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

Hello again Gene,

>you considered using one on each side of the driver

Do I understand correctly you mean that "Have you tried to connect capacitor directly to the SP unit's terminal inside the cabinet of NS-1000?". If so, I have not yet tried this method since now I am a little bit afraid of having further DIY inside the cabinet for possible unexpected damages, physically and/or electrically, to my treasure SP units....
 
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By symmetric loading I mean two series capacitors, of double the value (since when you series them the voltage rating increases additively and the capacity of each is halved thus make two in series ie 20 uf for the tweeters equal to your 10 uf single capacitor) and put one on either side of the inductive tweeter voice coil. I will make a sketch and shoot a picture of it.

In the past I have used series fuses, never series caps but I think I shall add them soon and use the above method. and fuses too if not some manner of DC detection relay. Since I have a stack of ancient amps.
 
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OK, understood well, and looking forward to seeing your schematic sketch.

You mean series of two 20 uF capacitors;

1/Call = 1/C1 + 1/C2 --> 1/10 = 1/20 + 1/20

C1 to be on the cabling board, and C2 to be directly connected to terminal of Be-TW Unit and ST Unit, right?

What kind of benefit and improvement are you expecting with this method in comparison with just one 10 uF in the line?

I am using 10 uF capacitors just for protection purpose, and therefore my plan is that the capacitors will be removed after I would fully decide the dedicated amplifiers for Be-TW and ST. The same idea also for the 68 uF protection capacitor for Be-SQ.
 
The "improvement" is unlikely to be audible. It's just the way I would do it. I think the presence of the capacitors probably is not audible too. Since you have installed bypass switches you can do blind testing with your Mrs. operating the switches for you.

I am inclined to install and leave blocking capacitors unless you notice them degrading the sound. Your circuit analysis is correct.
And I would do the same for the SQ. (140uF)

Tweeter Circuit.jpg
 
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OK, I got your idea, very interesting even though I assume we will not "hear" the difference... I may physically try your idea easily on my SP Cabling Boards, as you see them.

However,,,, if I would try these in my current system, I need to purchase 8 of 20 uF cap (for Be-TWs and STs) and also 4 of 136 uF cap (for Be-SQs).

As the audio signal actually goes through these (protection) capacitors, I always feel comfortable to select rather high-grade film caps for this purpose.

In February, I purchase these 68 uF and 10 uF caps from Parts Connextion in Canada through their web shop since they were considerably cheaper than to buy at audio shops in Japan;
WS000636.JPG


These Jantzen Audio Standard Z-Cap 400 VDC series are very nice audio caps as you may agree, but even the 68 uF ones are really big and somewhat expensive.

I can find 22 uF (two to make 11 uF) of Jantzen Audio Standard Z-Cap 400 VDC with reasonable price of USD 18.00;
https://www.partsconnexion.com/Jantzen-standard-cap.html
but I cannot find 136 uF one in this series; the biggest one is 100 uF USD 81.77 !!

In Jantzen Cross Cap series, we have Jantzen Cross Cap, 120.00µF (two to make 60 uF) 400VDC, USD 44.87;
https://www.partsconnexion.com/Jantzen-cross-cap.html
Maybe, this should be OK for Be-SQ protection, I assume.

Another cap brand I like is Mundorf, and in Mundorf MKP Series Film Capacitors, we have Mundorf Capacitor 100.0uF (two to make 50 uF) 250Vdc MCap® Classic (MKP) USD $54.04;
https://www.partsconnexion.com/mundorf-mkp-series.html

I am currently using 4 (four) of this Mundorf Capacitor 47.0uF 250Vdc MCap® Classic (MKP), two for each of L & R woofer network, two in parallel to make 94 uF, and they are working very nicely.

Even though I have not yet fully decided to try your idea in my system, do you have any suggestion for selection of enough high quality film caps?

Well, if possible, I would like to ask you to try your idea first to see feasibility (or any benefit, improvement in sound quality), since I already have my high-quality protection caps in my circuit.

I am a little bit worried about that, "in principle", you would double the possible mal-affection, even though not audible, of the caps with your idea which may be overwhelming the possible benefit for symmetric operation of amplifier.
 
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I am not "up to date" on capacitors. But I think I would be cautious about buying overpriced capacitors. I don't think they are necessary.

Some people spend so much on capacitors to upgrade their high level crossovers that it would be more economical to biamp or triamp. (totally different use case)

I have not yet used such capacitors for protection (neither single nor double) but I think I will since I don't want to shop for more beryllium drivers.
 
An Interlude/Intermezzo again, in the Project:

Further Insights on SP Attenuators and their Elimination in Multichannel System


Hello friends,

While waiting for next multichannel amplifier(s) trial session, I took further insights on the SP attenuators (ATTs) in my reference system and their elimination in the multichannel system. This is a follow-up post after my post #236.

As shown repeatedly in this thread, the schematic diagram of my present reference audio system with details of LC-network and SP ATTs is;
WS000775.JPG


And I tried to fully eliminate the ATTs from the circuit in this way;
WS000774.JPG


Here all the ATTs were fully removed from the circuit, and now software crossover EKIO successfully mimics the ATT settings, i.e., -9dB for Be-SQ and Be-TW, -13dB for ST. (I will touch on later in this post regarding the rationales of these gain values.)

In this full removal case, however, slight degradation of total sound quality was audible probably because of the change in impedance of the circuit by the complete removal of ATTs. I could hear slightly artificial, techno-like sound in comparison with my nice reference sound with working ATTs.

Then I tried another way of fully bypassing all the ATTs, but ATTs themselves are still recognized by amplifier, in this way;
WS000773.JPG


As shared in my post #234, #235 and #236, really nice sound quality, same as my reference sound with working ATTs, was achieved in this method.

During my very careful listening session with this scheme, I found, that the Be-SQ, Be-TW and ST are still very faintly responding to the change of the ATT values of the "bypassed" ATTs; why??

Then my old memory suddenly flashed back to me telling "Looking from the amplifier, the resistance (impedance) of ATT is not constant but it varies depending on the ATT dial setting!"

The absolute "energy" level of sound signal for Be-SQ, Be-TW and ST is relatively much smaller than that for woofer, and the consumption of considerable portions of the energy by the "bypassed" ATT (just as a resistor) would still slightly affect the "amount" or "volume" of the signal going into the SP units (drivers), I assume.

Such recognition led me to actually measure the resistance values of the three ATTs. I once have measured these about 25 years ago, but I fully forgot about the measurements... I tentatively disconnected the three ATTs from the circuits and measured the resistance values as shown here;
WS000772.JPG


These measured values allowed me to draw the graphical representations;
WS000771.JPG


Since the YAMAHA Be-SQ and Be-TW are rather high efficient SP units (drivers), and we can agree with YAMAHA's original standard ATT labeling of NORMAL position at physical/absolute attenuation of about -4 dB. In my audio system with fully renovated LC-network in outer box, the best attenuations were found to be "NORMAL-4dB" to "NORMAL-5dB" for both Be-SQ and Be-TW which means -8dB to -9dB physical/absolute attenuation in the circuit. As for the extremely highly efficient horn-type super tweeter FOSTEX T925A, -10dB to -13dB attenuation was found to be the best fit in my reference audio system. These are the rationales for EKIO's gain settings, i.e. now -9dB for Be-SQ and Be-TW frequency range, and -13dB for TW range.

Well, one further question came to me; "What would be the best dial settings of the 'bypassed' but still recognized ATTs in the circuits??"

Last weekend, I actually did several intensive listening sessions placing my ears very near to the units (drivers) and measuring the actual values of the ATTs by my tester while cables were disconnected, then connected again, change the dial position slightly for listening, repeatedly so many times.

And, I finally found, at least in my environment and system, around 20 Ohm of the total resistance (impedance) (R1 + R2) of the "bypassed" ATTs would give the best sound quality and the best balance between the SP units (drivers).
WS000770.JPG


These settings are fully identical to the use of 20 Ohm fixed resistors, as shown here;
WS000769.JPG


In conclusion, I would like to try and test the feasibility of having 20 Ohm resistors "in parallel" with Be-SQ, Be-TW and ST units (drivers) in my coming next multi-amplifier trial session(s);
WS000768.JPG


Through these measurements of ATTs and insights on ATTs for NS-1000's highly responsive Be-SQ and Be-TW as well as for highly efficient FOSTEX ST T925A, I learned that I should very carefully configure the gain and impedance also in my multichannel multi-amplifier(s) system.

Edit: I had very interesting and invaluable discussion on validation and justification of this issue in #99(remote thread), #100(remote thread), #101(remote thread) on the remote thread entitled "ASR dummy load configuration".
 
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How can I (should I) test/compare One Candidate Stereo Amplifier in Multichannel System?

Hello friends,

Now I am almost ready to move forward again, by step-by-step, aiming towards final decision on multichannel amplifier(s) in this project.

As shared in my post #175 on YAMAHA MX-A5200, and post #228 on DENTEC DP-NC400-4-EXP, trials and tests with candidate 8-channel multichannel amplifiers would be just straitforward; I can just use and try it in my "always-ready-to-test" multichannel system.

On the other hand, I am still reserving possibilities of using four (4) of candidate stereo amplifier, or even eight (8) of monaural amplifier. In this trial situation, ideally I would like to use four of the specific stereo amplifier at once in my multichannel system, but my dear audio-enthu friends and/or audio shop usually (always) offer me to test only one candidate amplifier for my trials and listening sessions; I fully understand that they seldom (or never) have four (4) of the specific amplifier for my test at my home.

In all of the cases if I would have one such candidate amplifier (AMP-X), of course I will first try and test it in my single-DAC plus single-amp system with LC-network, as the replacement of my ACCUPHASE E-460, for comparative listening sessions.

I really would like, however, also to try AMP-X in my multichannel system together with E-460 for intensive tests and comparative listening sessions, in conditions and limitations of;

1. I should try AMP-X to dedicatory drive woofers (WOs) and dedicatory drive Beryllium squawkers (Be-SQ) while ACCUPHASE E-460 is taking care of other one of these.

2. In such multichannel trials, the Beryllium tweeters (Be-TWs) and FOSTEX super tweeters (STs) should be driven by the third stereo amplifier (and the fourth stereo amplifier); in these trials, the third stereo amplifier may cover both of Be-SQs and STs. (Software crossover EKIO can take care of gains for them.)

3. Such third amplifier to drive Be-SQs and TWs would be rather reasonably priced low power amplifier with only RCA unbalanced inputs and bi-wiring speaker connection capability. Even though, I definitely do not like to use any of XLR-to-RCA adaptors and XLR-to-RCA cables, as strongly suggested not use these in DAC8PRO's Owner's Manual and Pavel's recent comments.

4. As for the signal input into the YAMAHA YST-SW1000 sub-woofers (SWs), either of "RCA unbalanced input" or "speaker-level branching input from WO terminals" would be applicable and acceptable.

Very fortunately, as I shared here before, DAC8PRO's AES/EBU digital OUT is the through CH-1+CH-2 with sync clock information for the second DAC (in my case I have ONKYO DAC-1000) which has AES/EBU digital IN and high quality R and L RCA unbalanced analog OUT. I can use this nice feature to meet the requirement-2 and -3 above.

As for the reasonable third stereo amplifier, I am just planning to purchase a very reasonable YAMAHA integrated amplifier A-S301 for my dining room's small audio system, and I can use it also here in my project to also fulfill the requirement-2 and -3 above.
https://usa.yamaha.com/products/audio_visual/hifi_components/a-s301/index.html
https://usa.yamaha.com/products/audio_visual/hifi_components/a-s301/specs.html#product-tabs

For my coming trials with AMP-X and ACCUPHASE E-460, the "tentative use" and specs of YAMAHA A-S301 would be acceptable for me, since it only drives Be-TWs and STs.

Consequently, the schematic diagram for the possible multichannel trials with "one" candidate AMP-X is sown below;

WS000795.JPG


Currently, I have scheduled trial plans with two of AMP-X; Benchmark AHB2 (from an audio shop in Tokyo) and ROTEL RB-1582 MkII from one of my audio-enthu friends;
ROTEL RB-1582 MkII;
http://www.rotel.com/product/rb-1582-mkii

I am also exploring possibilities to have trial sessions with other three or four different AMP-X in the near future.

I am a little bit afraid of, however, that my possible future posts here sharing my subjective (and slightly objective) results and impressions in detail may be recognized by the moderators of ASR as somewhat like an advertisement or commercial statements for these AMP-X, even though I have no conflict of interest at all with these AMP-X manufacturers and distributors; what do you think about this concern?

In any way, I would highly appreciate hearing your suggestions and comments on all the above.
 
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Elimination of Magnetic Susceptible Metals in Handling of Speaker Level Signal

Hello Friends,

Although I assume it is well known in Hi-Fi audio scene, this post is intending as a reminder for myself and all of us.

In speaker level signal handling, such as LC-network, attenuators, SP selectors and protection capacitors, we should carefully avoid any magnetic susceptible metals in the circuits even with connection terminal blocks, binding posts, etc.

In my current project, I use DIY renovated LC-network in outer box for reference sound system;
WS000816.JPG


and I also prepared SP cabling board with several connection terminal blocks;
WS000817.JPG


I very much carefully selected made of non-magnetic metals for all of the Y-lugs, R-lugs, connection terminal blocks (plates and screws) and binding posts. We can easily check of magnetic or non magnetic by a tiny magnet or a magnetized iron screw driver.

If we would accidentally use magnetic metals, e.g. for a terminal block (metal plates and/or screws), it will considerably deteriorate the sound by giving blur, low-resolution and uncleanliness to the sound quality.

I purchased audio grade non-magnetic terminals blocks at a DIY audio pro shop. The plate and the screw are made of non-magnet metal. Although I sometimes feel a little bit frustration since I cannot catch the screw by magnetized screw driver, it is really important to avoid any magnetic susceptible metals in these circuit to keep excellent Hi-Fi level sound quality.
 
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Further Tests and Trials with Several Stereo Amplifiers

Note:
I am an end user audio enthusiast and I have no conflict of interest at all with any of the manufacturers, import companies, distributors and audio shops relevant to this post. This post is not intending to share objective and/or subjective evaluations of the candidate amplifiers, but I would like to share about how I would test and try candidate amplifiers in this project.


Hello friends,

Now I am waiting for arrivals of several stereo amplifiers to be tested and tried in single amplifier reference system and also in multichannel system.

As for the 20 - 25 Ohm resistors to be inserted in parallel with Beryllium Squawkers (Be-SQs), Be-Tweeters (Be-TWs) and Super Tweeters (FOSTEX T925A) (STs), I decided to use Jantzen Audio's 22.00 Ohm 10W Audio-Grade Non-Inductive Wirewound Superres Resistor. I purchased 6 (six) of this resistor, and attached lead wires and Y-lugs;
WS000812.JPG

Edit: I had very interesting and invaluable discussion on validation and justification of this issue in #99(remote thread), #100(remote thread), #101(remote thread) on the remote thread entitled "ASR dummy load configuration".


During the coming two to three months, I will try and test several stereo amplifiers as AMP-X and/or AMP-Y;

1. in my single amplifier - LC-network reference sound system in comparison with my ACCUPHASE E-460,
and,
2. in my current multichannel multi-amplifier system together with E-460.

For 1. in single amplifier - LC-network reference system, in this scheme;
WS000834.JPG


And, 2. in multichannel multi-amplifier system, in this scheme;
WS000835.JPG


You would please note that, in the above multichannel multi-amplifier system, this time I would like to have AMP-Y to drive both of Be-TW and ST, with only one 1.5 microF capacitor in ST line for low-cut around 8,800 Hz -6 dB/OCT slope. As AMP-Y will be working only with above 6,000 Hz for highly efficient Be-TW and horn-type ST, we can use a Hi-Fi grade amplifier with small driving power as AMP-Y.

Another merit of above scheme would be I can fully separate the WO sound region (45 - 600 Hz) from SW region (15 - 50 Hz) with utilizing all of the 8-channels of DAC8PRO's output in completely sync manner. I assume it should be better that WO would not receive overlapped SW signal of 15 - 50 Hz.

The amplifiers to be tested in these schemes during the coming two to three months are;

BENCHMARK AHB2
https://benchmarkmedia.com/products/benchmark-ahb2-power-amplifier
https://www.audiosciencereview.com/...-and-measurements-of-benchmark-ahb2-amp.7628/

ROTEL RB-1582 MkII
http://www.rotel.com/product/rb-1582-mkii
https://www.soundstageaccess.com/index.php/equipment-reviews/533

TEAC AP-505
https://hifiheaven.net/shop/TEAC-AP-505-Ultra-Compact-Stereo-Power-Amplifier-Black
https://www.amazon.com/dp/B07SS29B4J/ref=sr_1_1?dchild=1&keywords=TEAC+AP-505&qid=1597466517&sr=8-1

SOULNOTE A-0 (10W 8 Ohm, as AMP-Y to drive Be-TW and ST)
https://www.kcsr.co.jp/detail_a0.html
https://www.amazon.com/SOULNOTE-Integrated-Amplifier-Domestic-Products】【Ships/dp/B0837PHVHR/ref=sr_1_1?dchild=1&keywords=SOULNOTE&qid=1597466787&sr=8-1

Edited to add on September 24, 2021:
and TEAC-AP505, ACCUPHASE A-35, YAMAHA A-S301, YAMAHA A-S3000 and SONY TA-A1ES.
Before this post, I have tested ACCUPHASE E-460, YAMAHA MX-A5200, DENTEC DP-NC400-4-EXP.
You may jump to
my post #311 for my (provisional) decision on amplifier selection.
My system configuration as of August 7, 2021 can be found in
my post #416.

In my near future posts, I will briefly share about progress and preferences with these amplifiers.
 
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Benchmark AHB2 Tests and Trials: Part-1
Comparison with ACCUPHASE E-460 in Single-Amplifier + LC-Network Reference Sound System


Note: I am an end user audio enthusiast and I have no conflict of interest at all with any of the manufacturers, import companies, distributors and audio shops relevant to this post. This post is not intending to intensively share objective and/or subjective evaluations of the candidate amplifiers, but I would like to share about how I would test and try candidate amplifiers in this project.

Hello friends,

Very fortunately, Benchmark AHB2 power amplifier with one year domestic warranty is now available in Japan through a sole import company and many of the audio shops, and I am having an opportunity to try and test one unit of AHB2 came to my home the day before yesterday;
https://benchmarkmedia.com/collections/amp/products/benchmark-ahb2-power-amplifier

You can see the inside photos of AHB2 at Benckmark's "Application Notes" page;
https://benchmarkmedia.com/blogs/ap...radical-approach-to-audio-power-amplification

Let me also paste the link for AHB2's Japan site just for my domestic audio enthu friends;
https://www.benchmarkmedia.jp/products/ahb2/

In this Part-1 post, I will describe my tests and trials with AHB2 in my reference single-amp + LC network sound system in comparison with my ACCUPHASE E-460 integrated amplifier.

Before fully installing AHB2 in my system, I carefully listened again to all of my audio sampler tracks using the reference ACCUHASE E-460 + LC-network system, with software crossover EKIO and DAC8PRO, in this scheme;
WS000846.JPG


WS000847.JPG


Then, I carefully replaced ACCUPHASE E-460 with Benchmark AHB2 like in this scheme;
WS000849.JPG


The actual set-up photos are;
WS000844.JPG

and,
WS000845.JPG


Under the AHB2, you can see one YAMAHA A-S301(B) which is not in use in this post, so you would please ignore it.

Although I have slightly adjusted the L&R input gain and the gains for Mid, High and S-High sound regions in EKIO's I/O panel so that the master volume in JRiver would give about the same sound volume, all of the other parameters remained unchanged as those in E-460 system.

I have no need to say much about amazing performance and excellent sound of AHB2 which are very well measured and reviewed by amirm;
https://www.audiosciencereview.com/...-and-measurements-of-benchmark-ahb2-amp.7628/
and also by many people at many places.

I found that the total sound quality and impression of AHB2 fit perfectly nice to my ears and brain after intensively hearing my reference sound with ACCUPHAAE E-460 and AHB2. I am especially impressed by AHB2's amazing linearity of the total sound quality over very small volume of the sound to massive large loud volume. I agree with many people who call AHB2 as a revolutionary excellent dream amp.

Here, I should only say that Benchmark AHB2 is a perfect and wonderful amplifier also in my reference single amplifier + LC-network system with my speakers consists of renovated YAMAHA NS-1000, sub-woofer YAMAHA YST-SW1000 and super tweeter FOSTEX T925A.

I also confirmed that we would need almost no warming-up of AHB2 as the owners' manual indicates; it gives excellent and consistent sound from just after the power-on, and all the way after.

AHB2 is also a physically and environmentally "cool" amplifier; the room temperature of my listening room is around 25 degree C, and my healthy body temperature is around 36.6 degree C; after continuous operation in relatively loud playback of my audio sampler tracks for more than 5 hours, the temperature of the top cover of AHB2 is slightly (about 2 - 3 degree C) higher than my body temp, so it would be constantly around 38 - 40 degree C.

This coming weekend, I will further try and test AHB2 in my multichannel multi-amplifier project without using the LC-network, where AHB2 would directly and dedicatory drive the Beryllium squawkers or the woofers.
 
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Phase and delay are basically the same thing, for any given frequency one can be coverted to the other.

Meh.....

Let's say you have a impulse noise. It's played on a 2 way speaker.

If the bass driver is phase shifted by 180 degrees many would say it's delayed by half a cycle, but this is technically incorrect.

It could be either delayed by half a cycle- caused by (for instance) using an IIR filter, or it could be out of phase, but not delayed, caused by (for instance) reversing the polarity of one of the drivers.

Understanding this, it would be incorrect to describe a sound that has 720 Deg phase error instead, you would describe it as 2 cycle delay.

The difference between the definitions matter when you deal with linear and min phase FIR crossovers (for instance), or hysteresis of coils. Analogue time energy storage always needs to be described in terms of time not phase.

This practice also hides sins that are not otherwise evident. For example, AVRs typically have a problem in that they use FIR filters, but due to audio sinc needs and processing power limits in an AVR, cannot allow EQ to cause the significant time delay of the signal necessary in order to mathematically solve the effects of EQ using time perfect methods (linear phase with time correction), so instead it introduces pre ringing if FIR (minimum phase) or post ringing if IIR. As pre ringing is very audiable if using high Q filters, this limits the ability of the AVR to fix the signal.
Having tested this using filters made in RePhase, (I require very high amounts of EQ as I use horns, so decided to test some very steep filters for audiable effects) Pre ringing in my example is 100% audiable with a clear ghostly sound coming before all sounds. Interestingly, once heard, it seams much easier to detect at lower levels afterwards (the listener knows what to listen for- at lower levels, it sounds 'glassy', like most AVRs I've heard up to now!- obviously this is a subjective opinion, but there we go.

In short, not all phase change causes delay, and delay does not always mean phase change. You can have one, the other or both, that is why you use impulse response as an example to show the difference, as it's a time limited event, just as music is.
 
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Meh.....
..........
If the bass driver is phase shifted by 180 degrees many would say it's delayed by half a cycle, but this is technically incorrect.
..........
In short, not all phase change causes delay, and delay does not always mean phase change. You can have one, the other or both, that is why you use impulse response as an example to show the difference, as it's a time limited event, just as music is.

Hello Lbstyling,

Thank you so much for your nice comments and info on phase and delay issue. As I wrote in my very first post, delay and phase issues are always one (two) of our main concerns in multichannel project, and we already had a lot of discussion and info exchange in this thread, and your above post added an important insight for my better understandings.

I have been also feeling, just like you are, a little bit frustration and difficulty-in-understanding on QMouse's comment of "Phase and delay are basically the same thing, for any given frequency one can be coverted to the other. " I do hope you and QMouse would kindly start a new thread on these issues for our further discussion and better understandings, and I believe such new thread shall greatly contribute for our practical implementations of best or better handling of phase and delay in multichannel system.

Since I am not an expert on these phase and delay issues like you and QMouse, my approach in this project has been rather empirical and experimental one based on my fundamental knowledge learnt mainly from;

https://en.wikipedia.org/wiki/Audio_crossover
and
http://linea-research.co.uk/wp-content/uploads/LR Download Assets/Tech Docs/CrossoverFilters White Paper -C.pdf
and
http://www.acourate.com/freedownload/XOWhitePaper.pdf

as I summarized in my post #132 the list of public domain documents and books for basic understandings on audio crossover.

I have been firmly sticking to the nice software crossover EKIO throughout in this project, and as Guillaume of LUPISOFT kindly informed, "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." Very fortunately, I found that mild slope -12 dB/Oct LR filters are the best fit for my speakers, i.e. renovated YAMAHA NS-1000's woofer (WO), Be-squawker (Be-SQ), Be-tweeter (Be-TW), super tweeter FOSTEX T925A (ST) and sub-woofer YANAHA YST-SW1000 (SW); I do not use large horns for low and mid ranges.

Now, I pleasantly feel that your post above is somewhat validating my empirical and experimental approaches in EKIO crossover configurations with all -12 dB/Oct LR filters and the "invert" is checked for SW and Be-SQ sound regions. (Be-TW and ST are "inverted" at the SP unit terminals). Since WO, Be-SQ and Be-TW are rigidly mounted in NS-1000's nice cabinet with YAMAHA's historical design and the positions of SW and ST can be physically adjusted by moving them back-and-forth, I am currently applying no delay at all in EKIO's configurations at least until I would fully decide the multichannel amplifiers in this project.
WS000847.JPG


WS000848.JPG


For my ears and brain, this approach is perfectly in conformity with my reference sound system of single amplifier plus the renovated LC-network.

Very fortunately, I could also confirmed that I hear no post nor pre ringing at all with -12 dB/Oct LR IIR filters in EKIO configurations with my rather sensitive speaker environment.

Maybe, I will get back to the fine tuning of phase and delay, by EKIO configurations or more preferably by physical movement of SW and ST, after I would fully decide and install all the amplifiers; until then, your and QMouse's further discussions and suggestions will be highly appreciated.
 
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Hello Lbstyling,

Thank you so much for your nice comments and info on phase and delay issue. As I wrote in my very first post, delay and phase issues are always one (two) of our main concerns in multichannel project, and we already had a lot of discussion and info exchange in this thread, and your above post added an important insight for my better understandings.

I have been also feeling, just like you are, a little bit frustration and difficulty-in-understanding on QMouse's comment of "Phase and delay are basically the same thing, for any given frequency one can be coverted to the other. " I do hope you and QMouse would kindly start a new thread on these issues for our further discussion and better understandings, and I believe such new thread shall greatly contribute for our practical implementations of best or better handling of phase and delay in multichannel system.

Since I am not an expert on these phase and delay issues like you and QMouse, my approach in this project has been rather empirical and experimental one based on my fundamental knowledge learnt mainly from;

https://en.wikipedia.org/wiki/Audio_crossover
and
http://linea-research.co.uk/wp-content/uploads/LR Download Assets/Tech Docs/CrossoverFilters White Paper -C.pdf
and
http://www.acourate.com/freedownload/XOWhitePaper.pdf

as I summarized in my post #132 the list of public domain documents and books for basic understandings on audio crossover.

I have been firmly sticking to the nice software crossover EKIO throughout in this project, and as Guillaume of LUPISOFT kindly informed, "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." Very fortunately, I found that mild slope -12 dB/Oct LR filters are the best fit for my speakers, i.e. renovated YAMAHA NS-1000's woofer (WO), Be-squawker (Be-SQ), Be-tweeter (Be-TW), super tweeter FOSTEX T925A (ST) and sub-woofer YANAHA YST-SW1000 (SW); I do not use large horns for low and mid ranges.

Now, I pleasantly feel that your post above is somewhat validating my empirical and experimental approaches in EKIO crossover configurations with all -12 dB/Oct LR filters and the "invert" is checked for SW and Be-SQ sound regions. (Be-TW and ST are "inverted" at the SP unit terminals). Since WO, Be-SQ and Be-TW are rigidly mounted in NS-1000's nice cabinet with YAMAHA's historical design and the positions of SW and ST can be physically adjusted by moving them back-and-forth, I am currently applying no delay at all in EKIO's configurations at least until I would fully decide the multichannel amplifiers in this project.
View attachment 79204

View attachment 79208

For my ears and brain, this approach is perfectly in conformity with my reference sound system of single amplifier plus the renovated LC-network.

Very fortunately, I could also confirmed that I hear no post nor pre ringing at all with -12 dB/Oct LR IIR filters in EKIO configurations with my rather sensitive speaker environment.

Maybe, I will get back to the fine tuning of phase and delay, by EKIO configurations or more preferably by physical movement of SW and ST, after I would fully decide and install all the amplifiers; until then, your and QMouse's further discussions and suggestions will be highly appreciated.

I cannot see how you can have evaluated the amount of pre-ringing or post ringing you can hear in your system as you will have never heard the (post ringing in your case) removed. Almost nobody has heard it removed under any situation.

I would suggest to evaluate it, you would need to add it until it's audiable, so you know what it sounds like, ( create a high Q filter in REW, and add it to JRiver filters) then remove it and see. I personally think it's only audiable in the mid range, and only on exceptionally low distortion drivers, but that's a subjective opinion.
 
I cannot see how you can have evaluated the amount of pre-ringing or post ringing you can hear in your system as you will have never heard the (post ringing in your case) removed. Almost nobody has heard it removed under any situation.

I would suggest to evaluate it, you would need to add it until it's audiable, so you know what it sounds like, ( create a high Q filter in REW, and add it to JRiver filters) then remove it and see. I personally think it's only audiable in the mid range, and only on exceptionally low distortion drivers, but that's a subjective opinion.

Hello again, Lbstyling,

As wrote in my post #143, 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, using 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.

Very fortunately, even with extremely low distortion YAMAHA Beryllium mid-range driver, as well as the low distortion Beryllium tweeter, no post ringing at all with EKIO's IIR -12 dB/Oct filters, and that should be quite OK and enough in my project.

Even I assume no further experiment with " intentionally added audible post ringing" would be needed in my project, I am personally interested in your kind suggestion on adding audible pre and/or post ringing in the sound in my system. I would like to reserve such experiments for my near future home work!
 
Benchmark AHB2 Tests and Evaluations: Part-2
Incorporation in Multichannel Multi-Amplifier System and Listening Sessions


Note: I am an end user audio enthusiast and I have no conflict of interest at all with any of the manufacturers, import companies, distributors and audio shops relevant to this post. This post is not intending to intensively share objective and/or subjective evaluations of the candidate amplifiers, but I would like to share about how I would test and try candidate amplifiers in this project.

Hello friends,

I assume all of you may agree with me that, in my current project, the extremely low distortion (almost zero distortion) Benchmark AHB2 should directly and dedicatory drive the extremely low distortion YAMAHA Beryllium mid-range squawker driver (Be-SQ). Yes, yesterday and today, for the first time, I incorporated AHB2 in my multichannel multi-amplifier system for intensive listening sessions.

As now I have one AHB2 for my tests and trials, and I would like to drive woofers (WOs) by my ACCUPHASE E-460, this time I tentatively use YAMAHA A-S301(B) integrated amplifier to drive Be-tweeter (Be-TW) and super tweeter (ST) FOSTEX T925A, as I carefully considered in my post #251.

Since A-S301(B) has only RCA unbalanced input and I do not like to use any of the XLR-to-RCA adaptors or cables, I applied the DAC8PRO's AES/EBU (CH1+CH2) digital out into my ONKYO DAC-1000(S) and the RCA unbalanced out of DAC-1000(S) into A-S301(B) to drive Be-SQ and ST. Only one low-cut 1.5 microF capacitor (AUDYN Cap 1.5 uF 630V) is in the SP line for ST for -6 dB/Oct at around 8,800 Hz. I use A-S301(B) in "PURE DIRECT" mode bypassing all of the tone control, balance control and loudness compensator.

Furthermore, just only for current trials with AHB2, I dared to use OPPO SONICA DAC with its ASIO driver to give RCA unbalanced input to sub-woofer (SW) YAMAHA YST-SW1000. The main reason for this is that I do not like to have overlapped low signal of 15 - 55 Hz with the signal (45 - 600 Hz) for WO. I know well that, in this case, DAC8PRO and SONICA DAC is not in full sync and there would be possibility of drift into out of sync between the two DACs for the long music tracks. In my current trials, however, I rather frequently reset DAC-1000(S) to prevent the possible "out-of-sync", and it worked just fine.

Consequently, the total scheme of the tested system is like this;
WS000860.JPG


You may look at the SP cabling boards;
WS000861.JPG


On each board, you can see three blue protection capacitors (68, 10, 10 uF Jantzen Standard Cap 400V), one small black low-cut capacitor AUDYN-CAP 1.5 uF 630V, and three Jantzen Audio 22.00 Ohm 10W SuperRes resistors.

The software crossover EKIO's configurations remained unchanged;
WS000862.JPG


and,
WS000863.JPG


I did intensive listening sessions for two days with this setup, and I found and confirmed that the combination of E-460 for WO and AHB2 for Be-SQ is really amazingly wonderful giving extremely high quality sound all the way.

I also found that the use of an integrated amplifier YAMAHA A-S301(B) for Be-TW and ST was well fit for the sound of WO and Be-SQ, and it would be also convenient to control the Be-TW and ST volume level by the volume of A-S301(B) using the remote controller. This finding is now well encouraging me to further evaluate rather small power but high quality integrated amplifiers with XLR balanced input, like SOULNOTE A-0, in this project for Be-TW and ST.

Before finishing my two-day listening sessions, I carefully bypassed the three protection capacitors for Be-SQ, Be-TW and ST to check the possible audible effect by them;
WS000864.JPG


Fortunately, I found little, almost no, inferior effect of these protection capacitors (Jantzen Audio Standard Caps). As I really would like to protect my treasure Be-SQ and Be-TW, I decided to use these protection capacitors at least all the way throughout my coming tests with several other amplifiers.

Let me just share with you a little bit of my subjective evaluation and impression. At the end of today's listening session with AHB2 in above scheme, I seriously listened to this nice album which I always use as a reference of full orchestra sound with soprano solo and chorus, Schubert "Rosamunde (Complete)", Kurt Masur (conductor), Gewandhausorchester Leipzig, with Elly Ameling (sopraso) and Rundfunkechor Leipzig (Leipzig Radio Chorus), Philips ASIN: B00000E2SS;
WS000867.JPG


This album was recorded in December 1983 with really amazing recording quality. The first track "Overture" is pp to fff full orchestra sound with very nice 3D perspectives, and with E-460 for WO and AHB2 for Be-SQ, together with SW, Be-TW and ST, the sound is really amazing; I feel as if I am sitting on the best seat in the Neues Gevanndhouse Leipzig. Among the very comfortable silky but vivid and gentle orchestra sound, now I can identify each individual violinist in the full orchestra. Really excellent sound quality by the audio system with AHB2, and again amazing recording quality.

Elly Ameling sang only in track-5 "Romance" for just 3 min 47 sec and her posture and beloved voice were best seen and heard ever before, even though I have listened to this album more than hundred times. My tears really down.

The famous ppp orchestra piece of track-7 "Ent'acte to Scene 3" is always a great challenge to audio system for orchestra string sound in good S/N without distortion; I cannot find suitable words how beautifully AHB2 and my system played this track also...

Track-9 "Chorus of Shepherds" is another challenge to audio system for the balance of 4-part chorus and orchestra sound in the nice acoustic hall. In the middle of the track, each of the solo singer from soprano, alto, tenor and bass sing in the center-back of the stage, and we need excellent 3D perspectives and sound resolution for very much impressive listening experience which were perfectly achieved by the audio system with AHB2.

I will continue testing AHB2 for coming weeks. One of my audio enthu friends will bring his ROTEL RB-1582 MkII to my home in the middle of September for similar evaluation.

Hopefully, I would like to summarize my exploration for amplifiers in late October.
 
Last edited:
Benchmark AHB2 Tests and Evaluations: Part-2
Incorporation in Multichannel Multi-Amplifier System and Listening Sessions


Note: I am an end user audio enthusiast and I have no conflict of interest at all with any of the manufacturers, import companies, distributors and audio shops relevant to this post. This post is not intending to intensively share objective and/or subjective evaluations of the candidate amplifiers, but I would like to share about how I would test and try candidate amplifiers in this project.

Hello friends,

I assume all of you may agree with me that, in my current project, the extremely low distortion (almost zero distortion) Benchmark AHB2 should directly and dedicatory drive the extremely low distortion YAMAHA Beryllium mid-range squawker driver (Be-SQ). Yes, yesterday and today, for the first time, I incorporated AHB2 in my multichannel multi-amplifier system for intensive listening sessions.

As now I have one AHB2 for my tests and trials, and I would like to drive woofers (WOs) by my ACCUPHASE E-460, this time I tentatively use YAMAHA A-S301(B) integrated amplifier to drive Be-tweeter (Be-TW) and super tweeter (ST) FOSTEX T925A, as I carefully considered in my post #251.

Since A-S301(B) has only RCA unbalanced input and I do not like to use any of the XLR-to-RCA adaptors or cables, I applied the DAC8PRO's AES/EBU (CH1+CH2) digital out into my ONKYO DAC-1000(S) and the RCA unbalanced out of DAC-1000(S) into A-S301(B) to drive Be-SQ and ST. Only one low-cut 1.5 microF capacitor (AUDYN Cap 1.5 uF 630V) is in the SP line for ST for -6 dB/Oct at around 8,800 Hz.

Furthermore, just only for current trials with AHB2, I dared to use OPPO SONICA DAC with its ASIO driver to give RCA unbalanced input to sub-woofer (SW) YAMAHA YST-SW1000. The main reason for this is that I do not like to have overlapped low signal of 15 - 55 Hz with the signal (45 - 600 Hz) for WO. I know well that, in this case, DAC8PRO and SONICA DAC is not in full sync and there would be possibility of drift into out of sync between the two DACs for the long music tracks. In my current trials, however, I rather frequently reset DAC-1000(S) to prevent the possible "out-of-sync", and it worked just fine.

Consequently, the total scheme of the tested system is like this;
View attachment 79469

You may look at the SP cabling boards;
View attachment 79470

On each board, you can see three blue protection capacitors (68, 10, 10 uF Jantzen Standard Cap 400V), one small black low-cut capacitor AUDYN-CAP 1.5 uF 630V, and three Jantzen Audio 22.00 Ohm 10W SuperRes resistors.

The software crossover EKIO's configurations remained unchanged;
View attachment 79471

and,
View attachment 79472

I did intensive listening sessions for two days with this setup, and I found and confirmed that the combination of E-460 for WO and AHB2 for Be-SQ is really amazingly wonderful giving extremely high quality sound all the way.

I also found that the use of an integrated amplifier YAMAHA A-S301(B) for Be-TW and ST was well fit for the sound of WO and Be-SQ, and it would be also convenient to control the Be-TW and ST volume level by the volume of A-S301(B) using the remote controller. This finding is now well encouraging me to further evaluate rather small power but high quality integrated amplifiers with XLR balanced input, like SOULNOTE A-0, in this project for Be-TW and ST.

Before finishing my two-day listening sessions, I carefully bypassed the three protection capacitors for Be-SQ, Be-TW and ST to check the possible audible effect by them;
View attachment 79473

Fortunately, I found little, almost no, inferior effect of these protection capacitors (Jantzen Audio Standard Caps). As I really would like to protect my treasure Be-SQ and Be-TW, I decided to use these protection capacitors at least all the way throughout my coming tests with several other amplifiers.

Let me just share with you a little bit of my subjective evaluation and impression. At the end of today's listening session with AHB2 in above scheme, I seriously listened to this nice album which I always use as a reference of full orchestra sound with soprano solo and chorus, Schubert "Rosamunde (Complete)", Kurt Mazur (conductor), Gewandhausorchester Leipzig, with Elly Ameling (sopraso) and Rundfunkechor Leipzig (Leipzig Radio Choras), Philips ASIN: B00000E2SS;
View attachment 79474


This album was recorded in December 1983 with really amazing recording quality. The first track "Overture" is pp to fff full orchestra sound with very nice 3D perspectives, and with E-460 for WO and AHB2 for Be-SQ, together with SW and ST, the sound is really amazing; I feel as if I am sitting on the best seat in the Neues Gevanndhouse Leipzig. Among the very comfortable silky but vivid and gentle orchestra sound, now I can identify each individual violinist in the full orchestra. Really excellent sound quality by the audio system with AHB2, and again amazing recording quality.

Elly Ameling sang only in track-5 "Romance" for just 3 min 47 sec and her posture and beloved voice were best seen and heard ever before, even though I have listened to this album more than hundred times. My tears really down.

The famous ppp orchestra piece of track-7 "Ent'acte to Scene 3" is always a great challenge to audio system for orchestra string sound in good S/N without distortion; I cannot find suitable words how beautifully ABH2 and my system played this track also...

Track-9 "Chorus of Shepherds" is another challenge to audio system for the balance of 4-part chorus and orchestra sound in the nice acoustic hall. In the middle of the track, each of the solo singer from soprano, alto, tenor and bass sing in the center-back of the stage, and we need excellent 3D perspectives and sound resolution for very much impressive listening experience which were perfectly achieved by the audio system with AHB2.

I will continue testing AHB2 for coming weeks. One of my audio enthu friends will bring his ROTEL RB-1852 MkII to my home in the middle of September for similar evaluation.

Hopefully, I would like to summarize my exploration for amplifiers in late October.
That's a true enthusiast's system I really like it. :D
 
Last edited:
Hopefully, my next two or three posts will be due on September 19 or 20 sharing the tests and evaluations on ROTEL RB-1582 MkII together with Benchmark AHB2 in my single amplifier system and also in multichannel multi-amplifier system.
 
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