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

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dualazmak

dualazmak

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This thread is “very Japanese” and I like it a lot (please do not get me wrong). Keep your good work going on!

Really thank you for your kind encouragements.

Let's keep communication here on this thread and of course on your exciting new thread!
 

bbfoto

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The latest "startup/ignition sequences" and "shutdown sequences" in my DSP-based multichannel multi-SP-driver multi-amplifier fully active audio rig as of August 3, 2023

Hello friends,

Having the latest system configuration as of August 3 2023 shared in my above post #774, this post is intending to update my post #426.

A few of my dear ASR friends have contacted me asking about standard startup/ignition sequences and shutdown sequences in the multichannel multi-amplifier system together with JRiver MC (or Roon) and software crossover EKIO running on Windows 11 PC.

They were a little bit concerned about the maximum protection of speaker drivers from possible unexpected accidental pop damage, and I fully understand their worries even though I have the protection capacitors for midrange Be-squawkers, Be-tweeters and super-tweeters.

Although I daily perform these very familiar sequences almost unconsciously, I believe it would be worthwhile sharing the sequences with all of you visiting this thread.

As intensively shared in my post #774, the latest multichannel multi-amplifier system configuration can be summarized in these diagrams and photos;
View attachment 304426
(For the details of these diagrams and photos, you would please visit my post #774 where you can find the larger ones.)

Standard "Startup/Ignition Sequences":
01. Wake-up the AC Circuit Breaker (100V/15A, if it has been shutdown)
02. Power-on the root switch of 24-port AC power strip (series of four 6-port strips)
______This automatically wakes up DIY-12-VU-Meter-Array, KORG DS-CAC-10, BEHRINGER DS2800
03. Power-on YAMAHA A-S3000 after making sure the volume is at -infinity dB (no sound)
04. Power-on ACCUPHASE E-460 after making sure the volume is at -infinity dB (no sound)
05. Power-on SONY TA-A1ES after making sure the volume is at -infinity dB (no sound)
06. Power-on YAMAHA A-S301 after making sure the volume is at -infinity dB (no sound)
07. Power-on L&R sub-woofers YAMAHA YST-SW1000 by remote controller (preset volume 14:00 o'clock)
08. Power-on OKTO DAC8PRO; make sure the volume (master gain) is less than -75 dB
09. Power-on Windows 11 Pro PC
10. Check & confirm wake-up and running of ASIO BRIDGE at taskbar
11. Check & confirm wake-up and ready of DIYINHK ASIO at taskbar
12. Check & confirm all of the Window's sound I/Os (KERNEL, WASAPI, WMD, Mic, etc.) are muted-off
13. Launch JRiver MC (or Roon); make sure the master volume is at about 5 % (-77.5 dB)
14. Start playing JRiver the 6th track of my music sampler/reference playlist (piano solo)
15. Slightly volume-up JRiver to around 10 % (-55.0 dB); check the JRiver's small VU-EQ meter is properly dancing
16. Launch software DSP EKIO
17. Open/load the EKIO's standard configuration file into EKIO
18. Start playing EKIO
19. Check and confirm that all of the EKIO's I/O VU meter bars are properly moving in very low gain
20. Volume-up A-S3000 to 11:55 o'clock (-18 dB)
21. Volume-up E-460 to -17.0 dB
22. Volume-up TA-A1ES to -17.0 dB
23. Volume-up A-S301 to 09:45 o'clock (ca. -19 dB)
24. Volume-up DAC8PRO to -4 dB; check to properly hear the sound in very small volume
25. Carefully and slowly volume-up JRiver MC (as master volume) to usual listening volume using mouse wheel


Standard "Shutdown Sequences":
01. Stop playing JRiver MC (or Roon)
02. Volume-down the "Master Volume" of JRiver MC to less than 5% (-77.5 dB)
03. Stop playing software crossover EKIO
04. Volume-down OKTO DAC8PRO to -99 dB
05. Volume-down YAMAHA A-S301 to -infinity dB (no sound)
06. Volume-down SONY TA-A1ES to -infinity dB (no sound)
07. Volume-down ACCUPHASE E-460 to -infinity dB (no sound)
08. Volume-down YAMAHA A-S3000 to -infinity dB (no sound)
09. Power-off A-S301
10. Power-off TA-A1ES
11. Power-off E-460
12. Power-off A-S3000
13. Power-off L and R active sub-woofers YAMAHA YST-SW1000 by remote controller
14. Exit/Shutdown EKIO
15. Exit/Shutdown JRiver MC
16. Shutdown Windows 11 Pro PC
17. Soft-shutdown (Mute-off) DAC8PRO by remote controller
18. Switch-off the root switch of 24-port AC power strip (series of four 6-port strips)
______This automatically shutdowns DIY-12-VU-Meter-Array, KORG DS-CAC-10, BEHRINGER DS2800
19. Down the AC Circuit Breaker (100V/15 A, if needed for long shutdown; longer than a week or so)


Although these may look somewhat complicated at your first glance, I usually complete the startup/ignition) sequences in less than two minutes, and I complete the shutdown sequences in less than one minute.;)

Perhaps you should perform these "startup/ignition" and "shutdown" sequences more slowly in order to let the power grid "catch up" and not cause power sags and surges on the electrical grid. :p ;)

I suspect everyone in your surrounding neighborhood knows exactly when you have switched on and off your stereo system when their lights dramatically dim and then brighten again from the power sag and then surge, and the overhead power lines above the street must glow red and sizzle & crackle from the massive current draw. I also suspect that the regional power station operators and engineers also know when you switch your system on and off. :p ;)

Lots of lighthearted joking and kidding above, but you have put together a Truly AMAZING system...and an incredibly detailed ongoing analysis of the system with your changes/updates!


TBF, Perhaps the Acoustics of the room are somewhat more important as Igor suggested. Do you have an overhead "cloud" absorber/diffusor on your ceiling above your listening position?



Have you ever thought about using a high-end multi-channel car audio system DSP in your home setup? Alpine makes their new Alpine F # 1 STATUS system HDP-H900 DSP with 8-channel 8v Balanced Outputs.




Or their is the Audiotec-Fischer BRAX DSP with 8-in/12-out:

https://www.audiotec-fischer.de/en/brax/processors/dsp

Their are many others as well including the miniDSP C-DSP 8x12 with DIRAC Live (FIR +IIR filters). I use this miniDSP along with one of the Audiotec-Fischer HELIX ULTRA 12-channel DSPs separately in two of my high-end car audio systems. The difference in the spectral balance, image focus, soundstage, and hearing "into the room" ambiance is like Night & Day when the DIRAC processing is used. I also built and tried a PC-based VST DSP system similar to yours, but especially in the automotive environment it was too problematic and cumbersome to implement.

These car audio processors have a 12v (+) REM output that can be used to send a 12v Trigger to turn on other components in your system that have a 12v Trigger input, which may simplify your "startup/shutdown" sequences. The 12v REM output trigger can also be set in the software with a specified "turn-on" and "turn-off" DELAY in order to avoid loudspeaker damage from Turn-On or Turn-Off "Pops" or "Thumps."

Many of these newer car audio DSPs are also incorporating incredibly powerful software and hardware tools for Acoustic Measurements, including licensing some of the Real-Time Phase Analysis tools in Smaart v9, and using 7-mic arrays for spatial averaging measurements or Binaural Head microphone measurement systems. See the Gladen/Mosconi "BARNIE" system and JL Audio TuN software and MAX measurement systems.

^Just some random "food for thought".

I'll follow along as you continue with the evolution of your incredible system.

Carry on.

EDIT: I'll PM your regarding sharing some of our Reference Music Tracks. I am a Saxophonist and Drummer/Percussionist primarily focused on Jazz but I perform and love all types of music.
 
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dualazmak

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Lots of lighthearted joking and kidding above, but you have put together a Truly AMAZING system...and an incredibly detailed ongoing analysis of the system with your changes/updates!
Thank you for your kind attention on my project.

Yes, my gears including the four integrated amps are rather slow starter (each has proper delay relays in it) and rather SDG oriented; my dedicated circuit breaker (100 V, 15 A) for the audio system has never worked for my audio sessions!;)

TBF, Perhaps the Acoustics of the room are somewhat more important as Igor suggested. Do you have an overhead "cloud" absorber/diffusor on your ceiling above your listening position?
As I wrote here #311 and here #687, the ceiling of the listening room is covered by microporous diatom panels carefully selected with proper sound absorption performance. On the wooden floor, we have the large carpet also selected with suitable sound absorption properties.

Have you ever thought about using a high-end multi-channel car audio system DSP in your home setup? Alpine makes their new Alpine F # 1 STATUS system HDP-H900 DSP with 8-channel 8v Balanced Outputs.
I know well the great enthusiasms in car audio world/industry. I have been restricting myself not going into that direction mainly in consideration of WAF (wife acceptance factors.) She is also CFO and CICO (Chief Interior Coordination Officer).:D

As for audio in my car, my beloved TOYOTA CAMRY (hybrid, top grade, leather seats, very quiet, bought in 2020, now very quiet and smooth BRIDGESTONE REGNO tires, photo here), I have factory-installed TOYOTA-JBL car audio system; during our long drive, we enjoy endless play of music using USB-connected 128 GB Apple iPod which has all of my digital music library (about 25,000 tracks, still increasing) in mp3 compression.

We are very much satisfied with the car audio system since car driving environment is always not so quiet as our home listening room; I dare not to seek further high-end in car audio setups...

As for my "organization" of digital music library, please refer here.
 
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dualazmak

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EDIT: I'll PM your regarding sharing some of our Reference Music Tracks. I am a Saxophonist and Drummer/Percussionist primarily focused on Jazz but I perform and love all types of music.

You will be much welcome! Waiting for yout PM contact.
 

GXAlan

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Have you ever thought about using a high-end multi-channel car audio system DSP in your home setup? Alpine makes their new Alpine F # 1 STATUS system HDP-H900 DSP with 8-channel 8v Balanced Outputs.

Someone should get one and send it to Amir! JL Audio entered the home audio market with their car audio subwoofers and it ended up being a flagship product on the first attempt.

The Alpine unit looks very well built:
1695435005766.png


These are 1 GHz Griffin UL.
1695435101226.png

But at least in the USA, we have this warning
1695436132801.png
 

bbfoto

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Someone should get one and send it to Amir! JL Audio entered the home audio market with their car audio subwoofers and it ended up being a flagship product on the first attempt.

The Alpine unit looks very well built:
View attachment 313978

These are 1 GHz Griffin UL.
View attachment 313979
But at least in the USA, we have this warning
View attachment 313987

YES, Unfortunately, the Alpine F # 1 STATUS system is only available as a complete package...DSP, A&K DAP, Amplifiers, 3-way Front Component Speakers and Subwoofer... ~$35k USD if I'm not mistaken. OUCH! :p

But they do have a little brother "STATUS" HDP-D90 (non-F # 1) DSP for around $2,400 from Crutchfield in the U.S., but it also has a built-in Multi-Channel Class D Amplifier...


I purchased one from an Online seller in Germany for about $1800 shipped to the U.S. when the European V.A.T. was removed. :)

The Built-In Class-D amplifier provides:

8 x 50w @ 4-Ohm + 4 x 80w @ 4-ohm

And the Four 80w channels can be Bridged to provide 160w x 2 @ 4-ohms.

Those Bridged Channels could power a DIY Subwoofer driver that has 4-ohm Dual Voice Coils for a total of 320w RMS.

In some instances this would be a VERY NICE Complete "ACTIVE" One Box solution for a custom/DIY multi-way loudspeaker system or @dualazmak Yamaha NS-1000.

You Do Not have to use the Built-In Amplifiers...There are also TEN processed 4v RCA Preamp Outputs you can use to feed the inputs of other standalone amplifiers.

The Inputs are either RCA line level, Speaker-Level (High-Level), or Toslink Optical & RCA Coaxial Digital. The USB port is to connect a PC for the DSP setup and Adjustments with their software. But they have a companion "Front End" Source Player (HDS-990) "Head Unit" that has a USB input for File Playback from USB Memory drives.

https://www.crutchfield.com/p_500HDS990/Alpine-HDS-990.html


You can even Download the Software and try it in "DEMO mode" (most of the others I list below allow this as well)...


Audiotec-Fischer HELIX & BRAX, Gladen/Mosconi Aerospace/PRO DSP, Audison Forza Bit, JL Audio VXi & TwK-88/D8, Zapco HDSP-Z16 V, and others offer many high-end standalone DSPs and DSP/Amplifier units with different I/O configurations...the latter even has 3 different swappable DAC boards. And some of these DSP-Amplifiers use Class-A/B designs for everything but the mono subwoofer channels.

There are also the quite capable Pro-Audio Loudspeaker Management Processors such as the dbx DriveRack VENU360 and DEQX processors...



And Behringer still makes and sells their relatively inexpensive UltraDrive PRO DCX2496...I used this back in the mid-90's in my custom aftermarket stereo system in my Acura Legend coupe. There was a nice bloke at the time who made a Custom 12VDC PSU for this unit and the UltraCurve PRO DEQ2496 that would easily fit inside the original case. :)



Lots of options, though I realize that the possibilities are almost limitless when using PC-based VST/convolution plugins. :)

I'll stop filling up the thread now with all of this off-topic banter. :p
 
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mdsimon2

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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.

As I see it, you simply repeated the tests you had previously done without the analog mixer. Correcting for constant delay between two DACs is relatively trivial, what is not trivial is observing whether they maintain phase coherency.

The tests I described in posts #793, #794 and #796 would help explore the phase coherency and give some insight as to the nature of synchronization. I am most interested in seeing real time scope captures (either via a separate oscilloscope or using an ADC with REW's scope functionality) while playing a high frequency tone (15+ kHz) to both DACs to observe the relationship between each DAC output as shown in post #794. However, you seem unwilling to conduct these tests so probably best to move on.

Michael
 
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dualazmak

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As I see it, you simply repeated the tests you had previously done without the analog mixer. Correcting for constant delay between two DACs is relatively trivial, what is not trivial is observing whether they maintain phase coherency.

Yes, I well understand your point.

Nevertheless, I am now reluctant to perform further experiments along with your suggestions since my experimental usage of multiple DAC units driven by independent USB-ASIO drivers would be quite exceptional, and hence almost no practical merit/benefit (only useful for my niche experimental settings.)

I have already collected enough data clarifying my "personal curiosity", and I could also confirm and validate my experimental methodology to take snapshots of time-domain discrepancies. The possible sharing of further experimental data would not attract interest at all of people visiting here.

Consequently, you would please allow me ending up my experiments and discussion on the issue.
I thank you very much for your kind attention and invaluable suggestions so far kindly given to me.
 
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555

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Hello I am a new member from Japan, please let me participate on this interesting thread!
 
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dualazmak

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Hello I am a new member from Japan, please let me participate on this interesting thread!
You are quite welcome! Very nice to have a new friend from Japan in ASR Forum.:D
 
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dualazmak

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Frequency response of my BEHRINGER ECM8000 measurement microphone (specially selected unit in 2008)

Hello dear ASR friends,

This post is a follow-up of my posts here #813 and here #819.

On October 5 Thursday, I made one-day my-car round trip to the home audio studio of “that my friend (hereinafter in this thread Mr. TY)”, bringing my BEHRINGER ECM8000 measurement microphone (specially selected unit in 2008 fulfilling +/- 1.3 dB from flat level, i.e. within 2.6 dB in total).

I have been using my ECM8000 not so frequently, about once in two months, and I have been always keeping it with utmost care of avoiding physical shocks as well as preventing too much humidity for long hours; I have been keeping it in my active large (W x H x D = 42 x 87 x 40 cm) acrylic desiccator box (moisture-proof case) of less than 20 % humidity, less than 25 degree-C temperature, together with all of my cameras/lenses and precision measurement gears.

As shared in #813 and #819, Mr. TY has been working for a major recording company in Japan for long yeas, and he retired from the company around 2015; he is now enjoying his retired audiophile life having his home audio/recording studio with nice audio setup of excellent studio monitor SPs sub-woofers and ribbon super-tweeters capable of nice reproduction of 15 Hz to 30 kHz. Furthermore, he has a matched pair of EARTHWORK M50 measurement microphone with certified Fq response/calibration data obtained recently in this year.

Our current objective was/is to get Fq response deviation of my ECM8000 from his M50, and apply such deviation data to M50’s “certified” response curve to give approximate Fq response curve for my ECM8000.
WS00006433.JPG


Let me briefly describe about our simple measurement setup as follows.

Firstly, we carefully physically set M50, using tripod, about 1.5 m from L-channel of his SPs; M50 was connected to my audio interface TASCAM US-1x2HR for recording by using ADOBE Audition 3.0.1 on my laptop PC in 96 kHz 32 bit (floating) monaural format.

We used well prepared 96 kHz 24 bit stereo white noise (i.e. 10 Hz to 48 kHz white noise) for our recording (we played the L-channel only for our recording); I (we) have little interest on absolute/relative sensitivity of M50 and ECM8000, and hence the recording gain level was set about -12 dB for our 2-min consecutive recording of the white noise sound. Of course, the volume/gain for playing the white noise was rather low, well below the clipping level, of his excellent audio system so that the effects of “room modes” could be minimized.

Please note that we also recorded the sine sweep track (10 Hz to 48 kHz in about 1 min), but the stability and reproducibility of FFT Fq analysis was better by using cumulative/recorded white noise especially in low Fq zone below 200 Hz (ref. here #392 and here #404); in this post, therefore, I sticked to FFT Fq averaging analysis of “cumulative/recorded 2-min white noise” for relative comparison of M50 and ECM8000. (This kind comment by Dr. Floyd Toole given to me would be also of your reference.)

We did the exactly same recording using my ECM8000 set precisely identical physical geometry which we have set M50 under exactly the same audio settings.

For FFT averaging Fq analysis on the whole of the 2-min white noise recorded data, we use FFT size 65536 for 10 Hz to 500 Hz, and FFT size 512 for 500 Hz to 30 kHz, both with Blackmann-Harris FFT window in ADOBE Audition 3.0.1; here these FFT sizes were selected as the proper and reproducible smoothing factor for resulting Fq response curves.

Since Mr. TY is not comfortable about sharing here the intact Fq response of M50 in his audio setup (including his studio/room acoustic modes), let me share here only the “deviation data” for my ECM8000 vs. his M50, and the application of such deviation data on the M50’s “certified” response curve to give approximate Fq response curve for my ECM8000, as shown in this diagram.
WS00006442.JPG


This diagram shows the obtained Fq response of my ECM8000 in vertical dB scale of +/- 1.0 dB and in +/- 3.0 dB:
WS00006431.JPG


We were much surprised and relieved that my ECM8000 (specially selected in 2008) still has acceptably flat response throughout 10 Hz to 22 kHz within +0.7 dB/-0.5 dB from flat response, i.e. within 1.2 dB in the whole; you would please be reminded again that the special selection of my ECM8000 was done in 2008 fulfilling the criteria of within +1.3 dB/-1.3 dB, i.e. within 2.6 dB in whole at that time.

This diagram represents such measured Fq response of my ECM8000 in identical X-Y scale with the typical Fq response curve printed on the outer product box of ECM8000 which I purchased in 2008;
WS00006430.JPG


It is obvious that I have quite a lucky unit of ECM8000 having unexpectedly flat response even 15-year (or more) after its production.


Those who use BEHRINGER ECM8000, however, should well notice that this kind of acceptably flat response cannot always be expected for your unit, as @pma and @thewas kindly chimed to me in posts #809 and #810.

Furthermore, as I shared in my post #816, the recent products/versions of ECM8000 generally have considerably worse Fq responses; 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 or in the USA, I assume) has badly deteriorated to the point of that they can no longer justify the effort in dealing with as they declared here.

If I dare to plot Fq response curve of my specific ECM8000 on the distribution data of 125 units of ECM8000 (ref. here), it can be seen as in this diagram;
WS00006429.JPG


In any way, I am relieved (and a little bit surprised) that I could confirm my ECM8000 specially selected in 2008 still has very nice flat response within total 1.2 dB gain span over 15 Hz to 22 kHz now after 15-year later, I know well this is just quite rare and lucky case for myself.

Yes, I should continue keeping my own treasure ECM8000 very carefully in my active acrylic desiccator box (moisture-proof case), less than 20 % humidity, less than 25 degree-C temperature.
 
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TNT

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is less enough than the absolute minimum time-domain digital granularity of 22.7 uSec.
One should however observe that the sample time in a PCM flow has noting to do with the time resolution on the recovered analog side - which is much greater than what the sample time would suggest. To perform these tests, and evaluate what really matters, I would suggest to do the sync evaluation/comparison on the analog side.

//
 
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dualazmak

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To perform these tests, and evaluate what really matters, I would suggest to do the sync evaluation/comparison on the analog side.

Sorry, but I do not know/understand what would be your point.

In my async/sync measurements/assessments in my those posts (#783 and #804), always the analog output music/tone signals having timing markers from the DAC units were analyzed in 10 micro second (10 uSec) precision.
 

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Perhaps I misinterpreted this statement:

es.jpg


Perhaps you mean that what your method can observe/measure has a limitation of 22,7 usec because 16/44 dont have this limitation.

//
 
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dualazmak

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Perhaps I misinterpreted this statement:

View attachment 318581

Perhaps you mean that what your method can observe/measure has a limitation of 22,7 usec because 16/44 dont have this limitation.

//

OK, understood well.

My point was just simple; since the original test track used with timing markers are in 44.1 kHz 16 bit CD format of 22.7 uSec time resolution, my measurements/assessments in analog output signals in 10 uSec precision (by Adobe Audition 3.0.1) is more than precise enough.

Or in other words, less than 22.7 uSec precision measurement would not be needed, and my results obtained for the start timing difference 2.27 msec (post #783) and 2.28 msec (post #804) between OKTO and KORG are just the same, i.e. there is no meaningful difference between 2.27 msec and 2.28 msec.
 

TNT

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On the analog input side you can add analog time markers at tighter space than 1/2 the sample time and they will be retained on the output side of a DAC. This is the PCM systems time resolution and it is not limited by the sampling speed. Try it!

//
 
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On the analog input side you can add analog time markers at tighter space than 1/2 the sample time and they will be retained on the output side of a DAC. This is the PCM systems time resolution and it is not limited by the sampling speed. Try it!
//

Again, I am still afraid that I cannot fully understand what would be your suggestion, sorry about that.

In any way, you would be please kindly reminded that I have no further intention on performing any more experiments on this issue as I wrote in my post #828. In case you would like to further explore and discuss this issue, please start your new thread with proper and suitable title where you may refer to my posts #783 and #804, if you would need to do so.

EDIT: "your new thread" means not on "time resolution of PCM audio" but on "possible synchronization of multiple DAC units having different USB drivers connected to one PC" if needed. I assume almost no need/meaning for that, though, as you may kindly agree.
 
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pma

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On the analog input side you can add analog time markers at tighter space than 1/2 the sample time and they will be retained on the output side of a DAC. This is the PCM systems time resolution and it is not limited by the sampling speed. Try it!

//
You are absolutely right. It was explained many times, especially by @j_j , that time resolution of the digital audio is not limited by 1/Fs. Upper frequency limit and time resolution are 2 different things, however, many do not understand the difference. No need to “start new threads” on the obvious thing that was already beaten to death.

 
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dualazmak

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You are absolutely right. It was explained many times, especially by @j_j , that time resolution of the digital audio is not limited by 1/Fs. Upper frequency limit and time resolution are 2 different things, however, many do not understand the difference. No need to “start new threads” on the obvious thing that was already beaten to death.


Oh, thank you indeed for your clear educational message.:)
And I assume that I should understand @mdsimon2's suggestions in his post #827 based on your educational message.

Even though I now learned/understood your and @TNT's point, I have no further intention (as I wrote in my post #828) on performing any more experiment on this issue since there would be almost no practical value for my naive efforts on pseudo-synchronization of "analog output" from multiple different DAC units USB connected to one PC. I have already identified/confirmed msec level startup lags which well answered/cleared my primitive/naive curiosity.;)

EDIT:
BTW, I am sorry for my unclearness in my above post #837 for @TNT. I was suggesting to start his possible new thread not on "time resolution in PCM audio", but on "possible synchronization of multiple DAC units having different USB drivers connected to one PC", if needed. I assume almost no need/meaning for that, though, as you may kindly agree.
 
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Oct 10, 2017
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I was suggesting to start his possible new thread not on "time resolution in PCM audio", but on "possible synchronization of multiple DAC units having different USB drivers connected to one PC", if needed. I assume almost no need/meaning for that, though, as you may kindly agree.

There are a few solutions, all of them involve much swearing, cursing, and using a master clock across all of the DAC's or ADC's, OR (even worse) an asynchronous resampler.

Use a separate master clock. Always.
 
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