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

zergxia

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I measured and calibrated the subwoofer's delay using the method described by dualazmak (layer 507).

As you can see, it is not easy to correctly distinguish the pulse shape due to the highly reflective nature of the small room. But I still adjusted it as best I could.

I used the Equalizer APO for the FIR crossover of the subwoofer, and also used the overall EQ with linear phase.
Through this test, I found that the channel delay displayed by the Equalizer APO is not accurate. The actual channel delay cannot be compensated based on the delay value given by the software - the measured error is as high as 16ms!

Thanks dualazmak, the method you shared is awesome!

QQ图片20220317102655.png
 
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dualazmak

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Hello friends, Hello @zergxia,

We, me and zergxia, had communication (posts here and here) on the thread entitled "What is group delay?" followed by our PM communication informing and sharing the test tone signals which I prepared and utilized in my posts;

- Precision measurement and adjustment of time alignment for speaker (SP) units: Part-1_ Precision pulse wave matching method: #493

- Precision measurement and adjustment of time alignment for speaker (SP) units: Part-2_ Energy peak matching method: #494

- Precision measurement and adjustment of time alignment for speaker (SP) units: Part-3_ Precision single sine wave matching method in 0.1 msec accuracy: #504, #507

- Measurement of transient characteristics of Yamaha 30 cm woofer JA-3058 in sealed cabinet and Yamaha active sub-woofer YST-SW1000: #495, #497

- Identification of sound reflecting plane/wall by strong excitation of SP unit and room acoustics: #498

I thank you for @zergxia's interests on my unique methods.

Well, @zergxia,
I assume you understand everything well, but let me remind you just for sure as follows;

If your sub-woofer is delaying in 16 msec, it cannot be corrected by DSP, but you may intentionally "delay" other drivers by DSP in 16 msec to get the perfect time alignment between all the SP drivers.

The most delayed SP driver (usually it is sub-woofer) would be the base "zero time" unit, and the other drivers should be forced to daly by DSP to time-align (time-match) with the most delayed SP unit. I assume you understand well and did the adjustment accordingly.

I agree with you that the advanced(?) sound software, like Equalizer APO and REW, includes some "black-box" type delay and time alignment measurement and correction feature, but I often found that they would not provide accurate measurement and adjustment mainly because of many buffer(s) and latencies included in the measurement system including the audio interface for microphone. These drawbacks usually cannot be avoided if the method would be based on some "absolute delay" measurement approaches.

In our multichannnel and/or multi-SP-driver audio system, however, the "relative delay" or "relative time alignment" between the SP drivers should be important, and should be measured and adjusted. This is why I developed my own (primitive) method for time alignment by using the specially prepared series of tone burst with exact time location, and we can measure the "relative delay" by using only the microphone recorded air sound.

Looking at your air sound spectrum, I would like to suggest that you may better to perform the test(s) with reduced sound volume to minimize the room reflection(s) and standing wave(s); for the time alignment measurement and adjustment, only the precise time-point identification of the "tone kick-up", and the reading of the kick-up time-point would be easier with minimum room effects such as sound reflections, standing waves, etc.

Which microphone and ADC (audio interface) are you using?
 
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zergxia

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I use DAYTON AUDIO UMM-6 which is a USB microphone.
I couldn't go any further to increase the mic gain, so I had to turn up the speaker volume appropriately, otherwise the test signal would be drowned out in the background noise.


111.jpg

As shown in the picture, I used Equalizer APO for FIR convolution operation, and the software gives the channel delay (estimated value).


222.jpg

Before, I believed them too much and set the delay compensation value directly based on them. I never suspected it was wrong before using the method you shared.
 
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dualazmak

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I use DAYTON AUDIO UMM-6 which is a USB microphone.
....
Before, I believed them too much and set the delay compensation value directly based on them. I never suspected it was wrong before using the method you shared.

Thank you for your input and comments.

I agree with you that it is critically important to "measure and tune" the time alignment and other sound characteristics using the recorded air sound. As far as I can, I would like to perform the measurement(s) and fine tuning with no or minimum (and understandable) incorporation of "black box" type software manipulation or signal handling, as I have been sharing throughout this project thread.

I believe it is also important that we always need to establish objective "validation" procedures plus evidences for any of the sound measurement methods we would like to apply in our audio system tuning.
 
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dualazmak

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Can your system play CSGO? This game supports 5.1.

Even though I assume that I may configure my multichannel system for playing PC games support 7.5 and/or 5.1 surround audio, I have little interest on PC games.

So far, and also at present and in future, my multichannel multi-driver multi-way multi-amplifier stereo system is dedicating for pure HiFi stereo music listening.
 

Doodski

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Even though I assume that I may configure my multichannel system for playing PC games support 7.5 and/or 5.1 surround audio, I have little interest on PC games.

So far, and also at present and in future, my multichannel multi-driver multi-way multi-amplifier stereo system is dedicating for pure HiFi stereo music listening.
Heheh. I am a first person shooter aficionado but CSGO is a gong show for me. It's nuts! :D I like more stealthy stuff.
 

Sal1950

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So far, and also at present and in future, my multichannel multi-driver multi-way multi-amplifier stereo system is dedicating for pure HiFi stereo music listening.
Same here, I listen to music on my 5.2.4 system at least 3-4 hours a day. On the other hand I may watch one movie every couple weeks or so, the largest part of which are Sci-Fi or Action-Adventure flicks.
I've no time for games or vinyl records. ;)
 

adLuke

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Again, "Near ultrasound - ultrasound" ultra-high frequency (UHF) noises in HiRes music tracks, and EKIO's XO-EQ configuration to cut-off such noises

Hello friends,

I rather intensively investigated and shared this topic and my counter measure; "Near ultrasound - ultrasound" ultra-high frequency (UHF) noises in HiRes music tracks, in my posts #362-#386.

Although quite belated, today I noticed this important and interesting thread;
SOUND LIAISON, PCM DXD DSD free compare formats sampler. A new 2.0 version.
https://www.audiosciencereview.com/...pare-formats-sampler-a-new-2-0-version.23274/
where Sound Liaison generously described about sharing "free compare formats sampler. A new 2.0 version" for which they wrote that "We tried (and succeeded quite well) in recreating the sound the analogue vintage equipment added to the voice by using the tools we have in our digital workstation. But now without added high frequency noise."

These "free compare formats sampler. A new 2.0 version" tracks were newly prepared by Sound Liaison after amirm's specific post and amirm's YouTube video clip on Sound Liaison's original "compare formats sampler (1.0 version)".

Today I shared the following information in my post here on the interesting remote thread entitled "DSD is better than PCM!", and I do hope it would be allowed to share also in my project thread here, since the topic and information are still closely related to my system setup and configuration.

Now, I am just very much interested in "quality control, QC" of these "SOUND LIAISON free compare formats sampler 2.0 version", I quickly analyzed them by using MusicScope 2.1.0, as follows;
View attachment 185614

View attachment 185615

View attachment 185616

We may clearly see that these "version 2.0 compare formats sampler" tracks contain much less UHF (ultra high frequency) "noises" in comparison with their original version 1.0 tracks.

We can also find, however, that the HiRes format tracks still have "sound" or "noise" components beyond 25 kHz, I mean in 25 kHz to 176.4 kHz frequency area. Since I (we) can hear up to ca. 22 kHz, the "value" (and "meaning") of the sound over 30 kHz would be the subject of our further discussion.

At least in my latest audio system setup and configuration, PCM 88.2 kHz WAV or FLAC 88.2 kHz (up to 44.1 kHz in each of the L & R stereo channels) would be "sufficient" enough, since the digital XO/EQ "EKIO" can work up to 192 kHz 24 bit and also having protective -48 dB/Oct high-cut (low-pass) LR filters at 25 kHz to eliminate any UHF sound/noises over 30 kHz (please refer to my posts #362-#386).

Consequently, even though my digital music library (ca. 25,000 tracks) consists of so many formats of DSD (8x, 4x, 2x,1x) FLAC (44.1 kHz- 352.8 kHz) and WAV/AIFF (44.1 kHz - 196 kHz), now I usually set "JRiver's DSP studio" in 88.2 kHz output of all the music tracks stereo 2-channel (on-the-fly conversion) as well as I set EKIO in 88.2 kHz sampling rate.

The simple reason for my current choice of 88.2 kHz, instead of 96 kHz, is that majority of the tracks in my music library is CD ripped non-compressed 44.1 kHz 16 bit AIFF format. As for my music library organization strategy and policy, you would please refer to my post here in a remote thread.

And, I should not forget about sharing the important subjective impression that all of the 15 tracks in "SOUND LIAISON, PCM DXD DSD free compare formats sampler. A new 2.0 version" sound really amazingly nice!
Hello @dualazmak san,
thank you so much for sharing this monumental project.
I read all of your thread, but I guess I will have to go back to it many times before I can truly understand in full all the details inside it.
I would have a tonne of questions, but most of them are due to my ignorance and it is much better if I dig around the net to fill my gaps.
One thing I did not understand about the way you play your music: do you feed the music files to the DAC in the same format they originally come (PCM/DSD) or do you let JRiver MC do the conversion before sending data to the DAC?
I am asking because a post on mojo-audio.com (https://www.mojo-audio.com/blog/dsd-vs-pcm-myth-vs-truth/) strongly, very strongly, suggests feeding each DAC chip its 'native' format (just a few paragraphs before the summary you can find the reasoning behind the suggestion).
Maybe with the Sound Liaison, PCM DXD DSD... you could really tell if mojo-audio advice is really correct.
Hope this is not a redundant question and may help somehow in the quest for a great audio experience.
Thank you very much again for sharing all your efforts.
 

Newman

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I reckon that mojo-audio article deserves its own fact-check thread….really.
 
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dualazmak

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Hello @dualazmak san,
thank you so much for sharing this monumental project.
I read all of your thread, but I guess I will have to go back to it many times before I can truly understand in full all the details inside it.
I would have a tonne of questions, but most of them are due to my ignorance and it is much better if I dig around the net to fill my gaps.
One thing I did not understand about the way you play your music: do you feed the music files to the DAC in the same format they originally come (PCM/DSD) or do you let JRiver MC do the conversion before sending data to the DAC?
I am asking because a post on mojo-audio.com (https://www.mojo-audio.com/blog/dsd-vs-pcm-myth-vs-truth/) strongly, very strongly, suggests feeding each DAC chip its 'native' format (just a few paragraphs before the summary you can find the reasoning behind the suggestion).
Maybe with the Sound Liaison, PCM DXD DSD... you could really tell if mojo-audio advice is really correct.
Hope this is not a redundant question and may help somehow in the quest for a great audio experience.
Thank you very much again for sharing all your efforts.

Summary of rationales for "on-the-fly (real-time)" conversion of all music tracks (including 1 bit DSD tracks) into 88.2 kHz or 96 kHz PCM format for DSP (XO/EQ) processing

Hello @adLuke san,

Welcome to this project thread! Your above inquiry is nice and important point, indeed.

My present answer for you is "It is quite feasible enough and even ""needed"" to feed all the audio digital signals in 88.2 kHz or 96 kHz PCM (or 192 kHz, if you like) by JRiver's on-the-fly format conversion to be sent into DSP (XO/EQ) software EKIO. "

Various background and justifications for this answer are as follows;

Before starting this project, I had been enjoying music with ordinary PC audio setup with one DAC (OPPO Sonica DAC)) and one HiFi integrated amplifier (ACCUPHASE E-460) driving all the SPs through passive LC (inductors capacitors resistors) network. And I had been sticking to "native format feed" into OPPO Sonica DAC up to 1-bit/DSD256(4x), as you kindly pointed.

When I started considering possible multichannel multi-driver multi-way multi-amplifier project with software DSP (XO/EQ), I did intensive search and desk evaluations on various DSP software solutions, and I found the maximum PCM processing format is 192 kHz 24 bit in these DSP software solutions. (Even with the extraordinary expensive TRINNOV ALTITUDE 32 DSP processor, actually having PC in it, the internal DSP processing is up to 192 kHz).

I carefully considered the pros and cons of "DSP processing all tracks in 192 kHz or 96kHz" instead of "native format feed", and concluded that multichannel multi-amplifier approach would surpass the cons, at least in my system setup with still amazingly wonderful Yamaha SP drivers and cabinet.

Consequently, I decided to go into "multichannel multi-amplifier" world of "max. 192 kHz 24 bit processing", as you kindly have read through this project, including the "all in max. 192 kHz ASIO I/O within PC".

Then, rather recently, I (we) fully discussed and evaluated the UHF (ultra-high frequency) noise issue in poorly QC-ed HiRes music tracks including DSD formats, as you clearly noticed;
- "Near ultrasound - ultrasound" ultra-high frequency (UHF) noises in improperly engineered/processed HiRes music tracks, and EKIO's XO-EQ configuration to cut-off such noises: #362-#386, #518
I wrote that such a high amount of UHF noises would be "possibly" harmful (and useless, meaningless) for our tweeters and super tweeters. I also pointed they would be highly possibly harmful for our beloved pets including dogs, cats, birds.

Having my intensive objective measurements of these "poorly QC-ed" HiRes tracks, and having so many intensive discussions on "enough PCM bit rate in HiFi audio", now I conclude that 88.2 kHz or 96 kHz processing (i.e. up to 44.1 kHz or 48 kHz in L and R channels) would be just enough and feasible in my setup (and I believe so also in your setup) since I decided always having high-cut (low-pass) -48 dB/Oct filters at 25 kHz in my EKIO configuration to cut-off any of the possible UHF noises very frequently existing in HiRes tracks.

This means that I have finally landed on agreement with @mikessi's "enlightenment and belief" of "There is really no audible benefit to playback beyond 24/96 sampling, especially with any recordings other that those done with the most advanced high res gear and high fidelity values." 

Another important aspect of this issue would be relating to our hearing ability in high frequency zones. Recently, I participated in the interesting thread entitled "Audio Listening With Age Diminished Hearing". You would please read my posts #70, #72 and #74 on that thread.

BTW, as I wrote here, here and here, my digital music library of about 25,000 files consists of mixture of various formats;

16-bit/44.1kHz CD ripped non-compressed aif (majority!),
24-bit/192kHz down-sampled or up-sampled aif,
24-bit/96kHz flac,
24-bit/192kHz flac,
1-bit/DSD64(1x) 2.8MHz dsf,
1-bit/DSD128(2x) 5.6 MHz dsf,
1-bit/DSD256(4x) 11.2 MHz dsf,

and now JRiver MC feeds all of the tracks usually (mainly) in 88.2 kHz 24 bit (i.e. max. 44.1 kHz Fq window in 2-ch stereo) by on-the-fly conversion into EKIO for crossover/EQ processing. As I have high-cut (low-pass) -48 dB/Oct LR filters at 25 kHz, max. 44.1 kHz in L & R channels are more than enough.
 
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adLuke

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Tha
Hello @adLuke san,

Welcome to this project thread! Your above inquiry is nice and important point, indeed.

My present answer for you is "It is quite feasible enough and even ""needed"" to feed all the audio digital signals in 88.2 kHz or 96 kHz PCM (or 192 kHz, if you like) by JRiver's on-the-fly format conversion to be sent into DSP (XO/EQ) software EKIO. "

Various background and justifications for this answer are as follows;

Before starting this project, I had been enjoying music with ordinary PC audio setup with one DAC (OPPO Sonica DAC)) and one HiFi integrated amplifier (ACCUPHASE E-460) driving all the SPs through passive LC (inductors capacitors resistors) network. And I had been sticking to "native format feed" into OPPO Sonica DAC up to 1-bit/DSD256(4x), as you kindly pointed.

When I started considering possible multichannel multi-driver multi-way multi-amplifier project with software DSP (XO/EQ), I did intensive search and desk evaluations on various DSP software solutions, and I found the maximum PCM processing format is 192 kHz 24 bit in these DSP software solutions. (Even with the extraordinary expensive TRINNOV ALTITUDE 32 DSP processor, actually having PC in it, the internal DSP processing is up to 192 kHz).

I carefully considered the pros and cons of "DSP processing all tracks in 192 kHz or 96kHz" instead of "native format feed", and concluded that multichannel multi-amplifier approach would surpass the cons, at least in my system setup with still amazingly wonderful Yamaha SP drivers and cabinet.

Consequently, I decided to go into "multichannel multi-amplifier" world of "max. 192 kHz 24 bit processing", as you kindly have read through this project, including the "all in max. 192 kHz ASIO I/O within PC".

Then, rather recently, I (we) fully discussed and evaluated the UHF (ultra-high frequency) noise issue in poorly QC-ed HiRes music tracks including DSD formats, as you clearly noticed;
- "Near ultrasound - ultrasound" ultra-high frequency (UHF) noises in improperly engineered/processed HiRes music tracks, and EKIO's XO-EQ configuration to cut-off such noises: #362-#386, #518
I wrote that such a high amount of UHF noises would be "possibly" harmful (and useless, meaningless) for our tweeters and super tweeters. I also pointed they would be highly possibly harmful for our beloved pets including dogs, cats, birds.

Having my intensive objective measurements of these "poorly QC-ed" HiRes tracks, and having so many intensive discussions on "enough PCM bit rate in HiFi audio", now I conclude that 88.2 kHz or 96 kHz processing (i.e. up to 44.1 kHz or 48 kHz in L and R channels) would be just enough and feasible in my setup (and I believe so also in your setup) since I decided always having high-cut (low-pass) -48 dB/Oct filters at 25 kHz in my EKIO configuration to cut-off any of the possible UHF noises very frequently existing in HiRes tracks.

This means that I have finally landed on agreement with @mikessi's "enlightenment and belief" of "There is really no audible benefit to playback beyond 24/96 sampling, especially with any recordings other that those done with the most advanced high res gear and high fidelity values." 

Another important aspect of this issue would be relating to our hearing ability in high frequency zones. Recently, I participated in a interesting thread entitled "Audio Listening With Age Diminished Hearing". You would please read my posts #70, #72 and #74 on that thread.

BTW, as I wrote here, here and here, my digital music library of about 25,000 files consists of mixture of various formats;

16-bit/44.1kHz CD ripped non-compressed aif (majority!),
24-bit/192kHz down-sampled or up-sampled aif,
24-bit/96kHz flac,
24-bit/192kHz flac,
1-bit/DSD64(1x) 2.8MHz dsf,
1-bit/DSD128(2x) 5.6 MHz dsf,
1-bit/DSD256(4x) 11.2 MHz dsf,

and now JRiver MC feeds all of the tracks usually (mainly) in 88.2 kHz 24 bit (i.e. max. 44.1 kHz Fq window in 2-ch stereo) by on-the-fly conversion into EKIO for crossover/EQ processing. As I have high-cut (low-pass) -48 dB/Oct LR filters at 25 kHz, max. 44.1 kHz in L & R channels are more than enough.
Thank you very much @dualazmak san for your detailed answer.
The reasoning behind all of the choices is very clear and detailed.
Your way of explaining is of great clarity and has helped me learn a lot throughout your thread.
One thing I am missing is: in the article I quoted before (by mojo-audio) it was suggested to feed delta/sigma based DACs only with 1 bit DSD encoded music (also he mentioned multibit DSD music but that is not relevant here).
So this would mean that JRiver should convert first all the flac and aif files to dsf format on the fly before feeding the DAC.
At least this is my understanding. Hope I am helping to clarify instead of confusing the matter:rolleyes:.
 
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dualazmak

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Hello again, @adLuke,
So this would mean that JRiver should convert first all the flac and aif files to dsf format on the fly before feeding the DAC.

Yes, this is exactly what I was mainly doing before this multichannel multi-amplifier project in my audio setup like;

JRiver (all in DSDx2) --> ASIO USB driver --> OPPO SONICA DAC (or OKTO DAC8PRO as two channel DAC) --> ACCUPHASE E-460 --> LCR passive network in outer box --> all the SP drivers

As you can find in many of my posts in this thread, above "single DAC --> single Amp --> passive LCR network --> SP drivers" has been my "reference sound system" all the way through this project thread.

I still keep, therefore, the above reference sound setup to which I can roll back my entire system very easily, within 10 minutes, by using the outer LCR box (#250) and the SP cabling/switching board ( #004, #137, #250).

Just for example, you would please find such "roll-back and go-advance" comparative listening sessions in my posts #253 #258 #265 (evaluating AHB2) , and #307 #308 #309 (evaluating A-S3000 and TA-A1ES).

I believe that it is critically important having/keeping "concrete reference sound system" in this kind of step-by-step multichannel multi-amplifier audio project.
 
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dualazmak

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My nostalgia and preference for large glass-face VU meters: DIY of 12-VU-Meter Array in multichannel multi-driver multi-way multi-amplifier stereo audio system

Abbreviations in this post;
SW: sub-woofer, Yamaha YST-SW1000 (L & R)
WO: woofer, Yamaha NS-1000’s 30 cm JA-3058
SQ: squawker, Yamaha NS-1000’s 8.8 cm Beryllium dome JA-0801
TW: tweeter, Yamaha NS-1000’s 3 cm Beryllium dome JA-0513
ST: super-tweeter, Fostex metal horn T925A

Hello friends,

I always have profound nostalgia and preference for physical large glass-face VU meters. During the past several months, therefore, I have been enjoying my participation on the thread entitled "VU Meters: Let's See 'Em!!" and on the thread entitled " Douk VU3 Review (VU Meters)".

Now, as a follow-up of my posts here, here and here, I just completed my DIY of 12-VU-Meter Array using very attractive large glass-face VU meter NISHIZAWA R-65 in my present multichannel multi-driver multi-way multi-amplifier stereo system;
WS003747.JPG


WS003746.JPG


In this post, I am happy to describe and share the details of this DIY-built 12-VU-Meter Array.

I assume many of you who periodically visit this project thread are well aware of that I use four HiFi integrated amplifier in this project; YAMAHA A-S3000 (driving WOs), ACCUPHASE E-460 (driving SQs), SONY TA-A1ES (driving TWs) and YAMAHA A-S301 (driving STs). (My heavy L&R sub-woofers YAMAHA YST-SW1000 has built-in amplifiers in it.)

Among the four integrated amplifiers, E-460 has nice beloved beautiful log peak power meters (not pure VU meters), A-S3000 has really nice meters which can be switched either of log peak programme (PP) mode and pure VU mode. TA-A1ES and A-S301, on the other hand, do not have any VU/PP meter.
WS003692.JPG


You may agree with me that pure VU meter mode, not the PP mode, would fit very well with our air sound listening sensations; I have been always looking for the "same" nice large glass-face physical VU meters for monitoring the SP high level signals from these four amplifiers. I would like to have the relative gains of such possible VU-meter Array should fit well with the sound pressure level (gain) of actual room air sound at my listening position.

Nowadays, it has become more and more difficult, however, to find and purchase high-quality reliable large glass-face VU meters in the market. And this is why I recently participated on the thread entitled "VU Meters: Let's See 'Em!!"

As I shared here and here, it was my great pleasure finding the availability of NISHIZAWA VU Meter R-65 kit in Japan. The professional-use large glass-face VU meter R-65 (W111 mm x H77 mm: glass-face W100 mm x H40 mm) has already disappeared from the product list of Nishizawa Electric Meters Manufacturing Co., Ltd. (NEMM Co., Ltd.). Very fortunately, however, a small company SUZUDES10 issued a special contract with MEMM Co., Ltd. for small scale production and supply of their VU meter R-65, and SUZUDES10 replaces the bulb backlights with color LEDs. SUZUDES10 is also selling the R-65 VU meter unit of original bulb light as well as four different color LED ones;
http://suzudes10.shop-pro.jp/?mode=cate&cbid=2774641&csid=0

For the LED color variations, SUZUDES10 also contracted with NEMM Co., Ltd. to have white scale plate in R-65, instead of the original pale yellow one, for better LED color representations.
https://ameblo.jp/suzudes20/entry-12716821866.html

I am very happy finding such a small niche VU meter manufacturer is still alive and active in Japan, even though I do not know whether they would accept purchase order from abroad or not.

In any way, I first purchased one R-65 VU meter kit of special high-grade orange 2700K LED version, and fully tested and evaluated;
WS003574.JPG


The VU Meter R-65 and its dedicated VU amp board ATV205EXT are essentially compatible with the IEC 60268-17 VU meter Standard/Specification (3,600 Ohm, AC1.228 V0 - VU 600 Ohm: 0 VU with 1.228 V 1000 Hz, ballistic behaviors as specified by IEC), and the VU amp board ATV205EXT has further VR adjustment capability to fit with line-level input of 30 mV to 10 V, and the back panel VRs can adjust plus/minus 10 dB or more. The actual current consumption by the VU amp circuit board is about 35 mA inclusive of the LED lighting.
Edit: Please also refer to my post #545 for compatibilities with IEC 60268-17.
WS003772.JPG


On the board, we can find a JRC 2073D amp module and a JRC 78M06A three-terminal regulator.

According to their page here, they have voltage limiter circuit on the board protecting the nice VU meter R-65 from overshoot load, i.e. higher than 1.22V 0VU value at the internal 3,600 Ohm resistance.

This VU R-65 kit is powered by AC 12V 0.5A (AC-AC adaptor), and not by DC 12V. I have discussed with them about why not by DC 12V, and they said that, for beginners DIY construction, AC 12V should be better than DC 12V avoiding the possible wrong reverse polarity connection in case of DC 12V power supply. On the board, however, they also have DC 12V power supply terminals next to the yellow-cable AC 12V terminals. As you can see, the kit board also has protection fuse chip in the AC 12V power supply yellow lines.

In total, I found the VU meter R-65 and the board ATV205EXT are well designed with nice and durable components.

Using the one kit purchased (two R-65 VU meters for L & R channels), I fully tested and evaluated the VU functionalities, especially the wide flexibility of input level adjustment, in line-level signals, in headphone out signals, and also in SP high-level signals after high-to-low conversion into RCA line level by Audio-Technica's AT-HLC-130 (for which I will share in detail afterwards). Through these test and evaluation, I concluded that I can use 6 (six) of this VU R-65 kit for my DIY construction of 12-VU-Meter Array to be used in my present multichannel multi-driver multi-way multi-amplifier stereo audio setup.

I understand well that you are much curious about which signals I would like to monitor in my setup by using the DIY 12-VU-Meter Array.

1. Total sum of the digital signal (15 Hz - 25 kHz) coming from JRiver into EKIO for DSP XO/EQ processing, i.e. whole sum of digital through signal into EKIO, which should be properly DA-converted to be monitored by VU-01(L) and VU-02(R).

2. Line level crossover-ed ultra-low Fq signal (15 Hz - 50 Hz) driving active sub-woofer YAMAHA YST-SW1000 to be monitored by VU-03(L) and VU-04(R).

3. SP high-level signal of YAMAHA A-S3000 (45 Hz - 600 Hz) driving WOs to be monitored by VU-05(L) and VU-06(R) by using a high-to-low converter.

4. SP high-level signal of ACCUPHASE E-460 (600 Hz - 6 kHz) driving SQs to be monitored by VU-07(L) and VU-08(R) by using a high-to-low converter.

5. SP high-level signal of SONY TA-A1ES (6 kHz - 25 kHz) driving TWs to be monitored by VU-09(L) and VU-10(R) by using a high-to-low converter.

6. SP high-level signal of YAMAHA A-S301 (ca. 8.8 kHz - 25 kHz) driving STs to be monitored by VU-11(L) and VU-12(R) by using a high-to-low converter.

Furthermore, I set three strict conditions for these VU monitoring in my setup;

- The VU-meter configurations should "never" deteriorate at all the present excellent total sound quality of the system.

- The VU-meters should be placed in the end of the signal paths; the through-out signals from the VU boards should "never" be used for further listening purposes (except for special rare measurement cases, though).

- The DIY 12-VU-Meter Array should be easily moved to various positions in my listening room depending on the nature of my (our) listening and/or tuning sessions; accordingly, all of the stereo RCA cables (7 stereo RCA cables including one auxiliary/spare cable) and the one AC 12V PS cable going into the 12-VU-Meter Array should be about 7 m long, and the total weight should be small enough for easy position changes.

As for the AC 12V PS, I purchased and validated one AC 12V 5A transformer-type AC adaptor (INPUT: 100VAC 50/60Hz, OUTPUT: 12VAC 5A 60VA, max.41A) which can distribute AC 12V to all of the six ATV205EXT VU amp boards. I extended its 12VAC output cable in 7 m connecting to the PS distribution terminals in the 12-VU-Meter Array.
WS003771.JPG


Now, under these requirements, how can I get signals in my setup to be monitored by VU meters?

1. VU-01(L) and VU-02(R)
I can create new two output panels in DSP EKIO's configuration feeding all the Input-1(L) and Input-2(R) digital signals into them, with no crossover/EQ configuration at all except for the UHF (ultra-high frequency) noise cut off filter (48 dB/Oct high-cut [Low-pass] LR filter at 25 kHz), and can feed the digital signals (L & R) into my tiny spare DAC KORG DS-DAC-10 for only VU monitoring of its RCA line level output. Of course, EKIO can feed the digital signal into DS-DAC-10 through KORG's dedicated ASIO USB driver. DS-DAC-10's specifications, including the 10Hz - 40kHz +/-1dB Fq spec, are just more than enough only for the VU meter monitoring of the whole sum of the signal which EKIO is handling. In this case, although DS-DAC-10 is not fully in sync with my main OKTO DAC8PRO, I well tested and confirmed that there is no sync/async issue in just VU monitoring of the whole sum signal even for a continuous long time (over 12 hours) listening/tuning sessions.

2. VU-03(L) and VU-04(R); 15 Hz - 60 Hz driving SWs
In present system configuration, EKIO's crossover-ed L&R ultra-low Fq signal (15 Hz - 60 Hz) goes into DAC8PRO's CH1+CH2, and the L&R active sub-woofers YAMAHA YST-SW1000 receive the analog line-level signal by unbalanced RCA cable from the headphone line-out of DAC8PRO (CH1+CH2 under DAC8PRO's preamp/volume control) using a DAP XCALIBER XGA-18 stereo TRS-to-RCA adaptor. This means the DAC8PRO's CH1+CH2 XLR balanced out is open which can be used for VU meter monitoring using XLR-to-RCA adaptor cable (pin No.3 floated/non-connected) even if loosing 6 dB gain. The 6 dB gain loss can be easily compensated by the wide-range VRs on the ATV205 VU meter amp board.

3. VU-05(L) and VU-06(R); 45 Hz - 600 Hz driving WOs by YAMAHA A-S3000
I really would like to monitor the actual SP high-level signal driving WOs. Fortunately, A-S3000 is capable of bi-wiring SP drive having SP-A and SP-B high-level outputs. I can use SP-B high-level output into Audio-Technica's High-to-Low converter AT-HLC130 to give line-level RCA signal to be connected to the ATV205 VU meter board. Looking from amplifier, AT-HLC130 is minimal 600 Ohm load, and, therefore, no effect at all for the main SP-A output driving WOs.

It would be also nice that, if needed, I can disconnect the VU monitoring using the SP selector of YAMAHA A-S3000; SP-A, SP-B or SP-A+B. The SP-B selection disconnecting SP-A WO is also nice for possible VU meter monitoring of the signal "in silence", also if needed.

Audio-Technica's AT-HLC130 is a simpler version of AT-HLC150 which I fully evaluated and validated in my posts #401, #402 and #485. The only difference between AT-HLC130 and AT-HLC150 is that AT-HLC130 does not have gain down VR on the board, but AT-HLC150 has one VR;
WS003770.JPG


I believe the simpler AT-HLC130 without VR (above photo) should be better than AT-HLC150 for my present usage for only VU meter monitoring since the ATV205EXT VU meter board has wide-range (can be adjusted to 30mV to 10V) VRs on the board and also nice VRs (+/- 10dB or more) on the back panel.

Of course, I could fully evaluate and validate the specifications of AT-HLC130; it is really flat High-to-Low converter in 20 Hz - 40 kHz +/-1dB precision, and just fine for VU meter monitoring of the signals in my setup.

4. VU-07(L) and VU-08(R); 600 Hz - 6 kHz driving SQs by ACCUPHASE E-460
ACCUPHASE E-460 is also capable of bi-wire drive for SPs with SP high-level out A and B. Using another AT-HLC130 High-to-Low converter, I can do the same as I described above for VU-05(L) and VU-06(R).

5. VU-09(L) and VU-10(R); 6 kHz - 25 kHz driving TWs by SONY TA-A1ES
SONY TA-A1ES is a simple quasi Class-A amplifier with one SP high-level out. By branching at the SP out terminals to feed into another AT-HLC130 High-to-Low converter, I can do the same as I described above for VU-05(L) and VU-06(R).

6. VU-11(L) and VU-12(R); ca. 8.8 kHz - 6 kHz driving STs by YAMAHA A-S301
YAMAHA A-S301 is also capable of bi-wire drive for SPs with SP high-level out A and B. Using another AT-HLC130 High-to-Low converter, I can do the same as I described above for VU-05(L) and VU-06(R).


Having signals into 12 VU meters with definitely no deterioration to the present excellent total sound quality, the next critically important issue would be "How can I, how should I, adjust gains and relative gains for the 12 VU meters?"

My basic stance and requirements in this regard are as follows;

- I have little interest on "absolute" VU values. The relative VU value over the 12 VU meters would be my main interest in VU monitoring.

- The relative gains among the 12 VU meters should match the air sound pressure (gains) at my listening position.

- At the moment of the maximum (the loudest) actual room air sound pressure/volume (of course still below the system's clipping level) of full huge orchestra fff (sfz) tutti as well as of huge pipe organ's fff (sfz) tutti in my listening room environments, the VU-01(R) and VU-02(L), as well as VU-05(L) and VU-06(R) shall/should swing-up to around +2.8 dB red zone position.

- I shall/should not use the "through output" line level signals from the VU amp boards for further recording or any other listening purposes.

I can rather easily adjust the gain and relative gains of the 12 VU meters accordingly since I have already intensively measured the Fq responses of my multichannel multi-amplifier setup, and I have precise Fq response correlations for line-level signals vs. room air sound, and SP-out high level signals vs. room air sound, as follows;

- Frequency (Fq) responses in the completed system measured by using “cumulative white noise averaging method” under the present standard crossover configurations and relative gains_Part-1_Fq Responses in EKIO’s digital output level: #394

- Frequency (Fq) responses in the completed system measured by using “cumulative white noise averaging method” under the present standard crossover configurations and relative gains_Part-2_Fq Responses in DAC8PRO’s analog output level: #396

- Frequency (Fq) responses in the completed system measured by using “cumulative white noise averaging method” under the present standard crossover configurations and relative gains_Part-3_Fq Responses in amplifiers’ SP output level before protection capacitors: #401

- Frequency (Fq) responses in the completed system measured by using “cumulative white noise averaging method” under the present standard crossover configurations and relative gains_Part-4_Fq Responses in amplifiers’ SP output level after protection capacitors: #402

- Frequency (Fq) responses in the completed system measured by using “cumulative white noise averaging method” under the present standard crossover configurations and relative gains_Part-5_Fq Responses in actual SP room sound at listening position using one measurement microphone: #403

- Frequency (Fq) responses in the completed system measured by using “cumulative white noise averaging method” under the present standard crossover configurations and relative gains_Part-6_Summary, discussions, and a little step forward: #404, #405-#409

- Overhaul maintenance of super-tweeter FOSTEX T925A and further signal fine tuning thereafter: #485

- Latest best tuned total frequency response of the whole system as of January 2, 2022: #485

Even though I will not go into details here regarding my fine adjustments of gains and relative gains for the 12 VU meters, looking at the two diagrams below, you may easily understand my adjustment procedures using flat white noise and pink noise;
WS003769.JPG


WS003768.JPG


Now, I believe that it would be better to draw the system diagrams with 12 VU meters in it.

The present physical system setup is like this;
WS003767.JPG


The 12 VU meters can be installed in the setup like this;
WS003766.JPG


The new EKIO's XO configuration is like this;
WS003765.JPG

and,
WS003764.JPG


After having all of the above favorable preparatory work and planning, I placed purchase order for additional 5 kits each consists of R-65 VU meter (high-grade orange 2700K LED) and VU amp board ATV250EXT.

Although here I will not go into the details, the company SUZUDES10 was very much collaborative on my plan for 12-VU-Meter Array: they carefully selected 12 R-65 VU meters (including the two purchased in advance), installed the high-grade 2700K LEDs, carefully adjusted the light color and intensity over 12 VU meters, and also actually tested and burned-in all the VU meters and the VU amp boards for about non-stop 96 hours confirming LED emission color, intensity, uniformity and functionality.

I have been much impressed by such collaborative work and user oriented attitude given through a small VU manufacturer's craftsmanship in Japan. I found with my great pleasure that Japanese audiophile oriented craftsmanship still survives even in the really niche market of physical high-grade VU meters.


Then, I could move on to physical DIY design of my 12-VU-Meter Array as follows (red scale in mm);
WS003762.JPG

and,
WS003761.JPG


I decided to build with White Wood Veneer of 5.5, 13.0 and 16.0 mm thickness so that the whole 12-VU-Meter Array is light enough to be easily moved to various positions in my audio listening room (actual photos will be shared afterwards). The outer surfaces are to be DIY painted.

On weekend days in two months, I enjoyed my DIY of 12-VU-Meter Array.
WS003760.JPG


WS003759.JPG


Four of AT-HLC130 High-to-Low Converter with wiring completed;
WS003758.JPG


The gains and relative gains for the 12 VU-meters were carefully adjusted by the wide-range VRs on the ATV205EX VU amp board and by the fine tuning (+/- 10 dB) VRs on the back panel of ATV205EX;
WS003783.JPG


The total power consumption of the 12-VU-Meter Array is AC 12V 0.210A, only 2.52W.

I slightly modified the layout of the amplifiers and DAC8PRO on the large wide side-cabinet for nice usual/standard positioning of the completed 12-VU-Meter Array. This layout change was critical in getting approval and consent from our chief interior coordinator (my wife!);
WS003817.JPG


I really like the large glass-faces of the R-65 VU-meters and the high-grade 2700K LED illumination.

As described before, with the 7-m-long RCA cables as well as the AC 12V PS cable, I can easily move the 12-VU-Meter Array depending on the nature of my (our) listening and/or tuning sessions like the several photos below;
WS003744.JPG


WS003745.JPG


WS003741.JPG


WS003742.JPG


WS003743.JPG


WS003739.JPG


WS003738.JPG



Let me emphasize again that the function and specification, including the swing-up and swing-down transient behavior, of the large glass-face R-65 VU meter with high-grade 2700K LEDs and ATV250EX VU amp board are compatible with the IEC 60268-17 VU meter Standard/Specification (3,600 Ohm, AC1.228 V0 - VU 600 Ohm), and I am very pleased seeing the movements of the VU indicator needles very well match with my listening sensations to room air sound at my listening position (so that I carefully adjusted).

During the coming several weeks, I will continue fine tuning of the 12-VU-Meter Array while playing/listening all the tracks of Sony Super Audio Check CD and my latest audio sampler playlist; if needed for this, I will again measure the room air sound by using measurement microphone ECM8000.
 
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Neddy

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Oh....My....Goodness.
Amazing.
Well done!!!
 
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dualazmak

dualazmak

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dualazmak

dualazmak

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The latest total system configuration with DIY 12-VU-Meter Array, the best tuned time alignment and Fq response, together with further various setup info: as of May 30, 2022

Abbreviations
in this post;
SW: sub-woofer, Yamaha YST-SW1000 (L & R)
WO: woofer, Yamaha NS-1000’s 30 cm JA-3058
SQ: squawker, Yamaha NS-1000’s 8.8 cm Beryllium dome JA-0801
TW: tweeter, Yamaha NS-1000’s 3 cm Beryllium dome JA-0513
ST: super-tweeter, Fostex metal horn T925A

Hello friends,

Although somewhat overlapping with the contents of my above post #535, I assume it would be worthwhile, not only for your reference but also as memorandum for myself, summarizing the latest system configuration as of May 30 2022 and the various setup info in my multichannel multi-driver multi-way multi-amplifier stereo project.

Total physical system diagram as of May 30, 2022:
WS003815.JPG


The configuration of 12 VU meters by the newly installed DIY 12-VU-Meter Array:
WS003814.JPG


Please refer to my post #535 for details of the DIY 12-VU-Meter Array.

EKIO's I/O routing settings with 0.1 msec precision group delay configuration:
WS003813.JPG

As for the 0.1 msec precision perfect time alignment between all of the SP drivers, please refer to my posts #493, #494, #504, #507 and #520.

Details of EKIO's XO configuration:
WS003812.JPG

As for the -48 dB high-cut (low-pass) LR filters at 25 kHz for cutting-off possible UHF (ultra-high Fq) noises, please refer to my posts #362-#386 and #518.

Block diagram of master volume control, relative gains, VU meters, signal paths, etc.
WS003811.JPG

Please refer to posts #248 and #251 for the fine tuning 22 Ohm resistors in SQ, TW and ST speaker level signals.

My DAC8PRO's firmware is ver.1.32, and the DIYINHK ASIO USB driver is ver.4.59.0 provided by OKTO when I got DAC8PRO in May 2020; working perfectly fine on Windows 11 Pro.

Physical layout of PCs, DACs, amplifiers, 12-VU-Meter Array, etc.
WS003810.JPG


Actual photo of the usual/standard physical layout:
WS003809.JPG


Many people expressed their interest in seeing the backside wiring photo of my setup, so this is "that" as of May 30:
WS003808.JPG

I use the nice banana plugs, Audio-Technica AT6301 (please refer to my post here and here), only for VU-meter monitoring of SP high-level signals of A-S3000, E-460, TA-A1ES and A-S301.

Best tuned Fq response at my listening position and the Fq coverage by the 12 VU meters:
WS003807.JPG

As for the pros of the "cumulative white noise averaging method" for Fq response measurement, please refer to my post #404.

View of SPs from my listening position with the 12-VU-Meter Array placed in front of me:
WS003806.JPG

For the details of my listening room environment, you would please refer to the latter half of my post #311.

The "even now still amazingly wonderful" SPs:
WS003805.JPG

Please refer to my post here and here for the cabinet and drivers of YAMAHA NS-1000 (not NS-1000M). Also here is spec info for SW YST-SW1000. Please refer to my post here for ST T925A.

Again, as for the 0.1 msec precision perfect time alignment between all of the SP drivers, please refer to my posts #493, #494, #504, #507 and #520.

You would please find transient characteristics objective measurement data of 30 cm WO JA-3058 and SW YST-SW1000 in posts #495, #503 and #507.

As for the unique positioning of ST T925A, please refer to my post #27.

The cables/wirings for SPs:
WS003804.JPG

Please refer to my posts #028, #137 for the SP cables and the Y-lugs/R-lugs.

Also please refer to my post #250, #013(remote thread) for the elimination of magnetic susceptible metals in SP signal handling.

I have been rather frequently inquired about the "all in ASIO I/O configuration" within PC for software XO-EQ "EKIO"; let me share here again, therefore, the following four diagrams:
WS003858.JPG


WS003802.JPG


WS003801.JPG


WS003800.JPG


Several of my ASR friends also have inquired me about the completely silent audio dedicated PCs I use in this project. This is the updated info for my post #225; now both of the PCs run on Windows 11 Pro:
WS003799.JPG

As for the upgrade installation of Windows 11 avoiding TPM CPU RAM SecureBoot restrictions, please refer to my post here.

Regarding my policy and operation of "How to organize digital music library", please refer to my post here on remote thread.

You would also please find here my latest "Audio sampler playlist".

And, please find here (on this thread) and here (remote independent thread post) the Hyperlink Index of this project as well as some of my related posts on remote threads.
 
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