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

[dualazmak]

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.

 

[dualazmak]

Crowbar circuits and relays may not be fast enough to protect tweeters.

Yes, fully agree with you!

 

[dualazmak]

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:

Spoiler

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.

 

[dualazmak]

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.

 

[Neddy]

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'.

 

[dualazmak]

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.

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

(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:

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

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

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

 

[dualazmak]

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.

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.

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!

 

[gene_stl]

Yokogawa makes nice instrumentation.

 

[dualazmak]

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?"

​

 

[dualazmak]

I have just posted on another thread entitled "Music For Testing Bass";
- A nice smooth-jazz album for bass (low Fq) and higher Fq tonality check and tuning

 

[dualazmak]

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:

Spoiler: Summary__ rationales and pros of analog-level relative gain (tonality) controls

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.

Spoiler

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, please let me enjoy music based on my preferences and my tuning style!

Spoiler

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!

Spoiler

- 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:

Spoiler: 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).