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

GXAlan

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Thank you, I really do hope so!!;)
But if I will apply such DIY on NS-5000, the product warranty should completely sease/lost, I believe.:facepalm:

As you may also aware of, recently I intensively auditioned NS-5000 at Yamaha's audio dedicated cottage near Hamamatsu City, Yamaha's HQ place.
I wonder if you can order “spare parts” :)
 
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dualazmak

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I wonder if you can order “spare parts” :)

Yes, we could order "spare parts", including each of the SP drivers, for NS-1000, NS-1000M, NS-1000x, NS-2000 while Yamaha was continuing production and sales. I do not know, however, their present policy and practice for NS-5000, especially the possibility for poeple who did not buy the completed SP units.

Even if I could order SP drivers of NS-5000, they should be quite expensive, I believe, and I also need to build my DIY heavy-rigid cabinet and the their very unique backchamber units for midrange and tweeter which would be almost impossible...

So, the story is still only remain as my dream project.:D
 

TheBatsEar

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Yes, we could order "spare parts", including each of the SP drivers, for NS-1000, NS-1000M, NS-1000x, NS-2000 while Yamaha was continuing production and sales. I do not know, however, their present policy and practice for NS-5000, especially the possibility for poeple who did not buy the completed SP units.

Even if I could order SP drivers of NS-5000, they should be quite expensive, I believe, and I also need to build my DIY heavy-rigid cabinet and the their very unique backchamber units for midrange and tweeter which would be almost impossible...

So, the story is still only remain as my dream project.:D
The way i see it, the mid dome is supposed to be the star, everything else can be done with other drivers. It's like ATC speakers, you don't buy them for their tweeter or woofer either.

The tweeter chamber should be easy enough, just make sure the backwave isn't reflected into the tweeter again, a large, dampened back chamber should do the trick. The Helmholtz resonator to suck up a enclosure resonance is nice, but i doubt it makes that much of a difference.

I'll try to call someone in Germany to get a quote on the mid dome. Obviously i don't have much hope. ;)
 
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dualazmak

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Can I (we) temporarily synchronize outputs of multiple DAC units (each of them has own independent ASIO driver) in 10 micro second (0.01 msec) precision in DSP-based multichannel audio setup? Part-2: Simplified experiments without using audio mixer

Important top message:

This post is not intending to suggest/recommend you to apply or utilize the same setup and procedures in your DSP-based audio system, but I just would like to share my curiosity and experiments relating to the above titled subject for your possible reference and interest.

Hello @mdsimon2 and dear ASR friends,


Introduction:

This post is a follow-up of my previous post #783 on the titled subject.

As all of you can find in the posts after #783, we had really nice and worthwhile discussion, especially with @mdsimon2 who kindly shared intensively his experiences and data on his audio rig consists of Mac computer and MOTU (professional) multichannel DAC together with MiniDSP.

Since I use Windows PC and I have never used Mac, I believe his intensive discussions would be highly worthwhile for Mac people who would possibly use multiple DAC units in DSP-based multichannel audio system.

Furthermore, in his post #788, he also kindly suggested me additional simplified experiments to objectively observe/measure the relative time-domain discrepancy between two DAC units in my setup without using analog audio mixer which I was including in the recording chain throughout my experiments in my post #783.

Even though I do believe the analog line-level mixer, EDIROL M-10E, has absolutely no effect on time-domain sequence of output signals from multiple DAC units, his suggestions are acceptable for my consideration since I too always like as-simple-as-possible experimental setup to extract what I would like observe and measure.

During this weekend Saturday, I fortunately could spare my time for such additional experiments, as follows.
(Time now in Japan is already 20:08 Sunday evening.:))


General setups for the following Day-4 experiments

Except for otherwise specified in the descriptions on the following Day-4 experiments, all the setup have been strictly the same for those I applied in Day-1 experiment in my post #783.

The recording of the analog output sound from the DAC units was performed again in these procedures;

1. Start recording by Adobe Audition 3.0.1 on the second PC,
2. About 5 sec later, push/start "PLAYING" the DSP EKIO,
3. About 3 sec later after 2., start "PLAY" the test track by JRiver MC.

Consequently, the total length of the recorded sound is usually somewhat longer than the test track itself, but there is no problem at all since we are only focusing on "relative" time-domain discrepancy of two DAC units.

Again, for simplicity of my writing, please let me abbreviate as follows in this entire post;

OKTO as OKTO RESEARCH DAC8PRO,
KORG as KORG DS-DAC-10,
OPPO as OPPO SONICA DAC,
ONKYO as ONKYO DAC-1000(S).

Same as in my experiments in #783, I intentionally connect the DAC units (USB 2.0 compatible) to USB 3.0 ports (ASMedia SUB 3.0 eXtensible Host Controller - 0.96 [Microsoft]), and the USB tree configuration was confirmed by "USB Device Tree Viewer, UsbTreeView (x64)" (now v.3.8.8.0).

No other USB device was connected to USB 3.0 port; I connected my keyboard and mouse to USB 2.0 ports.

Of course, all the PC hardware/software settings/conditions remained unchanged throughout the present 4-day experimental settings.

Hereinafter uSec is "micro second", 0.001 msec.


Day-4 Experiment: Section 1/2

In his post #788, @mdsimon2 first kindly suggested very clever way of measuring relative time-domain discrepancy between the two DAC units without using analog audio mixer, in this way in below Fig.19;
WS00006266.JPG


I believe the above Fig.19 would be self-explanatory enough for your understanding.

First, I prepared one 2 min 53 sec stereo 44.1 kHz 16 bit track, where the L-channel plays steady gain well-QC-ed 20 Hz to 20 kHz precision sine tone Fq sweep (which I copy-pasted from Sony Super Audio Check CD, ref. #651), and the R-channel is complete silence all the way through during L-channel plays the Fq tone sweep. And, very importantly, I inserted the 10 kHz width 1 msec timing markers at exactly the same time position on the top and end in L-channel and R-channel.

In the upper diagram in Fig.19, the L-channel will be fed into CH7 of OKTO, the R-channel will be fed into R-Ch of KORG, then CH7 analog output of OKTO and R-analog-out of KORG, respectively, will be fed into L and R analog stereo line-input of the audio interface TASCAM US-1x2HR for sound recording by ADOBE Audition 3.0.1 in the second independent PC.

In this setup, OKTO will have workload of sine sweep, but KORG will play only silence, while the both have exactly the same timing makers on the same time position for precise measurements of relative time-domain discrepancy.

On the other hand, in lower diagram in Fig.19, I will easily "cross" the digital outputs of DSP EKIO into two DAC Units by crossing the selection of output destination of EKIO's output panels, so that OKTO will play only silence, and KORG will have workload of sine sweep.

OK, now let's see the results, as follows.

I first performed the Fig.19's upper diagram with no group delay setting at all in EKIO's two output channels;
WS00006265.JPG


Here we can clearly observe that KORG was the slow starter, and OKTO started 2.28 msec prior to KORG, and the 2.28 msec startup discrepancy was kept unchanged (no drift at all) throughout the playback of the track.
I repeated the above recording ten (10) times during my 10-hour Day-4 experiment, and confirmed the result remained unchanged.

The above finding encouraged me to give 2.28 msec group delay in EKIO's output channel for OKTO expecting the "outputs" of the two DAC units could be fully synchronized;
WS00006264.JPG


Yes, the outputs of the two DAC units were fully synchronized by the 2.28 msec group delay for OKTO; I repeated the above synchronized recording ten (10) times during my 10-hour Day-4 experiment, and confirmed the result remained unchanged.

Then, I proceeded into the "crossed" input setting, as shown in this Fig. 22;
WS00006263.JPG


As clearly seen in the above Fig.22, again KORG was the slow starter, and OKTO started 2.28 msec prior to KORG, and the 2.28 msec startup discrepancy was kept unchanged (no drift at all) throughout the playback of the track.

When I gave 2.28 msec group delay to EKIO's output for OKTO, the two outputs from the DAC units were fully synchronized as shown in this Fig.23;
WS00006262.JPG


I repeated the above synchronized recording ten (10) times during my 10-hour Day-4 experiment, and confirmed the result remained unchanged.


Day-4 Experiment: Section 2/2

In his post #788, @mdsimon2 also kindly suggested to give "steady gain high Fq constant tone" in one of the stereo channels, in replacement for the sine sweep, as shown in this Fig.24;
WS00006261.JPG


To prepare the 33 sec test track in above Fig.24, I copy-pasted the well QC-ed 16 kHz constant 20 sec tone signal from "Sony Super Audio Check CD", ref. #651.

I first performed the Fig.24's upper diagram with no group delay setting at all in EKIO's two output channels, and the result is shown in this Fig.25;
WS00006260.JPG


Again, we can clearly observe that KORG was the slow starter, and OKTO started 2.28 msec prior to KORG, and the 2.28 msec startup discrepancy was kept unchanged (no drift at all) throughout the playback of the track.

When I gave 2.28 msec group delay to EKIO's output for OKTO, the two outputs from the DAC units were fully synchronized as shown in this Fig.26;
WS00006259.JPG


I repeated the above synchronized recording ten (10) times during my 10-hour Day-4 experiment, and confirmed the result remained unchanged.

Then, I proceeded into the "crossed" input setting, as shown in this Fig. 27;
WS00006269.JPG


As clearly seen in the above Fig.27, again KORG was the slow starter, and OKTO started 2.28 msec prior to KORG, and the 2.28 msec startup discrepancy was kept unchanged (no drift at all) throughout the playback of the track.

When I gave 2.28 msec group delay to EKIO's output for OKTO, the two outputs from the DAC units were fully synchronized as shown in this Fig.28;
WS00006270.JPG


I repeated the above synchronized recording ten (10) times during my 10-hour Day-4 experiment, and confirmed the result remained unchanged.


Discussion and conclusion

The general discussion and conclusion are almost the same as I wrote in my post #783 based on my Day-1 through Day-3 experiments thereof.

The time-domain discrepancy between the two DAC units can be attributed to the "startup timing lag of DAC processing" between the two units, and the "relative discrepancy" remained unchanged (did not drifted) during the whole music track as well as during the specific audio sessions.

Consequently, I could/can temporarily synchronize (10 uSec accuracy/precision) the "outputs" of the multiple DAC units by compensating such "startup timing lag(s)" by group delay setting(s) in upstream DSP EKIO only if the "time-domain discrepancy" would remain unchanged during the specific audio sessions.

You need to note, however, that the above conclusion in my setup would possibly not always true for "your" DSP-based multichannel audio setup on Windows PC. In case if you would test/try the similar multiple DAC-unit setup, you need to objectively measure/confirm precisely the "relative time-domain discrepancy" before and after your specific audio session. (As you may agree, I essentially have no intention of "routinely" apply this approach in my multichannel audio rig.)

In my experimental setups, the connection of multiple DAC units to PC motherboard's USB 3.0 ports with no other USB device connected to USB 3.0 would possibly be contributing to the excellent reproducibility of the "relative startup lags", I assume.

The "relative startup timing lags" in my experimental setup can be summarized in this Fig.29;
WS00006268.JPG


As you can see in above Fig.29, the "relative startup lags" remained almost unchanged at least during the 4-day experimental sessions in my audio setup; they would possibly vary, however, depending on USB ASIO settings and/or on any of other unknown factor(s) of USB-ASIO time-domain sequences.

Focusing on the specific "startup timing lag" between OKTO and KORG in my setups, during my Day-1 through Day-3 experiments in #783, I could read/measure (by measuring on expanded x-axis time scale of Adobe Audition 3.0.1) it was 2.27 msec, while in my present Day-4 experiments on this post, it was read to be 2.28 msec.

You would please note there is essentially no significant difference at all between these 2.27 msec (2,270 uSec) and 2.28 msec (2,280 uSec), since the absolute time resolution is 22.7 uSec in the test track of 44.1 kHz PCM; the observed/read 10 uSec difference (2.27 msec vs. 2.28 msec) is less enough than the absolute minimum time-domain digital granularity of 22.7 uSec.


To finish this rather long post, let me repeat the top message I have written;
This post is not intending to suggest/recommend you to apply or utilize the same setup and procedures in your DSP-based audio system, but I just would like to share my curiosity and experiments relating to the above titled subject for your possible reference and interest.

I would highly appreciate if I could hear your further thoughts and discussion(s).
 
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dualazmak

dualazmak

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Hello again @mdsimon2 and dear ASR friends,

I believe that now I should end-up my experiments and discussion on the very much niche and limited "temporarily compromising synchronization method" for multiple ASIO DAC units in our Windows-PC + DSP-based multichannel audio setups.
(please PM me, if we need further discussion on this topic.)

If I (we) would like to seriously implement full synchronization of multiple DAC (or audio interface) units through ASIO routing, we need to utilize (pro-grade) audio interfaces (or DAC units, if any) which actually do support such usage, for example as I could find this clearcut FAQ page for RME Fireface UFX series; I hope your web browser would properly translate the page into English.
 
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phofman

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@dualazmak: While I just skimmed your very informative posts (but too long for me to read in detail, sorry)...

You have two USB audio devices. One is async (Okto using XMOS), the other I do not know (I could not find any relevant info for the Korg).

Each of the devices consumes samples at its own rate (Okto clocked by its crystal, Korg either by its crystal or by your PC USB controller clock). That means each ASIO driver for each device will call its bufferSwitch callback independently, fetching data independently from the playback software.

Now what does your DSP layer do with the asynchronous callbacks from your two asynchronously-running devices? It must somehow align them/make them synchronous so that the chain upstream from the two devices merging point runs synchronously. How does it actually do? Typically two options - asynchronously resampling for one of the devices, using one as master, the other as resampled slave (e.g. linux pulseaudio or jackd), or simply dropping/duplicating samples (e.g. gstreamer). Or it can simply hit buffer over/underflow and restart the devices, producing a very audible click. Of course resampling one stream introduces a delay for which the master stream should be delayed too, ideally...

Since the two USB devices by principle cannot be started at exactly the same moment, especially if controlled via two separate unsynchronized drivers, there will always be some time delay between the two analog outputs. And analog outputs of these devices will run more or less synchronously because the upstream chain must somehow synchronize the two streams. The master device will run clean unaffected stream, the "slaved" one will have its stream tweaked in a nicer or uglier way to make up for its differently-paced rate of consuming samples. Or both streams will be corrupted regularly if the DSP "handles" the synchronization by simply restarting the two device streams.
 
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dualazmak

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Hello @phofman, thank you indeed for your kind attention on my post #804 and hopefully on my post #783.

Since the two USB devices by principle cannot be started at exactly the same moment, especially if controlled via two separate unsynchronized drivers, there will always be some time delay between the two analog outputs.

Yes, I fully agree with you. This is exactly "the known issue" of my interest and curiosity, and therefore I planned to objectively and semi-quantitatively observe/measure what-kind-of and how-much "time-domain discrepancies" would be happening, if I dare to use multiple independent ASIO-driven DAC units in Windows-DSP-based multichannel audio system.

If the "discrepancy" would be totally attributable to the measurable "relative kickup/startup timing lag", and only if such "timing lag" would be kept unchanged (not drifting) during the playback of an audio track (or during the specific audio session), such "timing lag" may hopefully be compensated by using "group delay (0.01 msec precision)" capability of upstream/digital-domain DSP software, so that the "outputs" of the multiple ASIO DAC units could be synchronized.

This consideration/expectation has been my main motivation in my posts #804 and #783.

At least in my strictly fixed experimental conditions (including USB 3.0 connection of multiple DAC units), I could find/confirm that the "discrepancy" seemed to be totally attributable to the "reproducible kickup/startup timing lag", and I could temporarily/compromisingly compensate the "timing lag" by "group delay" settings in upstream DSP software, so that the analog "outputs" from the DAC units could be synchronized.

That (including the experimental methodology) is "the all" I would like to share at least in my experimental conditions, and I know well such findings cannot/should-not be generalized for other people's Windows-DSP-ASIO multichannel audio setup.

I hope above my basic stance (in my posts #804 and #783) on the present topic would be understandable for you and other people kindly visiting here.
 
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pma

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@dualazmak san, Thank you for introducing EKIO xover software. It is very easy to use. I am now working on a project of a small 2-way active speaker and will use the EKIO to make a software crossover.

The xover would be a 48db/octave LR. (Edit: of course baffle step compensation, gain equalization, slope for tweeter etc. will be added, based on REW measurements of individual drivers mounted in the cabinet)

SW_xover_FR.png


SW_xover_time.png
 
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pma

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BEHRINGER ECM8000 in my case, has reasonably flat response over 15 Hz to 20 kHz;
Unfortunately, it has not. This microphone is reasonably flat up to about 5kHz, but above 5kHz you definitely need to use a calibration curve, and this calibration will be different for almost every produced piece. I know very well what I am speaking about. Without a correction curve, you will get several dB peaking at high frequencies.
 

thewas

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Unfortunately, it has not. This microphone is reasonably flat up to about 5kHz, but above 5kHz you definitely need to use a calibration curve, and this calibration will be different for almost every produced piece. I know very well what I am speaking about. Without a correction curve, you will get several dB peaking at high frequencies.
Correct:
Stats2_ECM8000.png

Source: https://www.hifi-selbstbau.de/index...ierung-muss-das-sein-anfer&catid=36&Itemid=66

MicCal_ECM8000a125.png

Source: https://www.hifi-selbstbau.de/index...icht&catid=36:software--messtechnik&Itemid=66
 
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dualazmak

dualazmak

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Thank you for introducing EKIO xover software. It is very easy to use. I am now working on a project of a small 2-way active speaker and will use the EKIO to make a software crossover.

Yes, nice to hear so.

Our discussion on EKIO on this thread entitled "A comparison of convolution engines" would be also of your reference, I assume.

Furthermore, EKIO is stable, very light to CPU, accurate (including 0.01 msec precision group delay settings), reproducible (of course), and if you like you can change almost all the parameters "on-the-fly" while listening to the music.

Guillaume of LUPISOFT kindly responded to me informing (ref. here);
"EKIO uses IIR filters. The processing is done using a cascade of second order transposed direct form II biquad sections. Every calculation is done using 64 bit floating point numbers."

Please take care, however, that you should not change the gains by numeric keyboard input on-the-fly, but I strongly recommend you to use mouse wheel to get up-and-down of the gains; this is a precaution to avoid any "mis-typing" which may cause too loud harmful sound to your SP drivers. I once requested this "mouse-wheel gain operation" to Guillaume of LUPISOFT, and he very quickly incorporated my request in EKIO.

Just for your reference, my latest EKIO configuration (ref. here);
WS00006943.JPG
 
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dualazmak

dualazmak

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Unfortunately, it has not. This microphone is reasonably flat up to about 5kHz, but above 5kHz you definitely need to use a calibration curve, and this calibration will be different for almost every produced piece. I know very well what I am speaking about. Without a correction curve, you will get several dB peaking at high frequencies.

Thank you for your kind reminders on ECM8000.

I bought my ECM8000 (Made in Germany) on December 26 in 2008 at a Pro Audio Shop in Akihabara Electric Town in Tokyo.
I have a tiny "story" for my purchasing of ECM8000 maybe worthwhile sharing with you two.

At that time, I have intensively consulted with one of my good friends who was/is a skilled recording engineer working for one of the very well established recording companies in Japan asking for recommendation(s) for affordable (for me, as an amateaur audiophile) measurement microphone.

In very happy coincidence, at that time his company too was looking for general-purpose (not in serious situation) affordable measurement microphones to be used just for preparatory checking the acoustics of their recording venues. They signed a special contract with BEHRINGER Japan KK for purchasing of forty (40) units of ECM8000 (Made in Germany) on the condition of strict selection of the units fulfill the measured spec of within plus/minus 1.3 dB from the flat response throughout 20 Hz to 20 kHz; the measurements for the unit selection should be done in a contracted audio measurement company in Japan (I myself did not, do not, know what would be the precise calibration/measurement methods they have used.)

Then, BEHRINGER Japan KK selected about sixty (60) units of ECM8000 (Made in Germany) fulfilling the "special condition" and delivered the contracted 40 units to my friend's recording company with the individual actually measured data sheet. BEHRINGER Japan "voluntarily" kept the selected 20 units for backup purpose for my friend's recording company. As you can guess, my friend very kindly arranged that I could purchase one "so carefully selected" ECM8000 at a "BEHRINGER designated Pro Audio Shop" in Akihabara Tokyo.

Unfortunately, I could not get the specific "measured data sheet" for my selected ECM8000 due to the "Non-Disclosure Agreement, NDA" between BEHRINGER Japan KK and my friend's recording company, but my friend kindly checked and confirmed that I actually got the "selected" ECM8000 (Made in Germany, so printed on the box and warranty sheet) by referring the serial number of my unit.

That was/is the tiny story happened in late 2008, and I have been using "such" ECM8000 for 15 years since then.

I have never asked overhaul maintenance and calibration of my "such" ECM8000; after having your above kind reminders, therefore, I am now considering possible maintenance and calibration of my ECM8000 hopefully by a reliable company/laboratory in Japan. I may hopefully ask again my "that" friend for such possibility and recommendation, even though he has already retired from the recording company.
 

sarieri

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Anyone thinking about building am analog volume control for their multichannel DAC (use MUSES72323)? Indeed there are many ways to control the volume digitally, but in theory analog volume control still wins in terms of THD+N when the volume is low right?
 
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dualazmak

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Anyone thinking about building am analog volume control for their multichannel DAC (use MUSES72323)? Indeed there are many ways to control the volume digitally, but in theory analog volume control still wins in terms of THD+N when the volume is low right?

Although there are/were so many discussions on your point, "theoretically" I essentially agree with you.

I use the most upstream music player software JRiver MC for master volume controller, and usually keep the input/output gains in DSP EKIO and OKTO DAC8PRO in rather high gain, and I perform fine relative gain tuning (or tone control) by using precision and durable volume/gain controllers of my multiple HiFi "integrated amplifiers" as shown in this diagram (for my latest system setup, please refer to my post here);
WS00006942.JPG


As for the pros of using multiple HiFi "integrated" amplifiers, my post here would be your interest and reference, I assume.

This post on "startup/ignition sequences" and "shutdown sequences" would be also of your reference.
 
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dualazmak

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This is a follow-up of my above post #813 on ECM8000 microphone.

I could get information that Cross-Spectrum Acoustics used to deliver measured/calibrated ECM8000 with calibration data attached, but they gave-up such service since the quality of recent model/version of ECM8000 (not made in Germany, I assume) has badly deteriorated to the point of that they can no longer justify the effort in dealing with as they declared here.:facepalm:

Time really flies, and the product quality, even still having the same brand and product name, changed much worse; I am very sorry that the same thing has been frequently happening in our audio market/industry...

And, this is one of my concerns/reasons in sticking to reliable and durable domestic brand HiFi integrated amplifiers (Accuphase, Yamaha, old Sony) also in terms of maintenance/repair service availabilities.

Maybe, it would be the time for me to seriously consider purchasing a well calibrated Earthwork Audio's measurement microphone, rather expensive though. ___(Before that, of course I would like to "see and know" the present Fq response data of my strictly selected 15-year old ECM8000 made in Germany.;))
 
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thewas

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dualazmak

dualazmak

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You could even send it for small money to its past home country to be measured https://shop.hifi-selbstbau.de/produkt/mikrofonkalibrierung/
Its the same company that have measured already thousands of mics https://www.hifi-selbstbau.de/index...k/1000-mikrofonkalibrierungen-eine-uebersicht

Thanks a lot for the info!
I will firstly look for Japan domestic services (I have not yet talked to "that" friend) , and if we have none, then I will send mine to its home country. ;)
 
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dualazmak

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Hello again @pma, @thewas, and friends,

Regarding possible check (and calibration, if needed) for my ECM8000 (Made in Germany!), today I could talk to "that" friend, and I found another fortunate coincidence again like we had in late 2008 (ref. my above post #813).

He now has "matched pair of Earthwork M50" which recently perfectly calibrated, and he is very happy to objectively precisely compare my ECM8000 with his Earthwork M50s at his home using his nice audio setup, so that I would have precise calibration curve for my ECM8000!

Since I do not like physically sending my ECM8000 by transportation company, within a few months I will make round drive-trip to his home about 300 km away from my home.:D
 
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pma

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my ECM8000 (Made in Germany!)
It is not what it was few decades ago ;).

Very sorry for the off topic post, sumimasen, gomen nasai, Dualazmak San. I have quite some experience in Japan, I visited your country many many times since the year 2000, for business reason, with the sophisticated plasma spraying technology. I am proud I have friends in Japan and that I have understood your country a bit more than a random traveller.
The times have changed and “Made in Germany” is not the same thing as it used to be in the past century.
This thread is “very Japanese” and I like it a lot (please do not get me wrong). Keep your good work going on!
 
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