifier for my project. I have been really expecting such valuable input in this thread for my selection of multi-channel amplifier.

I am still carefully gathering info on multi-channel amplifiers, and in addition to your suggested 4x500W@4R 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.

Fortunately, my sub-woofers YAMAHA YST-SW1000 is an active type, and it has powerful dedicatred amp in it, so at present I have no need to worry about powerful amp for the sub-woofers.

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.

Until then, I continue to fully test EKIO and burn-in all the 8 channels of DAC8PRO. As I also wrote here, the latest DAC8PRO is amazingly nice.
 
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dualazmak

dualazmak

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An interlude to the next curtain…

It is a kind of big surprise for me that this thread has been visited and viewed more than 8,300 times since its start on April 9, i.e. during only about 5 weeks.

I assume many people visiting here are also interested in considering/planning same kind of pure Hi-Fi audio multi-channel project as mine, and find this thread somewhat suggestive and informative. Let me continue, therefore, to share and discuss my step-by-step progress in this project.

I hope this interlude post would be somewhat helpful for people visiting here rather recently who have little time to read through the entire thread from the very beginning.

The objective of this project is simple but unique… I have my present Hi-Fi stereo 2-way audio dedicated (not AV) single-amplifier 5-way speaker (SP) system (3-way YAMAHA NS-1000, super tweeter FOSTEX T925A and active sub-woofer YAMAHA YST-SW1000), and I would like to replace the existing “single DAC + amplifier + LC-network + attenuators” with “software digital crossover + multi-channel DAC + multi-channel amplifier”, to drive each of the SP units directly by dedicated amplifier; all the SP units should remain as they are now also in the coming multi-channel system.

My first priority in this project is to achieve overall sound quality (SQ) improvement by eliminating the “LC-network + attenuator”, and the overall sound should be properly compared “before and after” even after the completion of multi-channel system, so I should have an easy physical mechanism to switch back and forth between the two systems having the common SP units for overall sound comparison. This is the main reason that I need to preserve the existing “single DAC + amplifier + LC-network + attenuators” as my standard reference sound system.

The software crossover, which is currently EKIO in this project, processes all the sound signal in 192 kHz 24 bit (or less) PCM, and it cannot handle 1 bit DSD formats i.e. 1xDSD (DSD64), 2xDSD (DSD128) or 4xDSD (DSD256). The music players, e.g. JRiver and Roon, need to down-sample these DSD music tracks into 192 kHz 24 bit PCM to put them into EKIO through ASIO routing. This of course means that we will not be able to hear the DSD native bit stream sound in the coming multi-channel system, even though the overall SQ in 192 kHz 24 bit is also very nice and usually acceptable for my ears.

After the completion of my multi-channel system, however, I believe that I would occasionally like to hear/listen the native bit stream DSD sound using ASIO and DAC capable of up to 4xDSD (DSD256) or 8xDSD (DSD512). This is another reason to preserve the existing “single DAC (OPPO SONICA DAC) + amplifier + LC-network + attenuators” driving exactly the same SP units/system. OPPO SONICA DAC is capable of up to 8xDSD (DSD512), and each of the 8 channels of DAC8PRO is capable of up to 2xDSD (DSD128), both by direct ASIO routing (without EKIO crossover).

WS000551.JPG


I fully understand that this is ASR Forum, and various objective measurements are really major concern of several people visiting here. In this context, I have already installed, tested and somewhat validated the software REW, especially its REW-Wavelet analysis in line level signal and also in real sound using ECM8000 measurement microphone, as already shared in my post #16 through #24. I would like to try the similar (and further) objective measurements after having the multi-channel amplifier(s) which I am still waiting for the arrival.

I hope you may understand and agree with me, that the subjective ear-listening comparison of the overall sound “before and after”, as I wrote in above paragraphs, should be also very much important in this specific project using exactly the same SP units and alignments.

Now DAC8PRO has arrived, but I am still carefully investigating various info on possible multi-channel amplifier(s) as I wrote in my post #55, #88, #104 and #127. Furthermore, I will be able to try rather expensive DENTEC DP-NC400-4 (XLR inputs) hopefully in late June. I will decide the selection of multi-channel amplifier(s) in this project after my free trials with DENTEC DP-NC400-4 (XLR inputs).

Until then, I would like to fully test EKIO’s possible crossover configurations using DAC8PRO and single-amplifier system, and this will also simultaneously burn-in each of the 8 channels of DAC8PRO, although it looks intensive burn-in would not be needed for DAC8PRO.

Several my posts on trials/tests with EKIO and DAC8PRO will follow this post, hopefully during this week-end.
 
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dualazmak

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"First-to-try" EKIO Configuration: the simpler, the better

On top of this post, let me share with you this nice document on basics of crossover filters entitled “Crossover Filter Shape Comparisons : A White Paper from Linea Research” by Paul Williams, dated July 2013;
http://linea-research.co.uk/wp-content/uploads/LR Download Assets/Tech Docs/CrossoverFilters White Paper -C.pdf


Well, as I wrote in my first post, we have the following "four major issues” always associating with software (and also with hardware) crossover solutions;

- Digital I/O (full ASIO) for software crossover within PC
- Delay, latency and synchronization issues
- Phase issues (no single best solution we have...)
- Master volume and gain issues (including "where to do these?")


We know that these issues are always somewhat interdependent with each other.

Throughout this thread from the start on April 9 to today, we have already shared and discussed a lot about each of these major subjects, and at least for my current project and my environment, I could reach the currently optimal (but still not perfect though) "first-to-try" EKIO crossover (XO) configurations.

The "policy and approach" for such "first-to-try" EKIO configurations are as follows;

- the XO point (Fq) and the slope should first mimic the current LC(coils + capacitors)-XO-network circuit of my reference sound system,

- All of the XO filters should be 12dB/Oct slope LR (Linkwitz Riley) type, including the low-cut for 15 Hz, so that the theoretical phase status in five Fq regions would be;
(SL) reverse - (LO) normal - (MD) reverse - (HI) normal - (SH) reverse


- no delay setting in EQ filter and no group delay in each output channel,

- in general, no relative gain setting/control in each output channel,

- if Fq regions for tweeter and super tweeter would widely overlap, then give about -4 dB gain to the both,

- the input level to EKIO should be just below the clipping level to minimize the possible bit-loss,

- the total line level Fq response of EKIO should be almost flat throughout 20 Hz to 25 kHz,

- the output level (gain) of each of the channels should be as high as possible to minimize bit-loss in DAC processing by DAC8PRO,

- the master volume and relative gain for each channels (as well as the L to R balance) should be controlled by DAC8PRO's volume & gain controller (to mimic current attenuator settings).

The following points and reasons are staying behind the above "policy and approach";

- any delay setting in software XO inevitably brings more or less "phase complications",

- the relative 3D alignment of WO (woofer), SQ (squawker) and TW (tweeter) of YAHAMA NS-1000(rev) is tightly fixed in the solid cabinet by YAMAHA's miracle design to give the best 3D phase features (and I trust it), so no relative delay setting is given to these SP units, at least for the "first-to-try" trials,

- the SW (sub-woofer) YAMAHA YST-SW1000 has its own phase inversion switch and can be switched "normal or inverted", by a remote controller,

- the SWs also have variable hi-cut filter (24 db/Oct) and volume, both also controllable by a remote controller,

- L and R SWs can be physically moved within 40 cm region back-and-forth for fine delay tuning, if needed,

- STs also can be moved back-and-forth, but current rather unique position has been fixed through intensive listening sessions, and they should remain at that position.

Consequently, the "first-to-try" EKIO XO configuration should be;
WS000540.JPG


Please note that the phase "invert" is checked only in SL (super low) and MD (mid Fq) regions to give rather flat and smooth phase response in real sound, depending on the theoretical phase features of;
(SL) reverse - (LO) normal - (MD) reverse - (HI) normal - (SH) reverse
with the series of 12 dB/Oct LR filters including the 15 Hz low cut.

Also please note and remember that TWs and STs are inverted at their SP cable terminals.

YAMAHA intentionally inverted to TW, and I can easily understand and agree with it by listening to only the TW stereo sound using the phase check track of the "Super Audio Check CD"; YAMAHA selected the inverted connection through their very careful and intensive listening and measurement trials. I also intensively tested and measured FQ response curve and REW-Wavelet for ST in normal or inverted connection, and finally decided to invert to ST also.

And, I/O routing should be;
WS000541.JPG

As wrote in my post #12, the main reason for keeping the super-low (SL) 15 - 50 Hz channels in this EKIO I/O configuration is that in a few music tracks in my library ripped from CD or digitized from LP records, unpleasant small volume super low Fq noises exist around 20 - 35 Hz caused by hall air conditioning or by LP records' bending which can be muted (if needed) by the Mute buttons of SL-L and SL-R Output panels. Except for these rare "super low Fq noise" cases, of course I usually need 15 - 50 Hz sound mainly played by sub-woofers.

Since I am still waiting for the arrival of multi-channel amplifiers for my free trials, I intensively test this EKIO configuration using any of the two channles of DAC-8PRO and single-amplifier+LC network system through this output routing;
WS000542.JPG

You may easily understand that using this output routing, with using CH-1 and CH-2 of DAC8PRO, I can hear the total sound of EKIO XO settings, and also I can hear any one of the 10 channels using the Solo or Mute buttons at the output panels. Of course, I can hear any combination of the channels out of the 10 channels. This is a very nice flexible feature of EKIO, indeed.

By doing this, I am burning-in the CH-1 and CH-2 playing non-stop my audio sampler play list by JRiver or Roon. I can do the same with CH-3 + CH-4, CH-5 + CH-6 and CH-7 + CH-8, so that I may burn-in all the 8 channles of DAC8PRO.

Then what and how is the overall sound quality (SQ) with this "first-to-try" EKIO configuraion using DAC8PRO?

I am carefully comparing each of the 10 XO channels, as well as the all the channels together, with the non-EKIO sound, all in 192 kHz 24 bit. I also compare the EKIO sound (192 kHz 24 bit) with non-EKIO 2xDSD (DSD128) sound.

The overall SQ with this EKIO configuration is found to be just the same as the non-EKIO sound, both using DAC8PRO, in all aspects including S/N, sonority, cleanliness, 3D phase status (over 15 Hz to 20 kHz) and sound resolution, by using/listening the tracks of "Super Audio Check CD" and my "audio sampler play list".

I am very happy that I could re-confirm, this time with using DAC8PRO, that EKIO's XO functionality and overall SQ is really amazingly nice even though its processing is 192 kHz 24 bit.
 
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dualazmak

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"Second-to-try" EKIO Configuration trusting EKIO to Mimic Attenuator Settings

Yesterday, in my post #129, I shared and discussed the "first-to-try" simplest EKIO configuration with no delay nor channel gain in the configuration where, after the arrival of multi-channel amplifier(s), I need to ask DAC8PRO to control the relative gain between the 8 channels to mimic current attenuator settings.

This means that DAC8PRO and myself should always precisely memorize the relative gain settings between the channels, and such gain settings should be applied in DAC8PRO's 8-channel gain controller every time when using DAC8PRO in my system. It should be much more convenient and stress-free, however, if I could use DAC8PRO just only for Master Volume Controller forgetting about relative gains, i.e. no relative channel gain set in DAC8PRO.

Now, all of us well aware of amazingly nice S/N and linearity of DAC8PRO, and we can trust DAC8PRO for excellent sound quality (SQ) even with low gain (small volume) sound signals. Furthermore, in post #60, Burning Sounds kindly pointed and informed that "If I remember correctly EKIO is done in a 64bit environment (you can do considerable attenuation on any channel and not suffer quality loss) and has peak level meters for each channel - these should enable you to check that you are not running into clipping. "

So, why do not trust EKIO, other than DAC8PRO, to mimic the present attenuator settings in EKIO's gain control of each output channel?
OK, I now aware that I should try it as my "second-to-try" EKIO configuration, and I actually prepared and tested it today.

In my present reference Single-DAC+Single-Amplifier+LC-network+Attenuator system, the atteuator setting for squawker (SQ), tweeter (TW) and super tweeter (ST) is;
WS000554.JPG


This setting was fixed last summer through intensive listening and measurement sessions after the renovation of my YAMAHA NS-1000 including the new capacitors, coils and attenuators in "outer box" as shown in my post #5. The ST FOSTEX T925A is a very high efficient unit, and the -13 dB attenuation found to be optimal.

If EKIO can mimic this attenuator setting, and if I would like to hear the sound using the same single-amp system, then I should change these as;
WS000555.JPG

And, the possible EKIO I/O Panel setting should be;
WS000556.JPG

Please note that we have -5 dB gain (attenuation) in MD and HI channels, -13 dB gain (attenuation) in SH channels. This screen capture was taken while playing the track-12 flat pink noise of "Super Audio Check CD".

With these gains, overall Fq response, total phase, group delay, etc. are;
WS000557.JPG

WS000558.JPG


The total phase rotation is still simple enough, and the theoretical group and total delay are below 10 ms throughout 30 Hz - 20 kHz.
Using this "second-to-try" EKIO configuration into CH-1 and CH-2 of DAC8PRO, as well as only Master Volume control by DAC8PRO, I did really intensive "before and after" listening sessions today including;

- Phase check while listening to only each of the Fq regions, i.e. SL, LO, MD, HI and SH, using the track-2 "Phase Check",
- Phase check while listening to all the channels at once, using the track-2 "Phase Check",
- Five-location (positioning) check while listening to only each of the Fq regions, using track-3 "Five Location Check",
- Five-location (positioning) check while listening to all the channels at once, using track-3 "Five Location Check",
- High Fq linearity check while listening to only HI and/or SH, using track-20 "High Frequency Linearity Check",
- High Fq linearity check while listening to all the channels at once, using track-20 "High Frequency Linearity Check",
and
- intensive "before and after" listening to the tracks of my audio sampler play list.

I found that the phase status of each region is just OK as "before", five-location (positioning) of each region is also OK, the high Fq linearity is just really fine.

After the intensive "before and after" listening sessions with my audio sampler play list, I feel this "second-to-try" EKIO configuration gives slightly better overall SQ to my ears and brain, in terms of cleanliness, sonority, 3D perspectives, and sound resolution, maybe due to the less or no stress by the attenuators in present system.

In conclusion, this "second-to-try" EKIO configuration having the "mimic of attenuators" would be another nice candidate to try after the arrival of multi-amplifier(s). It should be very nice that I can use DAC8PRO's master volume controller only with no relative gain setting in DAC8PRO for the 8 channels. We can trust EKIO's gain control, i.e. attenuation, on any channel and not suffer SQ loss, as Burning Sounds kindly suggested in post #60.
 
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dualazmak

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I reported the contents of this post here and here. I hope it would be allowed to report the same in this thread.

My DAC8PRO arrived on May 9, and I have been using/operating it more than 12 hours everyday, all of the 8 channels in Pure USB Mode.

Yesterday, after about 8 hours non-stop operation, the status was;
index.php

Room temperature yesterday here was 20 degree C. My DAC8PRO, sitting on top of my audio cabinet, was working perfectly in Pure USB mode, 192 kHz 24 bit operation in all of the 8 channels. The top cover temperature is always well below my "healthy" body temperature, so the 26 degree C indication is just OK, I believe. Fortunately, I have no problem at all including the remote. My firmware is 1.32 as show by the above photo.

The electricity here in Chiba Prefecture, Japan, is 100V 50Hz. For the 3-pin power cable, I connected the earth-pin to my proper audio dedicated ground-earth by earth (usually green) cable. The ground-earth is common for my PC, amplifiers, DAC8PRO, OPPO SONICA DAC, CD player, LP Player, etc. Within DAC8PRO, at least the ground-earth (yellow-green cable) is connected to the metal base panel under the circuit boards, and to the cabinet.

DAC8PRO and Apple remote are working just amazingly nice with ZERO issues since its arrival on May 9.
 
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dualazmak

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Useful Public Domain Documents and Books for Basic Understandings on Audio Crossover

I assume it would be very nice to share links to Useful Public Domain Documents and Books for Basic Understandings on Crossover.

https://en.wikipedia.org/wiki/Audio_crossover

In my post #120 and #129, I touched on this nice PDF document;
http://linea-research.co.uk/wp-content/uploads/LR Download Assets/Tech Docs/CrossoverFilters White Paper -C.pdf
which is very nice for having basic understandings on various filter types, i.e. Butterworth (BW), Bessel, Linkwitz-Riley (LR), Hardman and Linear Phase (LIR) filters. I learned a lot especially about the "Phase Issues".

Yesterday, one of the Senior Member of ASR and great contributor in audio science kindly informed me this document;
http://www.acourate.com/freedownload/XOWhitePaper.pdf
Yes, I know this is an introduction to Acourate(TM) Audio Toolbox. I found the contents are really nice for my basic understandings on crossover.

Many of you visiting here are already aware of this famous one;
https://www.amazon.com/dp/B01FURPS40/#customerReviews

And, not only about crossover but also about all aspects of acoustics;
https://www.amazon.com/Master-Handbook-Acoustics-Sixth-Everest-ebook/dp/B00O2A7GYW/ref=sr_1_1?dchild=1&keywords=Master+Handbook+of+Acoustics&qid=1590153537&s=digital-text&sr=1-1

I highly appreciate if any of you would add and share any (but not so many, please) Useful Public Domain Documents and Books for Basic Understandings on Crossover.
 
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dualazmak

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Hello friends,

Just for your reference,,,
Pavel of Okto Research wrote important update information on DAC8PRO at here, and I responded to it here.
 
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Hello friends,

I am intensively testing various EKIO configurations, and also evaluating stability of EKIO and DAC8PRO in 192 kHz 24 bit processing; so far so good, I found no issues/problems, and they are giving wonderful and amazing sound quality even though still using single-amplifier plus LC-network system.

As I wrote in my post #104, I will soon (in late June) try and test two of DENTEC DP-NC400-4, Hypex NC400 (four NC400 in one amp) stereo 2-way 4-channel Digital Amplifier; I should be physically well prepared, therefore, for such trials using many SP cables and eight (8) XLR balance cables.

Since I assume many of you are interested in actual SP cable connections in my current project diagram shown below, I share several self-explanatory photos...
WS000536.JPG


WS000565.JPG



View attachment 059A8188_r1_trm.jpg

View attachment 059A8180_r1_trm.jpg

059A8226_r1rs.JPG


Please refer to;
my post #4 for SP cabling boards,
my post #5 for SP alignment and outer LC-network+attenuator box,
my post #27 for the unique positioning of super tweeter (ST),
my post #28 for SP cables and tin plated copper Y-lugs.
my post #84 for differences between YAMAHA NS-1000 and NS-1000M

The SP cables shown in above photos are AWG12 (3.5 sq mm), and I wrote in my post #28; "After my 30-year try-and-error journey, now I definitely believe that these very cheap tin plated pure copper Y-lugs and R-lugs are the best terminal-contact solution since rather soft metal tin plate effectively increases the contact surface area when tightly connect and squeeze." At some time point,I have decided never to use banana plugs for SP cable connections to the binding terminals.

The XLR cables I am using are reasonably priced 1.5 m Belden 1192A + gold plated NEURTIK plugs, very carefully soldered by skillful hands at one Japanese audio craft shop.
 
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Lbstyling

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Hello friends,

I am intensively testing various EKIO configurations, and also evaluating stability of EKIO and DAC8PRO in 192 kHz 24 bit processing; so far so good, I found no issues/problems, and they are giving wonderful and amazing sound quality even though still using single-amplifier plus LC-network system.

As I wrote in my post #104, I will soon (in late June) try and test two of DENTEC DP-NC400-4, Hypex NC400 (two NC400 in one amp) stereo 2-way 4-channel Digital Amplifier; I should be physically well prepared, therefore, for such trials using many SP cables and eight (8) XLR balance cables.

Since I assume many of you are interested in actual SP cable connections in my current project diagram shown below, I share several self-explanatory photos...View attachment 65105

View attachment 65119


View attachment 65110

View attachment 65111

View attachment 65112

Please refer to;
my post #4 for SP cabling boards,
my post #5 for SP alignment and outer LC-network+attenuator box,
my post #27 for the unique positioning of super tweeter (ST),
my post #28 for SP cables and tin plated copper Y-lugs.

The SP cables shown in above photos are AWG12 (3.5 sq mm), and I wrote in my post #28; "After my 30-year try-and-error journey, now I definitely believe that these very cheap tin plated pure copper Y-lugs and R-lugs are the best terminal-contact solution since rather soft metal tin plate effectively increases the contact surface area when tightly connect and squeeze." At some time point,I have decided never to use banana plugs for SP cable connections to the binding terminals.

The XLR cables I am using are reasonably priced 1.5 m Belden 1192A + gold plated NEURTIK plugs, very carefully soldered by skillful hands at one Japanese audio craft shop.

Does EKIO support linear phase FIR convolution? Or is it min phase FIR?
 
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dualazmak

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Does EKIO support linear phase FIR convolution? Or is it min phase FIR?

Hello Lbstyling,

I am contacting LUPISOFT to confirm it for sure.

While waiting for LUPISOFT's response to me on this specific inquiry, let me share with you some useful posts in the thread of "Reasonably priced good quality 6 or 8 channel USB DAC?" Me, as a beginner in multi-channel crossover system, decided to start with EKIO based on these suggestive posts well fit for my policy and approach of "the simpler, the better" and "the simplest would be the best" in my project, and also I can intensively and carefully compare the EKIO crossover sound with my current "single-DAC + Single Amplifier + LC-Netwark" sound, step-by-step.

Although I assume you are also aware of the following posts, these would be also somewhat useful for many people visiting this thread;

https://www.audiosciencereview.com/forum/index.php?threads/reasonably-priced-good-quality-6-or-8-channel-usb-dac.4986/post-112825
>Me personally I like direct convolution the best, and only use FIR even for purely minimum-phase corrections.
I like the brutal nature of this most simple algorithm


https://www.audiosciencereview.com/forum/index.php?threads/reasonably-priced-good-quality-6-or-8-channel-usb-dac.4986/post-112838
>Usually FIR is used for linear phase response but the more I'm reading about it, linear phase with pre-ringing is not a good thing at all. It is more audible and unnatural than natural minimal phase.
https://www.meldaproduction.com/tutorials/text/equalizers
(nice reference tutorial document!)

https://www.audiosciencereview.com/forum/index.php?threads/reasonably-priced-good-quality-6-or-8-channel-usb-dac.4986/post-112849
>Phase shift audibility is one thing, and it is indeed subtle (as many things hifi ;)), but FIR is certainly not limited to building linear-phase crossovers! FIR can do anything IIR can, with less quantization errors, and can do many other things like very complex EQs without penalties (the more IIR biquads you add the more errors you get, whereas in FIR only the final response matters), specific crossover slopes, steep (brickwal if you like), directivity-aware slopes (Horbach Keele), etc.
>Of course EQ should always be minimum-phase (but you can do this in FIR no problem ;)), linear-phase EQ are a nonsense as far as sound reproduction is concerned.
>That said a multi-way loudspeaker is not a minimum-phase device, because of the crossovers, and these can be addressed with FIR.


https://www.audiosciencereview.com/forum/index.php?threads/reasonably-priced-good-quality-6-or-8-channel-usb-dac.4986/post-112890
>As for off-axis ringing differences. I did a quick listen to off-axis simulation with 96dB/oct LR with IIR vs linear phase FIR and the FIR off-axis ringing sounded quite different from the IIR off-axis ringing. Everybody can hear this it was quite obvious. I did not test which one I prefer or if one is more audible than the other. But again logic tells me that pre-ringing is something quite weird, you will hear ringing BEFORE a sound is made. For instance a percussion sound.. That's not natural or correct to me. (It's different for a DAC where the pre-ringing is out of the audible range)
>So I very much stand by what I said. I think it's an error to use linear phase EQ for crossovers.
>As for the problems of IIR. They're not so big as I understand. And nothing which isn't fixed in more properly programmed IIR filters, 64bit and with either compensation programmed in for non 0 Nyquist responses in both phase and amplitude (like for instance Equilibrium has and several other plugins) or by oversampling.

https://www.audiosciencereview.com/forum/index.php?threads/reasonably-priced-good-quality-6-or-8-channel-usb-dac.4986/post-112915
>There is no pre ringing in a minimum-phase filter.
>As far as FIR is concerned, you of course need to truncate the response, so nothing infinite there ;)
>IIR is infinite, and errors do pill up as it goes.
>For the specific case of linear-phase FIR filter, there is no "latency vs quality" trade-off per se: what you get with too short FIRs is linear distortion, ie a magnitude and/or phase response that does not match the target.

>This can all be simulated and the choice of a given length for a given correction will give this or that amount of deviation, also depending on the chosen windowing algorithm

https://www.audiosciencereview.com/forum/index.php?threads/reasonably-priced-good-quality-6-or-8-channel-usb-dac.4986/post-112935
>More seriously IIR can be made to work, of course, but there are other advantages to FIR outside sound quality (algorithmic "purity") or phase-linearity: arbitrary magnitude and phase response, predictability, simplicity.

https://www.audiosciencereview.com/forum/index.php?threads/reasonably-priced-good-quality-6-or-8-channel-usb-dac.4986/post-113082
>My biggest objections against linear phase are theoretical regarding the pre-ringing, and latency. I will be using my speaker system for making music as well as listening to it. I could make 2 presets, one for making music one for listening and editing etc where latency isn't an issue.
>But I still think minimal phase is the "right" way to do things. Be it 24dB/oct LR4.

https://www.audiosciencereview.com/forum/index.php?threads/reasonably-priced-good-quality-6-or-8-channel-usb-dac.4986/post-113231
>Received a reply from Okto Research.
>Asked if it was ok to share online and it was.
>My original question was if it was possible for a custom build of the Okto DAC8 + Okto PSU + MiniDSP USB streamer in a case with a volume control pot on front for the ES9028 digital volume. Also asked if there was perhaps an alternative USB to i2s interface they would recommend over the USBStreamer. The reply was:
>"the unit you described seems very much like the one we are preparing to unveil in about a month :). Apart from the USB input (not sure yet if USBStreamer or our own design) it will feature OLED front panel with rotary knob and remote control. We put a lot of effort into the front panel design. I would suggest you to wait for that if you are interested. The price will be roughly around 1000EUR.
>Apart from the USBStreamer, the only multichannel USB to I2S module we know about is the one from DIYINHK which we haven't tested."
>I further asked if they have a linearity measurement of the DAC8 and how the DAC performs at higher sample rates. To which the reply was:
>"We didn't take a linearity measurement yet, but I'll remember to take it as soon I get to the audio analyzer again.

>There was basically no performance difference between 44.1kHz an 96kHz sample rate. At 192k, I remember that increase of THD/THD+N observable - I've searched through the measurements we've taken and found one of a 10kHz tone at 192kHz sample rate showing -122.7dB total THD instead of -124 to -130dB achieved at lower rates. Again, I'll measure more at the next opportunity."

It would be possible for me that I may also try to use Acourate(TM) Audio Toolbox after I could fully establish my current project with EKIO and DAC8PRO.

I will get back to you soon in detail after I could hear and confirm from LUPISOFT regarding your inquiry of "Does EKIO support linear phase FIR convolution? Or is it min phase FIR?".
 
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dualazmak

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Does EKIO support linear phase FIR convolution? Or is it min phase FIR?

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;
WS000567.JPG


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.
 
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