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

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dualazmak

dualazmak

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Nice cabling work. But why "protecting" capacitors? There is no need for but may influence the sound.

Protecting capacitors are there, to prevent an amplifier output device failure, from blowing the unobtanium (Be)midrange and tweeters. ( by applying a power supply rail voltage at high current availability) When properly selected which @dualazmak has done they are unlikely to be audible. They are essentially electronically transparent.

The argument to prevent bass tones from the midrange and tweeter is questionable since the crossover should do this. What about the woofer? Is there no protection capacitor? Prevention from high DC value can be done with so called crowbar circuit which shortens in case of DC. I agree that a capacitor is easier to implement. And using relais like in many amplifiers might not be perfect over time due to contact degradation.

Hello SSS and gene_stl:

Thank you for your interests and comments which I understood well.

At least in my setup, of course I carefully measured and confirmed that the protection capacitors are essentially transparent and "inaudible"; please refer to my posts here #402 and #485 (as well as #258 for subjective comparison).

I use these protection capacitors for protecting my treasure Be-midrange-squawkers, Be-tweeters, and metal horn super-tweeters from possible accidental intrusion of low Fq signal and/or DC.

In the course of intensive DIY of DSP-based multichannel multi-SP-driver multi-amplifier "fully active" audio setup, we have several possibilities of such unexpected "accidents", e.g. mis-configuration/mis-typing in XO/EQ/Group-Delay/Gain settings, mis-connection of line-level and/or SP-high-level cables, unexpected pop due to "excessively buffered" signal intrusion when changed the XLR cables, etc., etc.

I actually experienced a few such cases (even I have been always much careful though) that the protection capacitors actually did their job perfectly protecting my treasure SP drivers. Consequently and fortunately, I never lost/damaged my treasure SP drivers thanks to the protection capacitors.

I also understand @SSS's concerns on "relays" within amplifiers which are relating to QC, durability and warranty of amplifiers; I mean which and what relays the manufacturer would select and use (in some cases they use oxygen-free pressurized nitrogen or SF6 filled fully shielded/covered relays), as well as maintenance service availabilities even after the warranty period for long years. This issue wound be one of the critical factors for our amplifier selection. At least in my case, I very much carefully selected my four amplifiers in this respect too. You would please refer to my summary post here. Fortunately, my amplifiers are still in excellent perfect healthy conditions. My post here would be also your interest and reference, I assume.

Accuphase, Yamaha and Sony are still providing nice maintenance/repair services for any of their present and past products with reasonable cost, and we also have several domestic third-party maintenance/repair firms for these amplifiers.

BTW, just for your convenience and further overview, you can find here (on this thread) and here (remote independent thread post) the Hyperlink Index for this project thread.
 
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dualazmak

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dualazmak

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Non-magnetic is OK but also not necessary to my experience and opinion.

You would please refer to my posts:
- Elimination of magnetic susceptible metals in SP signal handling: #250, #013(remote thread), #023(remote thread)
Ando also here #9(remote thread); I wrote there:
If you take a look inside some rather high-end HiFi amplifiers, you'll see that the SP output wiring (and power wiring?) uses non-magnetic terminals and screws made of brass (no iron at all) or pure copper. However, this is also a common-sense measure to prevent sound quality deterioration in HiFi amplifiers. I remember it being pointed out and explained in interviews with a Yamaha amplifier designer and a Rotel engineer.

It is frustrating when working with magnetized screwdrivers (screwdrivers) because you can't catch the screws, though.

Yamaha's and Rotel's amplifier designers had a hard time persuading the assembly workers at the amplifier factories, but in the end they convinced them to use non-magnetic terminals and screws, giving priority to sound quality; I've also heard that the screwdriver, which uses a chuck to fix screws and bolts to the tip, was devised so that it could be used in factories. In my DIY audio setup, I have the same thing; I strictly/completely eliminate/avoid any magnetizable metal/screw in my SP cabling/connecting.

You would please also be reminded that the audible/measurable distortion was caused by iron (steel) plates at the SP binding posts of old-version of BUCKEYE 3 Channel Purifi Amp, and the cause (=steel plate on SP binding posts) were found/identified, then BUCKEYE replaced the parts with blass plates by a kind of recall announcement; please refer to the specific thread.

EDIT:
My (our) presently (as of today January 12) ongoing discussions here and thereafter on that remote thread would be also of your interest and reference.
 
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dualazmak

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Further I would crimp and solder the spades.

Regarding your specific point (solder or not on crimped spades/terminals), I once contacted with three of the companies producing tin-electroplating oxygen-free pure copper terminals. All of them answered that "You should never solder the crimped terminals!!" since the soldered portion may crack afterword if you would slightly bend the terminal (which would occur rather frequently) for screwing on the terminal-posts/binding-posts. In the worst case, the cracked solder chip may drop onto other electric/electronic parts and may cause unrecoverable short-circuit damages and even fire accidents... (It looks this is a common sense in electric/electronic engineering world, and is a fundamental educational tip in the industry).

The market-top company continued saying... If the crimping is done properly and tightly using exactly-size-matched pro-use robust crimper, the connection is made by crushing the wire and the contact at the same time. When crushing the contact, the wire is also crushed at the same time, so the wire extends. In this case, the cross-sectional area decreases after the plastic zone (the zone of permanent deformation). At this time, plastic deformation occurs, so the tensile strength becomes stronger. (It becomes stronger because the upper limit of the elastic range increases.) The residual stress at this time makes the wire and contact strongly connected. In this process, the oxide film on the surface of the wire and the oxide film on the surface of the contact come into contact with each other after being peeled off by friction, so the contact resistance becomes very low.
 
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Yes!
The company I worked for researched the crimp vs solder thing too - the conclusion was that crimping (properly done, per above) was superior, and lasts longer especially in 'touring' conditions, bc it is air tight (ie, reduces oxidation). Brittleness at the end of the solder joint (in wiring harnesses) caused wire fraying and breakage. It's 'all about the tools'.
 
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dualazmak

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Renewal of SP cabling boards beside SP systems

Hello dear ASR friends,

Abbreviations in this post;
- L&R sub-woofers (SWs),
- L&R woofers (WOs),
- L&R Beryllium-midrange-squawkers (MDs),
- L&R Beryllium-tweeters (TWs),
- L&R metal-horn-super-tweeters (STs),

Each of these is driven directly (with no LCR-network nor attenuator) by dedicated HiFi amplifier; SW has powerful built-in amplifier, and I use four stereo HiFi integrated amplifiers for others.

This post is a follow-up of my own post #895 entitled “Semi-annual intensive cleaning of all the metal-to-metal connectors/contacts, and complete renewal of all the tin-electroplated copper terminals with heat-shrink insulators”.

During the past almost four years, I have been objectively testing tuning measuring evaluating, and intensively subjectively evaluating the following items mounted on my L&R SP cabling boards:

- Parallel audio-grade resistors (now 22 Ohm) in SP high-level lines giving only a slight extra workload (above zero-cross level) to each of the amplifiers driving MDs, TWs and STs. (ref. #248, #251, #99remote thread, #100remote thread, #101remote thread)

- Protection capacitors for MD (68 uF), TW (10 uF) and ST (10uF) (ref. #4, #184, #402, #485, #890)

- High-path (low-cut) capacitors for ST (now parallel 1.5 uF x 2 =3.0 uF) (ref. #402, #485)

- Two of 8 Ohm 100W resistor as dummy SP for silent burn-in of amplifiers and measurement of amplifier’s SP output in silence (ref. #401, #402)

You would please note that all of these tuning and protecting resistors and capacitors in analog SP high-level lines are being "added” to the DSP EKIO’s XO/EQ/Group-Delay controls in upstream digital domain.
WS00006956.JPG


And the enlarged “SP Cabling Board” portion of the signal path;
WS00006901.JPG

(For other details of my latest system setup, you would please refer to my posts #774 and #858.)

Now that I believe I could have successfully accomplished my objective and subjective evaluation of these items on my SP cabling boards, I fully moved them (during last week) onto slightly larger new SP cabling boards for simpler connections (less connecting points), easier maintenance, pair-twisting cables as far as possible, as well as future possible implementation of other tuning items or other amplifiers.

The renewed L&R SP cabling boards beside the SP system, all the cables are AWG12 unless otherwise indicated:
WS00006904.JPG


This angle-shot photo tells you how the capacitors are tightly mounted on the board with hard-sponge cushions.
WS00006898.JPG


Now, the backside wiring/cabling view of right channel SP system is like this;
WS00006902.JPG


I would be more than happy if this post would be somewhat of your interest and reference.:)
 

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dualazmak

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Apparent power consumption of whole audio system during daily audio listening sessions: how SDGs-friendly is it?

Top notes:
Yesterday I have started a new thread entitled "Apparent power consumption of whole audio system during daily audio listening sessions: how SDGs-friendly is it? (not the idle power, please.)" on this topic with almost the same contents as I share in this post, and I hope and believe it would be allowed also having this post here on my project thread.
In case if you would be interested in sharing "your case" on this topic, you would please do so on the above new thread; your participation will be highly welcome!



Hello dear ASR friends,

Now that I believe I have (almost) completed the setup of my PC-DSP-based multichannel multi-SP-driver multi-amplifier fully active audio system (ref. here #774), yesterday I measured apparent power consumption of the whole audio system during my routine/daily music listening session while I was playing full orchestral fff Tutti quite loudly (almost maximum volume/gain in my listening room environments); the power consumption included one audio dedicated PC and its LCD monitor. The 55-inch OLED TV was not connected/powered-on.
WS00006957.JPG

The measurement was done after about 20 min warming-up of the whole system.

In Japan, our AC electricity is 100 V, and the total AC current for the system was around 2.72 A as shown above, so the "apparent power consumption" is around 272 W which is fortunately well below my expectation (or even my and my wife's a little bit of fear) for the multichannel multi-amplifier setup.

Just for your interest and reference, details of my latest audio setup can be found here #774 on my project thread, and the total physical setup is shown in this diagram; in daily audio music listening session, I do not connect/power-on my 55-inch OLED TV Panasonic TH-55Z1800.
WS00006940.JPG


You would please find the daily standard start-up/ignition sequences for my audio listening session here #776.

As you can easily guess, the main power eaters are four integrated amplifiers Class-AB Accuphase E-460, Class-AB Yamaha A-S3000, Class-AB Yamaha A-S301, Quasi-Class-A Sony TA-A1ES, and the L&R active subwoofers Yamaha YST-SW1000. I have been speculating that these power eaters consume more than 600 W at maximum gain/volume load, but the actual power consumption by them is less than 240 W at peak. (This was confirmed by turning-off all of these five and showing around 40 W residual power consumption by other audio gears including the PC and LCD monitor, DAC8PRO, 12-VU-Meter Array, etc.)

In case if I also use my 55-inch Panasonic 4K OLED TV TH-55HZ1800 for YouTube and other audio-visual sessions (ref. here and here), it will consume additional about 360 W, but I seldom do it in my "mainly-audio-only setup", i.e. less than once for 2 hours in three months.

In any way, we (my wife and myself) are now relieved knowing that our daily standard music listening session using my multichannel audio rig consumes rather SDGs-friendly only around 270 W which is well less than our prior thoughts/fears.

How about in your audio(-visual) setup during your ordinary/daily music (video) listening sessions? Is it SDGs-friendly based on your personal standard??

Your participation sharing "your case" on the newly started thread will be highly appreciated and much welcome!
 
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dualazmak

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Yokogawa makes nice instrumentation.

Yes, recently I also shared Yokogawa CL220 here showing its User's Manual too on the thread entitled "What is on your workbench right now?":D

 
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dualazmak

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dualazmak

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Again, summary of my rationales and pros of analog-level relative gain (tonality) controls in addition to gain controls in DSP configuration:

I recently shared this my post on another thread:
Thank you for your invaluable response and comments to me which should be also nice reference and of interest for many people following this wonderful thread!:)
Keith_W said:
Of course it is possible to tweak the volume of each driver on the fly by simply twiddling the knobs on each volume trim, but why would I want to do that? I carefully set up the system by measuring the volume of each driver and adjusted the trim. Then I taped over the volume controls and I don't touch them under any circumstances.

Again, I well understand and highly respect your above policy and approach of "fix and tape" the relative gain knobs in analog amplifier level.

As I repeatedly shared, however, at least in my case, I still would like to have my more-or-less freedom in relative gain (tonality) control on-the-fly at analog level, even though I usually "fix" (but I do not tape;)) each of the gain knobs of my four integrated amplifiers at my standard/default positions optimized for my daily classical music listening. Of course, I too have some relative gain adjustments also in my DSP configuration.
WS00006960.JPG
I occasionally further fine tune, however, the relative gains (tonality) (within about +/- 5 dB range) in analog amplifier level depending on genre of music, recording quality, age-dependent hearing decline of specific audience(s), etc.

Just for example, when I enjoy my beloved smooth jazz trio music albums of Karel Boehlee Trio (rather soft-side recordings, ref. here), I boost-up my midrange (driven by ACCUPHASE E-460) about 3.4 dB for best fit to my jazz listening preference. This is just my personal preference and "freedom", even though I well understand that some (or many?) people here in ASR Forum would criticize (or blame) me by saying "such a tonality tuning depending on genre (or on recording quality) of music tracks would be blasphemy or heresy in HiFi audio listening!" , but I do not care it:D, please let me enjoy music based on my preferences and my tuning style!
You would please find a typical case here:
- A serious jazz fanatic friend came to my home for audio sessions using my multichannel multi-driver multi-way multi-amplifier stereo system: #438

The flexible on-the-fly tonality fine tuning (especially in high-Fq zone covered by tweeters and super-tweeters) compensating possible age-dependent slight hearing decline would be a little more serious issue when I would invite "various" music-lover guests to our audio listening session at my listening room.
Even for myself and my wife, since we know the diagnostically-proven slight hearing decline above 7 kHz, I (we) prefer a slightly upward SPL for 7 kHz to 20 kHz.
Again, this is just my personal preference and "freedom", even though I well understand that some (or many?) people here in ASR Forum would criticize (or blame) me by saying "such a tonality tuning depending on hearing capabilities would be blasphemy or heresy in HiFi audio listening!" , but I do not care it; please let me enjoy music based on my preferences and my tuning style!;)
- Excellent Recording Quality Music Albums/Tracks for Subjective (and Possibly Objective) Test/Check/Tuning of Multichannel Multi-Driver Multi-Way Multi-Amplifier Time-Aligned Active Stereo Audio System and Room Acoustics; at least a Portion and/or One Track being Analyzed by Color Spectrum of Adobe Audition in Common Parameters: [Part-11] Violin Music: #643

And, you would please let me describe this paragraph again here:
I know that XOs (many parameters), time-alignment, phase tuning, EQ and gain in DSP would be more-or-less interdependent with each other, right? None of them can be changed/modified completely independently. On the other hand, analog-level gain controls do not affect the upstream DSP configurations.

I also shared this my post on another thread regarding analog-level relative gain control as one of the safety measures:
Given you can do all of this precisely in software, this looks like a preference for physical knobs and dials. If it's not that, what practical advantage do you think this would provide?

I know that XOs (many parameters), time-alignment, phase tuning, EQ and gain in DSP would be more-or-less interdependent with each other, right?
None of them can be changed/modified completely independently.
On the other hand, analog-level gain controls do not affect the upstream DSP configurations.

One of the other pros of HiFi analog-level relative gain tuning (tonality control) by knobs/dials would be it is very safe on-the-fly compared to on-the-fly DSP gain control.

In case if you would like to do it in DSP, especially on-the-fly, you always have possibility of mis-adjustment of gains, i.e. and e.g. 20 dB boost instead of intended 2.0 dB boost by numerical keyboard mistyping the value (and/or it would happen even using mouse wheel up-and-down in some DSP software tools) which may harm and/or destroy your precious SP drivers.

Consequently, I seldom change DSP parameters (especially relative gains) on-the-fly, while listening to music track;); I always stop playing and set the master volume in digital music player to minimum (minus infinity dB) position when changing the DSP parameter(s), and after the DSP modification and after start playing music, I very carefully and slowly gain-up the master volume (in my case JRiver MC) to check the given change in DSP parameters; in this way, I can avoid possible (but rare) harm/damage to SP drivers.

For further safety purposes, I also use protection capacitors for my treasure Be-midranges, Be-tweeters and metal-horn super-tweeters (summary ref. here).

I use DSP EKIO, and EKIO has very nice gain up-and-down by mouse wheel rotation (0.1 dB granularity); I have been always strongly recommending, therefore, to use mouse wheel rotation (not keyboard numeric typing), if you use EKIO and would like to do on-the-fly relative gain control (e.g. ref. here).
 
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dualazmak

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A new series of audio experiments on reflective wide-3D dispersion of super-tweeter sound using random-surface hard-heavy material: Part-1_ Background, experimental settings, initial preliminary listening tests

Abbreviations in this post;

- L&R sub-woofers (SWs),
- L&R woofers (WOs),
- L&R Beryllium-midrange-squawkers (MDs or SQs),
(I know that nowadays very few people use the word "squawkers", but please let me continue using "SQs" for midrange drivers.)
- L&R Beryllium-tweeters (TWs),
- L&R metal-horn-super-tweeters (STs),
Each of these is driven directly (with no LCR-network nor attenuator) by dedicated HiFi amplifier; SW has powerful built-in amplifier, and I use four stereo HiFi integrated amplifiers for others.


Hello dear ASR friends,

I (we) recently had very interesting discussions on super-tweeters on the thread entitled “Are Super tweeters worth it?”; you would please carefully read our recent posts and discussion in the post #43 through #61 on that thread.

In his post #45 there, @617 kindly shared/informed us really interesting (at least for myself) discussion which I may cite here as follows (bold red underline are added by me):
“Very interesting discussion. I think this kind of experimentation is the best instinct of the audiophile. Off-the-shelf products are designed for the masses, but only someone serious about musical enjoyment will build a system that is responsive to their exact musical taste.
One of the issues with supertweeters is that they are extremely directional, and so small changes in seating position could be the difference between hearing something and hearing nothing from them. This could explain some of the sensitivity to positioning.
I would experiment with running the supertweeter at an elevated level (perhaps up to +10db) but pointed towards the rear of the room, directly at a very hard surface, perhaps stone or glass. I think getting as much of this energy bouncing around the room would serve to create ambience.

Since I use still-amazingly-excellent highly efficient metal-horn super-tweeters FOSTEX T925A in my audio setup in very unique physical position/alignment singing together (Fq coverages are well overlapped) with YAMAHA Beryllium dome tweeters (TWs) JA-0513, his above interesting information strongly encouraged me to start a new series of audio experiments on “reflective wide-3D dispersion of super-tweeter sound using random-surface hard-heavy material”.

Intending to give you at-a-glance understanding on my present experiments, this Fig.01 would be very much suitable;
WS00007185.JPG


As for the (looks most suitable!) Bohemian crystal-glass salad bowl (14 cm diameter, 10 – 13 mm thickness, wight 850 g!), I found 10 of this bowl in our kitchen cupboard, and I cordially asked permission of my wife for using two of it for my present series of audio experiments. She kindly agreed by adding her strong warning of You should never damage or destroy them! We need to use all the10 glass bowls for serving fresh salad to all the people when our daughter’s and son’s families will come again to our home in coming summer vacation. These are really nice and cool salad bowls especially in hot summer season.”

Before to describe my experimental settings, you need to understand my SP system and room acoustics; my audio listening living room can be seen in this Fig.02;
WS00007184.JPG


You would please find the details of my present DSP-based multichannel multi-SP-driver multi-amplifier fully active stereo audio setup in my posts here #774, #858 and #895.

Please note that my listening position is not in equilateral triangle position with L&R SP system, but in isosceles triangle. We have our wide dining room open space behind between the L&R SP system (no rear bass-reflex port in SPs), and I also have acoustically-dead Japanese style tatami-floor room behind my listening position;
WS00007183.JPG
You would also refer to my post here;
- Not only the precision (0.1 msec level) time alignment over all the SP drivers but also SP facing directions and sound-deadening space behind the SPs plus behind our listening position would be critically important for effective (perfect?) disappearance of speakers: #687

I have repeatedly shared about the still-amazingly-excellent metal horn super-tweeters FOSTEX T925A (ref. #485) which are directly and dedicatedly driven by YAMAHA integrated amplifier A-S301.
WS00007182.JPG


You would please also refer to this post #435 sharing “Even Greg Timbers uses "reasonable and budget" Pioneer Elite A-20 for compression drivers (super tweeters) in his extraordinary expensive multichannel stereo system with JBL Everest DD67000 which he himself designed and developed”.

I have also repeatedly shared the reasons (and rationales?) of the very unique physical position/alignment of ST T925A in the SP system (ref. here #27);
WS00007181.JPG



Experimental settings: wide reflective 3D dispersion of ST sound using very hard random surface Bohemian crystal-glass salad bowl
After my careful/intensive consideration and very preliminary listening tests, I decided to fully evaluate these four physical positionings/alignments of the glass bowl reflector in front of ST T925A;
WS00007180.JPG


For example, the closer view of Case-2 can be seen in this Fig.07;
WS00007179.JPG


Prior to sharing my first listening session on these experimental setting, I would like to share the outline of my present audio system, especially the SPL related aspects of the TWs and STs;

The total physical setup of my audio system…
WS00007178.JPG
and the total signal path can be seen in Fig.09;
WS00007177.JPG

Now, I use VB-AUDIO MATRIX as system wide ASIO/VASIO/VAIO routing center;
WS00007176.JPG
WS00007175.JPG

As intensively shared here #485, I intensively measured the Fq dependency of SP high-level signals going into my TWs and STs; I have -12 dB/Oct high-pass (low-cut) filter in DSP EKIO at 6 kHz for TWs and STs, plus for STs I also have 3.0 microF high-pass tuning capacitor. Furthermore, for TWs and STs I have 10 microF protection capacitors which almost have no effect on Fq-SPL in the operating Fq zone of TWs and STs.
WS00007188.JPG

And, I also intensively measured the actual room air sound at my listening position for TWs only, STs only, and TWS+STs using amazingly flat-response (ref. #831) specially selected BEHRINGER ECM8000 measurement microphone;
WS00007190.JPG

In this new series of audio experiments, it is important to note/understand that the Fq coverages by TWs and STs are well overlapped (i.e. they are singing together) in my audio setting with mild-slope DSP XO filters and 3.0 microF tuning capacitor (gives further -6 dB/Oct at around 8.8 kHz) for STs. Furthermore, the ST FOSTEX T925A is a highly efficient metal horn of 108 dB/W (1 m) which means T925A sings about 15 dB louder when it receives the same gain level signal into TW YAMAHA 3.0 cm Beryllium dome JA-0513. Looking at the above Fig.12 and Fig.13, therefore, you can understand how carefully I set/tune the variable gain levels (by YAMAHA A-S301 integrated amplifier) driving ST T925A.

And, the Beryllium dome TW JA-0513 has relatively nice wide directivity, while ST T925A has rather narrower directivity; this is one of the main reasons for my great interests on 3D reflective dispersion of the sound given by ST T925A.
Edit: On these points, you would please also refer to my Part-2 post;
A new series of audio experiments on reflective wide-3D dispersion of super-tweeter sound using random-surface hard-heavy material:
Part-2_ Comparison of catalogue specifications of metal horn super-tweeter (ST) FOSTEX T925A and YAMAHA Beryllium dome tweeter (TW) JA-0513; start of intensive listening sessions with wide-3D reflective dispersion of ST sound: #921

The present best tuned total SPL curve measured at my listening position;
WS00007172.JPG

In this new series of experiments, it would be really important that I can independently control the relative gain of STs only which can be done by YAMAHA A-S301 integrated amplifier directly and dedicatedly driving STs T925A with no effect/change in DSP XO/EQ configuration in upstream DSP-EKIO;
WS00007171.JPG

The details of DSP-EKIO configuration can be seen in Fig.16;
WS00007170.JPG

Using the safe and reliable Mute/Solo buttons on the DSP-EKI’s output panels, I cay very easily, even on-the-fly, listen to the only TW+ST high-Fq sound, and since TWs are driven by SONY TA-A1ES integrated amplifier and STs are driven by YAMAHA A-S301 integrated amplifier, I can hear STs only, TWs only, as well as TWs+STs in any relative gain combinations with no effect on their upstream DSP configurations.
WS00007169.JPG

As you may well agree, the effects of wide 3D reflective dispersion of ST sound would somewhat contribute (preferably or badly) to reproduction of “stereo sound stage/image” as well as to enlarging (or narrowing) of so called sweet spot (or sweet sphere) around the center of listening position; these effects would be also greatly dependent on room acoustics environment; in my listening living room I have more or less random and asymmetrical furniture and other objects alignment/positioning.

Consequently, you would please note that it would be quite difficult to give any “generalization” of the result(s) of my present experiments to other people’s audio system/setup.

These also mean that it would be somewhat difficult to objectively measure such effect(s) using only one measurement microphone, and furthermore I have almost no capability nor advanced tools/gears to objectively measure and share the results. My information exchange on this series of experiments, therefore, would be rather subjective hearing/listening reports as you would kindly agree and accept.

“Nevertheless!”, I believe it would be worthwhile to share the results with you all, since in my initial preliminary listening tests (to be shared below in this post), I could already hear very interesting “positive/favorable” effects by the wide 3D reflective dispersion of the sound given by STs T925A.


First preliminary listening/evaluation session using “Phase/Polarity Check Track” and “Five Sound Positioning/Allocation Test Track” of SONY’s “Super Audio Check CD”
To begin with, I forced myself to hear/listen the only ST T925A sound, in L&R stereo in a little bit of high gain, using the very well prepared Track-2 “Phase/Polarity Check Track” of SONY’s “Super Audio Check CD” (ref. here #651 and the PDF booklet attached thereof.)

I know that in many home audio setup of other people, sometimes it would be rather difficult in clearly identifying the difference between “in-phase sound” and “out-of-phase sound” in super-tweeter covering Fq range of 8 kHz to 22 kHz.

In my audio setup, however, thanks to the excellent TWs and STs as well as thanks to the reasonably nice room acoustic environment, I can always clearly distinguish the difference between “in-phase” and “out-of-phase” in super-tweeter only sound with my present Case-0 reference setup (with no dispersion glass in front of STs T925A) shown in the above Fig.06.

Then, I have quickly and repeatedly tested Case-1 through Case-4 with various positioning/direction of the heavy hard random surface glass 3D reflective dispersion material, the Bohemian crystal-glass salad bowl, in front of STs T-925A.

Similar to reference Case-0 test, in each of the four cases, I could easily distinguish the difference between “in-phase” and “out-of-phase” in super-tweeter only sound, and very much interestingly the easiest case (my wife also agreed) was given by the Case-2 of reverse-straight reflection/dispersion setting.

More interestingly, the Case-2 gives the largest “sweet spot/sphere” around the center of my listening position which we (with my wife) could confirm by rotating our head and even moving 3D our body around the listening position.


Next, I quickly and repeatedly heard/listened to the only ST T925A sound, in L&R stereo in a little bit of high gain, using the very well prepared Track-3 of “Five Sound Allocation/Positioning Check by Solo Drum Roll Including Cymbals” (ref. #651). Thanks to the audio scientists/engineers in Sony Corporation at that time (1983), this test track was/is very nicely prepared for typical “five sound allocation/positioning test” not only due to best gain balances but also more importantly the “phase alignment/tuning” for the sound stage allocation.

All of the Case-0 (reference) through Case-4 settings give very nice five sound position allocation hearing at my usual listening position. We then repeated the hearing session while rotating our head as well as moving around the listening position to confirm which of the five (5) Cases would give largest “sweet spot/sphere” around the center of the listening position.

Very interestingly, the most favorable hearing result was given by the Case-2 setting of reverse-straight reflection/dispersion; the more firm/stable five sound position allocation (vs. Case-0 reference) remained really unchanged even if we moved (3D) considerably more remote from the center of the listening position.

Encouraged by the above favorable results, we repeated the hearing session by listening to the TWs+STs sound in my usual relative gain setting of “ST gain-B” shown in above Fig.15. Even with TWs+STs sound, we could confirm that the most favorable hearing result was given by the Case-2 setting of reverse-straight reflection/dispersion including the largest “sweet spot/sphere” around the center of listening position.


First preliminary listening/evaluation session in Case-2 setting with a whole CD album tracks of excellent-recording-quality smooth jazz music
For my fun and preliminary confirmation, today, during my 30 min (200 kcal) ergometer exercise/workout, I listened to a whole smooth jazz ripped CD album of “Perfect Moment” by Peter White recorded in 1998 in really nice recording quality. I cited/referred this album in my post #591 and in this post #31 on another my thread in terms of excellent reference music containing nice bass and treble sound.

The ergometer bike is set just behind my usual listening sofa, and my head position was about 50 cm rear and 60 cm higher than usual listening position sitting on the sofa. Of course, all the SP drivers, SW WO MD(SQ) TW ST, were actively singing/working as usual with my default configuration (as in SLP curve shown in Fig.14 above).

Even in this off-center listening position, I could enjoy considerably better “sound staging” as well as slightly better sound clearness/cleanliness compared to my usual Case-0 reference setting; it maybe also attributable to some initial placebo effect, though…

I “felt” better 3D sound staging around the SP systems not only left-right but also up-down as well as (more importantly) front-rear sound allocations given by this enjoyable smooth jazz album using the Case-2 reverse-straight reflective sound dispersion of the high Fq sound of STs T925A.


During the coming several months, I would like to continue my (our) comparative listening sessions with Case-0 (reference) through Case-4 reflective dispersion settings shown in Fig.06 under some variations of relative gain of STs while keeping all the other parameters should remain unchanged; we will carefully and intensively listen to all the 60 tracks of my consistent “Audio Sampler/Reference Music Playlist” (ref. #669, #670 and my remote thread entitled “An Attempt Sharing Reference Quality Music Playlist: at least a portion and/or whole track being analyzed by 3D color spectrum of Adobe Audition”).

My another thread entitled “Music for Testing Treble (High Frequency) Sound” would be also of your reference and interest relating to this new series of audio experiments on ST sound dispersion.


In any way, after having my very limited preliminary 2-day experiments, I now feel and assume that the Case-2 reverse-straight reflective dispersion configuration for T925A would be interesting and preferable at least for me and for my wife in our own listening room acoustics/environments…

Please stay tuned on my further coming posts on this new series of audio experiments, and I would also highly appreciate hearing your comments suggestions and advices on my present ongoing experiments.
 
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617

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Thank you for pointing me towards your experiments. I love seeing this kind of adventurousness on ASR, it is far more interesting than seeing people argue about what to buy.

One thing I would suggest is that while we can be very particular about what we perceive, it is sometimes too easy to ascribe a hypothesis about the reason why we hear what we hear. I think when you are tweaking your sound like this, it is a mistake to hypothesized why it sounds a certain way - the sound field you are creating is complex, almost alive, and it may be beyond our ability to measure and understand.

One of the most intelligent people in audio, although controversial, is romy the cat. His basic ethic, as far as I can understand, is to embrace subjectivity in audio, but to apply the discernment and personal taste the way a great artist does. Great art is never accomplished when the artist tries to make everyone happy; music created by a focus group would be terrible, yet we accept this kind of middling, blinkered ethic in audio, making the speakers for everyone, but thrilling nobody. Well, those speakers may be excellent, but they should not prevent people like you and I from exploring music through crystal bowls!

By the way, ten bowls might be perfect for a multi channel set up.
 

Sal1950

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Please stay tuned on my further coming posts on this new series of audio experiments, and I would also highly appreciate hearing your comments suggestions and advices on my present ongoing experiments.
Best of luck and have fun.
I can only make one comment-suggestion, DON'T BREAK either of the bowls.
You will NEVER live it down. :p
Cheers
 
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dualazmak

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- the sound field you are creating is complex, almost alive, and it may be beyond our ability to measure and understand.

Thank you for your very quick response! Yes, I fully agree your point.
As I wrote in my post above, I would like to share my subjective listening experiences/impressions in my audio setup in my acoustic environments.

I hope that at least the approach which you kindly suggested and I am testing would be of interest for other people...
 

TheBatsEar

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You should never damage or destroy them!
It seems you wife is exceptionally tolerant when it comes to stuff that is important to you.
Let's be honest, your setup looks more like a space control center than a laid back music listening spot.

Anyway, great work as usual, you certainly go the extra mile to get the most out of the hobby.

I recently bought two multichannel DACs as well, the MiniDSP Flex HT and the HifiBerry DAC 8x.
The flex will be used on my PC rig for multichannel games with Genelec speakers, but the second will be used on my hifi system.

Sadly the Yamaha NS-1000 mid driver became unobtainium around here, so the speakers in the Wharfedale Linton 85 and two subwoofers will have to suffice for now.
The plan is to get them "activated" like your setup, but i don't have the right amplification yet.

I also lack the courage to open up the Wharfedale speakers, but i'm working on it ;-)
 

Roland68

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Hello SSS and gene_stl:

Thank you for your interests and comments which I understood well.

At least in my setup, of course I carefully measured and confirmed that the protection capacitors are essentially transparent and "inaudible"; please refer to my posts here #402 and #485 (as well as #258 for subjective comparison).

I use these protection capacitors for protecting my treasure Be-midrange-squawkers, Be-tweeters, and metal horn super-tweeters from possible accidental intrusion of low Fq signal and/or DC.

In the course of intensive DIY of DSP-based multichannel multi-SP-driver multi-amplifier "fully active" audio setup, we have several possibilities of such unexpected "accidents", e.g. mis-configuration/mis-typing in XO/EQ/Group-Delay/Gain settings, mis-connection of line-level and/or SP-high-level cables, unexpected pop due to "excessively buffered" signal intrusion when changed the XLR cables, etc., etc.

I actually experienced a few such cases (even I have been always much careful though) that the protection capacitors actually did their job perfectly protecting my treasure SP drivers. Consequently and fortunately, I never lost/damaged my treasure SP drivers thanks to the protection capacitors.

I also understand @SSS's concerns on "relays" within amplifiers which are relating to QC, durability and warranty of amplifiers; I mean which and what relays the manufacturer would select and use (in some cases they use oxygen-free pressurized nitrogen or SF6 filled fully shielded/covered relays), as well as maintenance service availabilities even after the warranty period for long years. This issue wound be one of the critical factors for our amplifier selection. At least in my case, I very much carefully selected my four amplifiers in this respect too. You would please refer to my summary post here. Fortunately, my amplifiers are still in excellent perfect healthy conditions. My post here would be also your interest and reference, I assume.

Accuphase, Yamaha and Sony are still providing nice maintenance/repair services for any of their present and past products with reasonable cost, and we also have several domestic third-party maintenance/repair firms for these amplifiers.

BTW, just for your convenience and further overview, you can find here (on this thread) and here (remote independent thread post) the Hyperlink Index for this project thread.
I just read this and wanted to point out that there are several projects in the diyaudio forum for speaker protection circuits with SSR relays.
Maybe these circuits are fast enough for sensitive tweeters. In addition, these SSR circuits eliminate the usual limitations and disadvantages of relays in the signal path.
 
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dualazmak

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I just read this and wanted to point out that there are several projects in the diyaudio forum for speaker protection circuits with SSR relays.
Maybe these circuits are fast enough for sensitive tweeters. In addition, these SSR circuits eliminate the usual limitations and disadvantages of relays in the signal path.

OK, thank you for the information.
At least I myself do stick to, however, safe and (almost completely) transparent rather affordable 10 microF capacitors to protect my tweeters and super-tweeters.
 

BR52

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OK, thank you for the information.
At least I myself do stick to, however, safe and (almost completely) transparent rather affordable 10 microF capacitors to protect my tweeters and super-tweet
And it's more reliable, because it's passive. SSR's can introduce some nonlinearities as well.
 
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