The Black Hole......

This is what I'm not getting, the time resolution in the digital domain doesn't seem to require a bandwidth equivalent to what would be required in the analogue domain?

It simply are different meanings of the term "temporal resolution" ; Kunchur used it in his later explanation in the sense of resolving two events following after a short time period (otoh his experiments do not examine this resolving ability).

The two consecutively happening sound events the 44.1 kHz (Fs) system can't resolve while a 192 kHz (Fs) could.

The example at the Troll Audio website calculates the amount of timing error (Jitter) that could occur without reducing the effective number of bits (resolution, ENOB). A discrete quantizer (if working perfectly) comes with an uncertainty of 1/2 LSB and any additional error that should that leads to an error above a 1/2 LSB should be avoided as it effectively worsens the system ability.
Worst case condition for a reasonably used 44.1/16 system is a 20 kHz sinus with maximum level, therefore it is used in the Troll Audio example.
 
Here's another one from 4 days ago:
YouTube
"Hi-Res Audio, who cares? Music lovers, engineers, record labels, streaming services?"
He must have been reading this thread 😀

Thank you,
Even so interesting is the following video, comparing Pink Floyd's "Dark Side Of The Moon" in 6 different versions:
YouTube

1) Copy of the original master tape, 2 tracks, 15 ips
2) LP from 1973
3) Cassette in XDR, no Dolby but HX Pro from 1988
4) Reissue from LP in 2003
5) Together with 44.1/16 CD from 2003
6) And SACD also from 2003

Interesting to hear what you think of it.

His rating for CD and SACD was equal as good, no benefit for the SACD.
LP's were excellent and so was the Cassette apart from it's slight hiss
Master Tape number 1.

Hans
 
It simply are different meanings of the term "temporal resolution" ; Kunchur used it in his later explanation in the sense of resolving two events following after a short time period (otoh his experiments do not examine this resolving ability).

The two consecutively happening sound events the 44.1 kHz (Fs) system can't resolve while a 192 kHz (Fs) could.

The example at the Troll Audio website calculates the amount of timing error (Jitter) that could occur without reducing the effective number of bits (resolution, ENOB). A discrete quantizer (if working perfectly) comes with an uncertainty of 1/2 LSB and any additional error that should that leads to an error above a 1/2 LSB should be avoided as it effectively worsens the system ability.
Worst case condition for a reasonably used 44.1/16 system is a 20 kHz sinus with maximum level, therefore it is used in the Troll Audio example.
Jitter has nothing to do with what Kunchur tried to prove, but where he failed as I've mentioned before.
The effect of jitter on the clock is that it creates IM distortion, whereas the resolution in bits creates quantisation noise of sqrt(1/12Q) with Q the step size of 1 bit.
This quantisation could also cause trouble when the signal becomes too small leading to noise modulation, unless dither is added to prevent this.

So for a correct digital recording very low jitter and dither are compulsory.

Hans
 
Jitter has nothing to do with what Kunchur tried to prove, but where he failed as I've mentioned before.
The effect of jitter on the clock is that it creates IM distortion, whereas the resolution in bits creates quantisation noise of sqrt(1/12Q) with Q the step size of 1 bit.<snip>

I know that Kunchur wasn't talking about jitter, but as the Troll Audio example was introduced in this context and obviously leads to some question, I tried to explain.

As we know, the results depend on the function the jitter amplitude follows, if it is deterministic the result will be the IMD you've mentioned, if it is non-deterministic the result will be a raised noise floor.

It is a valid hypothesis that Kunchur's result could be explained by the small level difference in the fundamental (especially as it could be that training played a role) but we can't take it for granted.
The level increment detection ability for pure tones (with long duration) is an example for the so-called near miss of Weber's law, as it actually gets better with increasing sound intensity. At the 69 dB level which was used in Kunchur's experiment, the JND is actually thought to be around 0.5 - 1dB (somewhat speculative at the frequency of 7 kHz) but considering the usual spread among the population it seems very likely that at least a small proportion will be able to do better.

Otoh it seems quite unlikely that Kunchur's participants were coming from that specific proportion by chance, so further experiments are needed.

The modern model for our hearing sense is that it works like a time frequency analyzer, but I was wondering why Kunchur's probing of the temporal resolution (as he calls it) with the lowpass filtering experiment examines something different from his definition for the temporal resolution of the digital 44.1 kHz system. Due to temporal pre- and postmasking the human listening can't most probably not resolve that kind of small differences (intrachannel, interchannel or ITD is something very different).
But maybe I've to reread his publications.
 
1) Copy of the original master tape, 2 tracks, 15 ips
2) LP from 1973
3) Cassette in XDR, no Dolby but HX Pro from 1988
4) Reissue from LP in 2003
5) Together with 44.1/16 CD from 2003
6) And SACD also from 2003
I would say that test is invalid.
At first I would immediately discard all analog sources - there were waaaay too many unknown variables involved in producing digital files from those.
What's left are just CD and SACD. How the test files were produced from these two is also unknown.
From analog output of some CD/SACD player?
If not, were these exact 1:1 data rips? Verified? Sure? How did he get to the protected DSD layer? 😉
What software was used to upsample 44.1kHz and downsample 2.8224MHz data? Bug free? Enough precision?
Why the conversion was done to 192kHz, which is not multiple of 44.1 or DSD rate and therefore "less ideal"?
For me that's one signal manipulation too many.
 
Thanks Richard for your recommendation for Hi-Res sources. I am going into this area soon.
For everyone else, please understand that Richard or myself may be considered 'experts' due to our many decades of serious attention to audio improvements, but we don't know EVERYTHING! Sometimes, you just have to look around for yourself to get the answer to your question. I feel that we sometimes get blamed for not answering something that is beyond our direct knowledge, unfairly.

Let me know how I can help in any way... resources, people, mfr, parts, T&M or ideas and concepts etc.

-Richard
 
I do wonder how many mastering copies were sent out around the world and then got 'copied' by the night shift to take home...
Yes, that's one possibility. There were also copies of the copies of the copies that were sent to licensees for producing bin masters for casette duplication, which were simply left sitting on their shelves instead of being destroyed. At some later time they found their way into the private hands. (I'm speaking from my own experience..😉)
Also, given the right equipment, one can easily produce counterfeit tapes. Take a Hi-Res file, tweak it a bit to your taste, encode in Dolby A (which will add some veil and "warmth"), record on the right machine, preferably using old stock of tape and voila, you have your "copy of the master tape". Make up some story how you got it and some sucker will fall for it.

I hope you all realise that I'm being sarcastic here, not that I encourage such practices...
 
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Quite a few of the Hi-Res lovers never take the trouble of listening to audio fragments in several formats and reporting their findings.
Is this because their opinion is molded in concrete, or do they prefer to talk instead of listening.
It's O.K. for me, just wondering.

Hans
 
I would say that test is invalid.
At first I would immediately discard all analog sources - there were waaaay too many unknown variables involved in producing digital files from those.
What's left are just CD and SACD. How the test files were produced from these two is also unknown.
From analog output of some CD/SACD player?
If not, were these exact 1:1 data rips? Verified? Sure? How did he get to the protected DSD layer? 😉
What software was used to upsample 44.1kHz and downsample 2.8224MHz data? Bug free? Enough precision?
Why the conversion was done to 192kHz, which is not multiple of 44.1 or DSD rate and therefore "less ideal"?
For me that's one signal manipulation too many.

Good to hear that you watched the video.
Your comment is all true, one can manipulate things in all directions.
He nevertheless gave it a serious try and I found it amusing.

But you can simply listen to it as it is without bothering about all the technical details.
That's why I asked what the experiences from other forum members were.
It only takes a few minutes and doesn't prove a thing.

Hans
 
I think there are some people for whom it sounds better just because a different light has come on on the DAC.

I am happy to consider the possibility that the filtering has a bigger issue than the sampling rate. Although the $4500 Chord upsampler seems a rather expensive way to address it Chord Electronics Hugo M Scaler upsampling digital processor | Stereophile.com. But i think the difference is minimal either way.
 
I was lucky to have a meeting in 1990 where Angela Visser, miss world 1989, and Audrey Hepburn attended.
Angela was perfect in every way, but totally unattractive, whereas Audrey radiated attraction and emotion in all directions.

As I've said before: Music is all about emotion and has little to do with perfection, just my opinion.

Hans