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MDAC First Listen (part 00100011)

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Ron,

Sorry, I believed I answered, but the question is asked often so maybe I missed it.

I have no idea of the upgrade cost to ZFoils at this time as I need to build the PCB first and try the various combinations that achieve the best result (there is potentially 60 ZFoils resistor locations).

Here it is.
Peter
 
Hi John

Here is the serial number for the Mdac AH001842BLC3703 Could you pm me with the results if its an upgrade model or not.
 
Hi John

Here is the serial number for the Mdac AH001842BLC3703 Could you pm me with the results if its an upgrade model or not.

Mike,

Renata searched our database - and its not a serial number that's known to us as an upgraded unit.

Normally there's a second serial number add or a larger Lakewest Label with upgrade details.

I sometimes perform one or two tweaks to units that pass my way for repair - but it does not appear to be an official "Paid for" full on upgraded unit.

If you have more history of the unit (or owner) then I'll for sure recall if it visited us here in Czech Rep. at anytime.
 
Before investing, please note that there's no added benefits increasing the frequency.
Contrary to what many people believes regarding frequency in digital sound, it doesn't bring more resolution, only increases the limit of the high frequencies further into bat and dolphins perception domain.

If, for a given sound wave, you need "2 dots" for a precise wave, adding 2, 10 or 100 dots more, doesn't change neither the calcs nor the resulting audio wave.

Two links if you care to take a further look into the subject:

http://xiph.org/~xiphmont/demo/neil-young.html

http://www.trustmeimascientist.com/...io-poll-neil-young-and-high-definition-sound/


Michael
OK I'm both going off half cocked and shooting from the hip because I haven't yet read whatever those links lead to, but I'm impulsive, so:
That's a big "if", what if 2 dots aren't enough for a precise wave? And, if you increased the sampling frequency to infinity, wouldn't that be analog? Given that analog must in principle be better than digital if both were perfectly executed (because you get everything) doesn't increasing the frequency just take you nearer to the theoretical ideal?
I will follow those links BTW.

I think I can hear hornets.
 
That's a big "if", what if 2 dots aren't enough for a precise wave?

Well, 2 dots is enough. The Nyquist-Shannon sampling theorem has been mathematically proven. So your big "if" becomes "what if mathematics doesn't work?"

And, if you increased the sampling frequency to infinity, wouldn't that be analog?

No.

I will follow those links BTW.

Good, because they answer your questions pretty well :)
 
OK I'm both going off half cocked and shooting from the hip because I haven't yet read whatever those links lead to, but I'm impulsive, so:
That's a big "if", what if 2 dots aren't enough for a precise wave? And, if you increased the sampling frequency to infinity, wouldn't that be analog? Given that analog must in principle be better than digital if both were perfectly executed (because you get everything) doesn't increasing the frequency just take you nearer to the theoretical ideal?
I will follow those links BTW.

I think I can hear hornets.
I don't suppose John would thank anyone for turning this thread into another instalment of the endless debate on this topic, but I would suggest that the answer can be summarised as follows
1. What cannot really be disputed is that the information in a band limited signal can be perfectly captured by sampling it at (just over) twice the rate of the band limit eg a signal limited to 0-20Khz provided you sample at (just over) 40Khz. In this sense the two dots really are enough for a wave provided that it is suitably band limited

2. Thus the issue really turns on the question of whether
a. a precisely limited frequency band (say 0-20kHz) is enough to reproduce everything you can hear
b. the process of band limiting (before sampling and as part of the D/A process) might somehow create audible artefacts ie can you tell the difference between a signal which has been so band limited and one which has not.
1. is a mathematical proposition, 2. is essentially a psycho-acoustic one although obviously realising a solution in the real world involves engineering.


[I should have stopped there but couldn't resist the following. Feel free to ignore.

3. I think the concepts of like "theoretical ideal" and "perfectly executed" are
are potentially confusing here. Even "analog" is when applied to the real world as opposed to a system for recording it. There is a frustratingly common sleight of hand which leads to the following false syllogism
a. the world is analog
b. analog music reproduction systems are analog and digital ones aren't
c. analog music reproduction systems are like the real world and digital ones aren't

The problem comes with confusing one property of analog systems (not being discrete time/discrete quantity) with being analog (ie reproducing a party of the real world eg sound pressure by reference to another (voltage level, undulations of a groove in vinyl)). If you run through the argument substituting discrete time/discrete quantity for "analog" in a, and for the second analog only in b., then where do you get? That (arguably) analog systems are like the real world in one respect. Incidentally it may be that the real world is not discrete time/discrete quantity, but that's another story.]
 
Well, 2 dots is enough. The Nyquist-Shannon sampling theorem has been mathematically proven. So your big "if" becomes "what if mathematics doesn't work?"



No.



Good, because they answer your questions pretty well :)

I honestly think that most of the differences we perceive in HQ audio is more related to the 24bits vs 16bits than to sampling...and probably to the different processing involved (low-res audio is usually up-sampled and filtered).
 
I honestly think that most of the differences we perceive in HQ audio is more related to the 24bits vs 16bits than to sampling...and probably to the different processing involved (low-res audio is usually up-sampled and filtered).

While I agree with the latter part, I still haven't come across a recording using more dynamic range than what 16 bits provides.
 
Well I tend to agree with Ian's thinking - if only because despite what's been apparently "Proven", Digital is sonically a pale comparison to a High quality analogue source - such as turntable or Master Tape.

On every possible occasion I like to spend time in recording studios basically to "set a reference" for myself. What I mean too say is to hear Music "direct" though studio monitors / Mixing desk before is recorded onto a medium, learn about recording techniques, understand the artists etc etc - I'm grateful for these times as it helps me as a "HiFi" designer to have a target!

One thing I can say is that most studio equipment thesedays is total crap - so much worst then HiFi, its truly horrid! then not to mention the oxidised patched boards and miles of "unknown" signal cables... Studios like to run cables back and forth just to maintain "flexibility".

A good friend has a 24track tape machine 1 floor down from the main studio - the cable for all 24 replay and record channels runs across hallways and up/down a staircase, and yet he produces better audio quality recordings then most. Yet when recording onto decent Virgin tape on 2 track 1inch machine (with correct Bias settings) the quality is STUNNING - yet the second its digitized even at 192/24 gone is the realism I'm hearing in the studio monitors.

So you can tell me all day how "perfect" digital is, yet I can hear for myself clear as day that it is anything but perfect!

MDAC2 L3 with its ADC input is really a personal experiment to understand what's at play - I plan to plug the MDAC2 into the stereo mix channel in the studio and experiment with filters, modulators, sampling rates to try and close the sonic gap with analogue tape machines which even on 1inch / 2 track measure relativity poor!

I've mentioned in the past that I believe possible root cause is the Digital filters required in PCM systems as DSD gets very close.

A manufacturer asked me to design a digital delay line to replace the old hard to source "Analogue" CCD delaylines - the manufacturer required it to be based on DSD as they felt PCM was unable accurately capture the transient nature of music.

During the development of the delayline I could listen to a direct Mic feed and it was SO apparent that PCM system (that could sample upto 250KHz odd) was unable to capture the musical transients - and emphasised the sibilance in female vocals. The DSD / PCM path was selectable on the PCB using the same front and backend ADC / DAC just the internal data path between devices was selectable on the "fly" with a toggle switch on a long lead (DSD / Direct / PCM).

Without checking the orientation of the switch (which would change depending how you held the switch on its lead) one would not know what mode the Delay-line was in, and being able to switch between direct and DSD or PCM path was a perfect "Blind test".

I say "Blind test", but it soon became all too apparent that when anyone was demonstrated the system they you could ALWAYS pickout the PCM mode - where as the DSD path was impossibly close to the direct path.

It just so happens that an accountant was working on the books in the Studio / Lab and she randomly join in one of test - its was fascinating and reassuring to see she had exactly the same observation we "engineers" had - this lead to a so often repeated conversation about how poor PCM is apparently unable to capture what I refer to as "the leading edge transients" - and destroy the innate beauty of female vocals adding a hardness / sibilance...

So you can tell me all day how mathematically perfect digital is - but until I hear it with my own ears its only words and opinions - I'm not some stupid lemming who blindly believes everything that's "scientifically" proven to me despite my own experiences.

I don't design turntables or Tape machines - I tend to design digital products so its REALLY not in my interest to say "My" designs are poor but sadly that's the reality, PCM digital appears imperfect and (I) we would like to understand why.

The reason I've spent 3 years developing the MDAC2 (thats 3 years of my life I'll never get back) and yet we plan to build only say 200 units is to serve as the very highest Digital "Development" platform with ADC and DAC path which can be configured in every conceivable (and yet to be conceived :) ) data format / Mode enabling us to experiment with "Digital" with simple software (Firmware) updates.

The great thing is there will also be 200 or so "blind testers" who can forward there own listening observation and discuss it here openly on the forum :) This is where it well get exciting, and hopefully people will be able to judge for themselves at home what I experience in the studio / lab - then that's a worthwhile result for three years work (not to mention all the work still to come with the FPGA / DSP firmware / software) :)
 
Yes

The real comparison - the real subject for a blind test, if you want one - is reproduced sound vs. original live sound

Very little comes close
 
Yes

The real comparison - the real subject for a blind test, if you want one - is reproduced sound vs. original live sound

Very little comes close

Yes, very much so, I'm glad I had the opportunity to design the delay line as originally it was not such an interesting project - but it left lasting memories and opened my eyes to DSD and also a trained me to hear the faults of PCM via a live Mic feed.

What was also surprising was how good "Horrid" CCD devices can sound - OK noisy etc. but they managed to preserve the realistic nature of Music, where PCM digital could only portray a pale imitation.
 
What was also surprising was how good "Horrid" CCD devices can sound - OK noisy etc. but they managed to preserve the realistic nature of Music, where PCM digital could only portray a pale imitation.

Interesting.
What was the typical sample freq of a CCD delay line?
What filtering (if any) was applied before the CCD? (brick wall?)
 
Gosh, IIRC the sampling rate was about 1MHz to 2MHz (depending upon the required delay) and not so much a brickwall but a more gentle LPF.

IIRC, there was a 2 phase clock so they would effectively double the sampling rate - or some odd... sadly they are only New Old Stock so we did not spend much time with them, only as we where challenged "If you can make a digital delay line sound as good as these".

We run them fast / Max during the listening test as we where not interested in delay time - but the point was made...
 
2. Thus the issue really turns on the question of whether
a. a precisely limited frequency band (say 0-20kHz) is enough to reproduce everything you can hear
b. the process of band limiting (before sampling and as part of the D/A process) might somehow create audible artefacts ie can you tell the difference between a signal which has been so band limited and one which has not.

Indeed there is some experimental medical evidence that brain activity can be detected when people are played sounds > 20kHz. Though the mechanism for this is not yet clear. It may not be "hearing" in the classic sense, but it's obviously part of experiencing sound.

- Richard.
 
Gosh, IIRC the sampling rate was about 1MHz to 2MHz (depending upon the required delay) and not so much a brickwall but a more gentle LPF.

IIRC, there was a 2 phase clock so they would effectively double the sampling rate - or some odd... sadly they are only New Old Stock so we did not spend much time with them, only as we where challenged "If you can make a digital delay line sound as good as these".

We run them fast / Max during the listening test as we where not interested in delay time - but the point was made...

I was interested in what we'd get/hear using analogue sampling - changing the sample frequency, filters etc
 
2. Thus the issue really turns on the question of whether
a. a precisely limited frequency band (say 0-20kHz) is enough to reproduce everything you can hear
b. the process of band limiting (before sampling and as part of the D/A process) might somehow create audible artefacts ie can you tell the difference between a signal which has been so band limited and one which has not.

Well put, I'm inclined to trend towards #2 - certainly I believe the brain requires certain "information" that is removed in a BW limited Sampled system.... the method by which our brain process "Time Domain" information.

Transient information and "sound stage" are the two aspects that for me are so compromised with PCM 176KHz yet DSD64 which has the same digital data BW is able to preserve.
 
I was interested in what we'd get/hear using analogue sampling - changing the sample frequency, filters etc

You know like Tape and Turntables, the performance of CCD devices can only be described as very crude yet they sound more realistic them PCM 215KHz...

I'd like not to be biased, but based upon experience I believe the MDAC2 DSD recordings will be closer to the original... but there are many "blind" tests ahead.

I'd like to know if we can improve upon PCM performance - certainly the optimal transient filters of the MDAC are an example of trying to get closer to the original with PCM - but they very much have there own limitations.
 
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