Blue or red pill?

[fas42]

1 minute ago, PeterSt said:

 

No. This only happens with the "digital" comparison I talked about and this was/is covered for (a hard pain, but it was/is).

For what the subject of the thread ia about, this is harmless because we talk about two (or more) takes and all will behave the same (delay and such is always the same).

We're not looking at molecule level, right ?

(too much noise for that )

 

Doing a "digital" comparison is the only worthwhile option - an analogue comparison is so dependent on the recording apparatus, so much distortion and noise will creep in that it would be a nightmare ...

 

An immediate capture to digital representation at the "horse's mouth", at the highest possible accuracy, would be the starting point - then, maths can take over and make of it something worthwhile.

[fas42]

8 minutes ago, manisandher said:

 

You mean before any digital-to-analogue conversion has taken place?

 

But Mans has already done this analysis and has concluded categorically that the 24/176.4 digital captures taken during the A/B/X are bit-identical (ignoring the first 10k samples on some of them).

 

What would you be able to do that Mans can't?

 

Mani.

 

No. Where the "damage is done" is in the conversion to analogue, within the playback chain. A capture of that analogue, using a precision ADC at a point close to where that first conversion took place, is the "horse's mouth". Of course, there is always a possibility that the presence of the capture ADC in turn interferes with, degrades the analogue input, via some subtle coupling mechanism - can the measurer take a reading in an experiment, without unduly affecting what he is trying to measure? - an ongoing dilemma for all scientists.

[fas42]

6 minutes ago, PeterSt said:

For fun : look closer at the anomalies above. You can see that the repetition is the most frequent in time. Use the large excursions to see the smaller ones throughout. The distance is e.g. 00:0048.010 to 00:0048.030 (merely 0.08 to 0.28). The 48 is seconds. So this happens each 2/100 of a second. You know what ? this is 20us. And you know what more ? this is the actual Windows time resolution (at least back then).

Now that ...

Find that timer (or in your own software or in other software running), eliminate it and have again better sound.

 

Now this is interesting - you're implying you can pick the converter glitching, in time with the the ticking of a clock somewhere in the m/c - this is the "smoking gun", if that's what it's showing?

[fas42]

6 minutes ago, PeterSt said:

Frank, I read your last post in response to Mani. But this is what you said causing Mani to pose the question :

 

 

Are you able to improve your Dutch a little ?

 

It's all Greek to me ...

[fas42]

3 minutes ago, PeterSt said:

No, not really.

...

And then to think we explicitly work in the analogue (D/A) domain using the PC for it.

Don't you get the creeps by now ?

 

Well, I've got the 's - that's for sure ...

[fas42]

7 hours ago, STC said:

 

Thanx Mansr. Never knew that. So somehow data transmitted via spdif is always without errors? If so how are getting bit perfect via spdif?

 

People have far too much of a hangup about any errors in the digital stream - I'm using an older NAD CDP at the moment, which from new was notorious for not liking burnt CDRs; and mine is the same. Yes, it can play the disk but it's doing it hanging on by its fingernails - constantly audibly glitching, as error correction and interpolating does its damn best ... but this doesn't get in the way of the music coming through! The tonality and qualities that I look for are all still there - it's like listening to an LP that's full of crap in the grooves, on a top notch vinyl system; the latter is not "degraded" by playing from media in poor shape.

[fas42]

11 minutes ago, sandyk said:

 Frank

You have of course tried cleaning the Laser diode area (carefully) with something like Isopropyl Alcohol ?

 If so, the Laser Diode is probably on the way out.

A few minor masked errors will not alter how the music sounds, yet so many have a hangup about Bit Perfect copies.

 

Alex

 

Of course ... no, this is a generic issue with NAD CD players of that vintage; if you look around forums, you will come across plenty of discussion, and throwaway lines regarding this. I think even the NAD company acknowledged this, in some printed material. Amusingly, the older Philips HT box had a dodgy laser - that is in fact why it was dumped by the owner, could no longer read DVDs reliably - a good clean helped, but if not run for a while would play up a bit, initially.

 

Yes, other factors are much more important - get those right before obsessing about "digital perfection" ...

[fas42]

In all this processing, the "jewel" is the circuitry which is doing the actual job of converting the waveform which has digital meaning, into an analogue relevant form - the tight group of electronics which is doing this has to be as pristine as possible ... all the digital side can be as dirty, as jittery as one can get away - on the scope it can look a complete mess, so long as the digital data content is never lost or corrupted. But the core conversion area has to be like the clean room in an integrated circuit fab shop - scrupulously bereft of any unwanted "stuff"; outside can be as noisy as you like, so long as nothing can worm its way into the clean area, via any gaps in the "shielding".

 

Then, in the following circuitry which conditions and carries the analogue representation to the amplifier, etc, all care must be taken.

 

Worrying about everything in a sloppy way will probably still contaminate the waveform; concerning oneself with precisely what matters and getting this right, is the optimum strategy.

 

 

[fas42]

36 minutes ago, Ralf11 said:

that is inside of an IC ...

 

In some cases, not in others - say, dCS, or MSB. If there is only a single chip doing the job then that single part can be screened as necessary - this sort of engineering is trivially applied in RF circuits. Also, every single pin going into the chip could a carrier of unwanted noise, etc - this needs to be carefully scrutinised, thought through, dealt with appropriately.

[fas42]

9 hours ago, manisandher said:

Perhaps the only way to make a DAC truly immune to these sorts of influences is to block the digital signal altogether - if the digital signal can get through, so can noise at the same sort of frequency. In which case, we'd better learn to live with, and perhaps start manipulating to our own advantage, the noise entering the DAC... which I think has been @PeterSt's quest for the last 10 years or so.

 

Mani.

 

That's an unacceptable compromise, to me. Engineering is all about having each part of a system doing its job well enough for the final result to reach a certain standard - and the standard should be, only what was encoded in the recording should be audible.

 

Digital audio is hard, because interference effects can intrude, far too easily - at the moment. Which is a long way from saying it's impossible to having the path immune to such factors ... it just takes a bit of dedicated focus to exorcise all the demons that get in the way of satisfying listening, I've found.

 

To my eyes, a lot of the implementation is less than optimum, in standard designs - just changing the practices here will have major impact ... it's all a learning curve.

[fas42]

4 hours ago, esldude said:

There are some poorly designed devices easily corrupted by noise though most aren't. I've no idea about the DAC they used for the test.  We've not heard from the analysis of the recordings yet so jumping the gun a bit to say very much either way. 

 

 

All audio playback devices are affected by noise - that's how the universe works. The question is rather whether the effects are too low level to be audible in any context ... there is no White Hat DACs, and Black Hats DACs split - there is a continuum of behaviour characteristics, and the aim is to get what one is using at the right end of the behaviour spectrum.

[fas42]

2 hours ago, semente said:

 

What kinds of distortion could produce these effects besides jitter?

Phase distortion, IM?

 

Referring to these comments,

 

Quote

 

I was listening to the piano transients - were they 'incisive' or 'blurred'/'soft'? With the female vocals, I was listening to her sibilance - acceptable, or too sharp and annoying.

 

Generally, it's much more about 'focus' than it is about the 'quantity' of any particular thing. For example, I didn't hear more bass, or more highs in one over the other, or anything like that.

 

 

 

 

This is exactly the type of thing I listen for when optimising a system - blurred/soft transients, or sharp and annoying sibilance are classic markers of faulty playback - a clear indicator of some degree of incompetence of the playback chain.

 

What kind of distortion? Ummm, I would be hesitant to apply some nicely defined term to it - somewhere, part of the circuitry is not working as well as it should, and I'm far less interested in getting some 'evidence' that reads well in a journal article, than in simply fixing the flakiness that's causing the problem ...

 

 

 

[fas42]

In an audio system, some factor was changed  - and the sound changed, was audibly variant ... and this to some people is remarkable ... if only the universe was nice and simple, like in the good ol'' days when you just had those protons, electrons and neutrons swirling around - then some bastards came along and messed it all up with crazy ideas like quarks, and quantum behaviour, etc, etc ...

 

To me it would astounding to be able to perfectly replicate the analogue output of an audio setup over a number of runs, to the point of not being able to point to clear differences by some measurement - as always, it comes down to unburying that part which exactly correlates with what our hearing system picks up on; as Peter would say, good luck on doing that!!

[fas42]

3 hours ago, semente said:

But apparently it's not the circuit, just a change in software settings, because one of the options sounds good/better...

 

The change in software settings alters how the hardware which is sourcing the recording operates - the nature of its electrical activity, over time, is different - the circuitry which is concerned with creating and processing the analogue representation should be completely oblivious to that change; but in the real world that is not the case.

 

It is the circuit, in the sense that it is too sensitive to its environment - the, obvious, solution is improve the engineering, after the fact if necessary, to stop this being the case.

[fas42]

3 hours ago, adamdea said:

 

You could, with equal or greater validity, pose the question “is it possible to design a dac so badly that its output varies audibly according to the non-information properties of the input data? And you may (but only may) have answered that question.

 

I prefer the term "robust", rather than "isolation" - will the system, moving beyond component behaviour, always "sound the same" no matter what is happening in its environment? IME, this is extremely difficult to achieve - a combination of the sensitivity of the human hearing system to subtle anomalies, and the way audio systems are put together.

 

IOW, "all DACs are badly designed" ... IME, it's trivially easy to do something relatively innocuous in the electrical surroundings of an audio system, which causes the sound to change - just because standard measurements pick up nothing abnormal is meaningless - "you're not measuring the right things!".

[fas42]

13 minutes ago, Andyman said:

 

What constitutes a dac?

 

As designers are becoming more creative in finding ways to “improve” (isolate?) the signal sent to the dac (rendu, regen, lush cable, whatever) it would reasonably follow that if accepted these concepts work, they may in future all be incorporated into one box; presumably still a dac. 

 

Not contrary to to your definition but maybe moving the goalposts.

 

System rather than component thinking will solve most of the problems - this will offend the hobbyist crowd who just want to play with Lego bricks; but if your goal is to experience the recording, rather than the playback rig, than this approach will be the most effective route.

[fas42]

5 minutes ago, acg said:

Better isolation for sure,  lower noise floor,  a bit or so of extra resolution and better SQ but something that did not matter before now matters... and SFS changes if anything are more audible. 

 

Go figure!

 

A key Law of Audio applies, in all situations ... the closer one is to hearing the raw state of the recording, the more everything one does will have an audible impact. There is no way around this - at some point decide that you're "close enough" ... and, be happy ... .

[fas42]

On 14/04/2018 at 6:40 PM, manisandher said:

 

 

I'm not sure what a 'diff' program would find (I know you have to be careful with dud results from some of them). I suspect the signals are too complex or "chaotic" to identify any consistent differences between the two playback means, even within just a small section of time. The fact that the human ear can suggests that this is perhaps another inappropriate measurement.

 

But if anyone is interested in trying this, I'll be uploading soon the analogue captures that we took when Mans was here.

 

Mani.

 

A well done diff'ing analysis should find something of significance - they most certainly will vary, the hard bit will be pinpointing those aspects which matter to the ear, in the sense that we are talking about it here ... I'm certainly interested in having a go, .

 

It's trivially easy to see how badly the soundfield is "mangled" in some uploads of original versus system reproduction waveforms - typically, transients are way off what they should be ... in the situation here it will be far, far more subtle ...

[fas42]

Just downloaded the files, and my first thought is that this piece of music is as far from what I would use to evaluate a rig as one could get ... "You wish to testdrive the Ferrarri, sir? ... OK, a few rules: never go over 2,000 RPM; never go beyond 2nd gear; only on the smooth road in front of the showroom; there's a glass to water inside that will tip over and let us know if inertial forces are too great ... enjoy the run, sir!!" ...

[fas42]

11 minutes ago, sandyk said:

 

 This track is at quite a low level, and will highlight any masking low level noise which DOES affect how individual female voices can sound.

 

I work in a different fashion - I don't care whether her voice is different, I want it to to be convincing ... I have a recording of Odette, the Harry Belafonte one, from the '60s, which fits the bill of being extremely low level - the system either makes her voice a caricature of that style of singing; or, a living, breathing person.

[fas42]

Sorry, I've had to deal with those boring other things in life for the last day or so, have not read the recent posts  - just fired up Audacity, and almost immediately came across a distinct variation in a waveform peak, beween A and B - in the analogue captures of course - enough to be audible, was it a foible of the recorder? Is there a pattern, or was it a single glitch? Will have little time to investigate further for a day or so, but there is a first tiny glimmer of something ...

[fas42]

11 hours ago, PeterSt said:

 

Frank, if it was only well after 50ms in the file(s) ...

 

Well after half a second in - another quick look, there is tonnes of of glitchy variations between the two, only about 40dB down from the magnitude of the of the signal itself at a particular point - I would be amazed if this wasn't audible to least some people.

 

Now, is this 'noise' an artifact of the recording process, or an indication of something meaningful - will need much more careful investigation, to be looked at in a day or two ...

[fas42]

I note that I'm way behind where people are at - need to read all this carefully, before adding anything further, .

[fas42]

1 hour ago, pkane2001 said:

 

I didn't realize they were different. Will try again.

 

I was suspicious when the results of diffmaker looked mostly the same as when I aligned the two waveforms by hand. I still think it isn't working quite right. 

 

 

Some time ago I deliberately slightly adjusted, disturbed a perfect copy of a clip in various ways - and fed the two to DiffMaker to compare. It never was able to pick what I had done, the results had no bearing at all to the variations I had engineered - hence I have zero confidence in the program to tell me anything useful.

 

I'm going to be doing comparisons of the clips here completely manually - should be able to start doing this later today.

[fas42]

One thing I have noted is that in the specific area where I first saw a 'glitch' in the analogue A, is that when I looked at the corresponding area in the other 2 captures of ana A that they were also 'not right', but in 'random' variations of the first. Was this purely noise, variation by the DAC, or variation in the ADC? ... to be looked at more ...