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If the Quad philosophy is that an amplifier should be a straight wire with gain....

I started to wonder whether I was going mad and had misremembered that the temporal resolution results I could recall were in the millisecond range. Random search pulled up this
comparing musicians and non-musicians with 3 different temporal resolution tests -duration discrimination, pulse train duration discrimination and gap detection. I think you can see from figs 1 and 2 that they lie in the ms range for both musicians and non-musicians.

I claim no expertise but does this support the notion that humans have temporal resolution abilities which could not be captured by 16/44? That humans have unlimited and uncharted temporal resolution. That there is soooooooo much "we" don't know about the uncanny hearing ability of humans?
aaaargh. I promised myself I wasn't going to fall into this rabbit hole. But....
The key text relied on by Kunchur is reference 146 which is this (Leshowitz)
I don't have access to the original text but compare Kunchur's

"This agrees with the measured ~4–10 μs TR thresholds for discriminating the gap between double pulses [146], which
is the only relevant experiment that could be found in the literature (as discussed below, various other “temporal resolution” experiments do not correctly probe “transient resolution” as defined here)*. Note that TR has no direct connection with fmax. Thus high-frequency hearing loss will not compromise the synchronicity detection between frequencies that are still audible" AND
."....The quintessential experiment for this is [146], which compared a pair of 10 μs pulses separated by a space ∆t versus a single 20 μs pulse. This produced a discernability of ∆t ~ 10 μs when the stimuli were isolated and ∆t ~ 4 μs when they were repeated with a periodicity of 0.2 ms. [146] was inconclusive as to the spectral versus temporal basis of the discrimination, and it correctly pointed out (first sentence on their page 464) that JNDs measured with continuous tones cannot be quantitatively applied to analyze transient signals"

with the abstract of [146] itself
"Observers were asked to discriminate between a pair of 10‐μsec pulses and a single 20‐μsec pulse having the same total energy. The independent variable was the time, ΔT, between the two 10‐μsec pulses. The stimuli were also presented as elements in a periodic pulse train. The ΔT required for resolution of two clicks (two‐click threshold) was 10 μsec. Whereas the addition of a steady background noise produced a remarkably small change in the magnitude of the two‐click threshold, performance deteriorated markedly when the pulses were low‐pass filtered. It appears that discrimination of slight changes in the energy spectrum of the two transient signals, especially in the high‐frequency region (8000 Hz and above), underlies the ear's sensitivity to a temporal discontinuity."

Obviously one has to be cautious because it's just the abstract, but ....it seems on the face of it seems that Leshowitz thought it was the variation in frequency which mattered - which is supported by the fact that the low pass filtering caused performance to deteriorate. But Kunchur has seized on this (quite old) test in isolation and chosen to interpret it as relating to purely temporal resolution (and note the only form of temporal resolution which matters) rather than frequency discrimination.
Why? Because as he just-about acknowledges- the more usual (and it seems fairly frequently repeated) measures of temporal resolution as used for example in lots of papers about the effect age related hearing loss, show a much more modest temporal resolution requirement for ordinary hearing (which of course deteriorates with age-see below).

Funny isn't that those other tests are not quoted at all, given that this supposed to be a "review of the human auditory system."
Interestingly another point he mentions repeatedly is in this "review of the human auditory system" is that people with age related hearing loss** only have their frequency range restricted but can still distinguish subtle time domain changes. Here Kunchur shows that he isn't just a physicist he is absolutely up to date with the audiophile agenda. Any fool looking at the literature of age related hearing loss will immediately see that err yes it affects temporal resolution not just frequency resolution. And now that takes us back to
"Note that TR has no direct connection with fmax. Thus high-frequency hearing loss will not compromise the synchronicity detection between
frequencies that are still audible"
which he gets from a paper whose abstract says
"performance deteriorated markedly when the pulses were low‐pass filtered. It appears that discrimination of slight changes in the energy spectrum of the two transient signals, especially in the high‐frequency region (8000 Hz and above), underlies the ear's sensitivity to a temporal discontinuity"
huh?
Anyway it would be nice if someone could get hold of the 1971 Leshowitz paper itself . Maybe kunchur is right about it...

* ie none of the researchers into human perception have thought it worth analysing temporal resolution in the way Kunchur does. They just analyse it in other ways that don't matter, the idiots.
** like most audiophiles
 
Not really, to capture a 1ms interval, you just need 2 kHz sampling. 16/44 would sample the briefest interval heard about 50 times.

However it depends what is meant by 'resolution'. The sampling rate can (via Nyquist) indicate the timing of an event to better than the sampling interval. Hence inter-sample peaks can be recorded and replayed. However the finite sample rate affects *resolution* of two very close peaks. Which aren't then different 'events' but an 'event' composed of two parts very close together. Which simply makes it harder to know what 'event' means.
 
The mistake often made is that the total signal strength of the noise 'floor' is compared to the signal strength of the single tone.
But the noise signal is summed over the bandwidth of interested, say 20kHz, whereas the tone is singular.

The portion of the noise signal that is spectrally near enough the tone so as to have masking power over it is but a fraction of the total noise signal. Hence the tone can be picked up relatively far below the (summed) noise.

The curio here is that people didn't make much fuss about modulation noise generated by analogue tape recording. Much bigger than the dither used for common digital sampling. Indeed, people seem to say they prefer analogue. (Hint: and 'vinyl' also has its own version of modulation noise. Molecules, etc, innit. 8-]
 
Anyway it would be nice if someone could get hold of the 1971 Leshowitz paper itself . Maybe kunchur is right about it...

What's it in? If it is AES then I or another AES member may be able to liberate a copy. Indeed. if it is early I'll have it on the CDROM set they used to issue before they hid stuff on the net!
 
What's it in? If it is AES then I or another AES member may be able to liberate a copy. Indeed. if it is early I'll have it on the CDROM set they used to issue before they hid stuff on the net!
J. Acoust. Soc. Am. 49, 462–466 (1971)
- journal of the acoustical society of America.
Bit of a tricky one. I’m guessing one needs some sort of general academic institution access.
 
Meanwhile, I decided to see what Brian Moore says about Leshowitz in Introduction to the Psychology of Hearing, and it very much looks like my hunch was (surprisingly) bang on the money
P.170
“A major difficulty in measuring the temporal resolution of the auditory system is that changes in the time pattern of a sound are generally associated with changes in its magnitude spectrum. Thus, the detection of a change in time pattern can sometimes depend not on temporal resolution per se, but on the detection of the spectral change. As an example, consider the task of distinguishing a single brief click from a pair of clicks separated by a short time interval. Assume that the energy of the single click is the same as that of the pair of clicks, so that the two sounds are similar in loudness. At first sight, this task appears to give a direct measure of temporal resolution. The results show that subjects can distinguish the single click from the click pair when the gap between the two clicks in a pair is only a few tens of microseconds (Leshowitz, 1971). This appears to indicate remarkably fine temporal resolution.
The interpretation is not, however, so straightforward. The magnitude spectrum of a pair of clicks is different from the magnitude spectrum of a single click; at some frequencies the single click has more energy and at others it has less energy. Subjects are able to detect these spectral differences, either by monitoring the energy within a single critical band, or by detecting the differences in spectral shape of the two sounds (as occurs in profile analysis; see Chapter 3). The spectral differences in this case are most easily detected at high frequencies. When a noise is added to mask frequencies above 10 kHz, the threshold value of the gap increases dramatically. Thus, the results of this experiment cannot be taken as a direct measure of temporal resolution.

There have been two general approaches to getting around this problem.

One is to use signals whose magnitude spectrum is not changed when the time pattern is altered. For example, the magnitude spectrum of white noise remains flat if the noise is interrupted, i.e., it a gap is introduced into the noise. The second approach uses stimuli whose spectra are altered by the change in time pattern, but extra background sounds are used to mask the spectral changes.”
Wow!
It seems to me that Kunchur has either totally misunderstood the field or written something downright misleading.
Given that I am quoting from what I understand to be the standard British university textbook on the subject, it seems fair to say that the article he cites is not generally accepted as a measure of temporal resolution as such.
And of course the generally accepted resolution tests show resolution around the millisecond rather than microsecond range.
This is highly problematic bearing in mind Kunchur’s claim that he has somehow demonstrated that dacs have insufficient resolution to capture the abilities of human hearing.
Where is the evidence that 16/44 let alone 24/96 can’t capture any of this or that a moderately good dac can’t reproduce it?

(I wonder how this was done in the Leshwitz test- live or from a recording.)
 
with all this talk about distortion I was reminded of this article
The late Arny K took me to task on the Logitech forum on this very point. He made a convincing argument (in typical bruising fashion) that any distortion becomes bad distortion via intermodulation, just as stated in first page of your link. I now see it this way at least when music becomes more complex (as it often does, and I like complex music).

But I'm not 100% sure! Any physical device will involve some compromises, and it's better to choose compromises that are less disturbing at least some of the time, given perfection isn't an option.
 
Personally I always took the “straight wire with gain” quote to mean flat across the audible frequency range,inaudible distortion,no added noise,nothing added nothing taken away.
Now personally I’d consider something to be inherently broken if it can’t achieve that (load dependcy,ps noise etc can change it)

Also with regards to distortion,I’ve suspected for a while if one listens to classical,jazz etc,ie unamplified music lacking distortion outside of the room and recording,one might fancy a shed load of distortion in the electronics.
But if your a metal fan,or a prog rocker,ie music with plenty of added distortion you’d probably prefer none added.
possibly maybe…..
 
Also with regards to distortion,I’ve suspected for a while if one listens to classical,jazz etc,ie unamplified music lacking distortion outside of the room and recording,one might fancy a shed load of distortion in the electronics.
But if your a metal fan,or a prog rocker,ie music with plenty of added distortion you’d probably prefer none added.
possibly maybe…..

I would have thought the opposite, that dry-sounding, close-miked studio mixes would benefit more from harmonic distortion than recordings which have capture some venue ambience.

Personally I'm for 'transparent' electronics regardless of genre (and find low distortion a must with Classical music).
 
I get the impression that whenever a piece of equipment produces harmonic distortion there also a bit of intermodulation distortion in the measurements.
Whilst the former might be generaly accepted as being euphonic (if low in level and low order), the latter will make the sound 'congested' or 'mushy', lacking 'clarity' and 'separation'. This is made more obvious with sonically complex music and large ensembles.

Loudspeakers also produce intermodulation distortion, with 3-ways performing better than 2-ways in this particular parameter, and 3-ways+sub even better:

lvxCSGC.png

source: Neumann
 
I get the impression that whenever a piece of equipment produces harmonic distortion there also a bit of intermodulation distortion in the measurements.
Whilst the former might be generaly accepted as being euphonic (if low in level and low order), the latter will make the sound 'congested' or 'mushy', lacking 'clarity' and 'separation'. This is made more obvious with sonically complex music and large ensembles.

Loudspeakers also produce intermodulation distortion, with 3-ways performing better than 2-ways in this particular parameter, and 3-ways+sub even better:

lvxCSGC.png

source: Neumann
I’d agree with this based on my empirical playing with the Nelson Pass NuTube pre amp. Adding quite significant (up to 2%) amounts of low harmonic distortion to audio, would give some music a much more expansive sound stage and subjectively improve things. More complex recordings didn’t have such a positive result, becoming as you suggest ‘congested and mushy’. Interestingly the amount and phase of the distortion changes the depth of the soundstage quite significantly from memory.
 
Meanwhile, I decided to see what Brian Moore says about Leshowitz in Introduction to the Psychology of Hearing, and it very much looks like my hunch was (surprisingly) bang on the money
P.170
“A major difficulty in measuring the temporal resolution of the auditory system is that changes in the time pattern of a sound are generally associated with changes in its magnitude spectrum. ...

Yes. Hence my comments a while ago. The above essentially looks at it from another angle. Two 'clicks' that start to merge to any extent mean you alter the frequency/temporal response. And the human sensors are a series of *resonant* detectors with variable 'Q' and gain affected by recent sound levels. The system is very nonlinear.

This is why I tend to be skeptical about people worrying about modest-to-low level THD. Or sensors nominally also generate distortions. Our brain function then 'interprets' this away from our awareness.
 
The late Arny K took me to task on the Logitech forum on this very point.

Diversion: That's the first mention I've seen here of Arny! :) My contacts with him were via Usenet (which I still use) and took part in 'debate' (sic) with him, Trotsky, and others. Sorry to hear he is dead. Now wondering if Pinky has also left the planet. Usenet has been duller recently. 8-]
 
I get the impression that whenever a piece of equipment produces harmonic distortion there also a bit of intermodulation distortion in the measurements.
Whilst the former might be generaly accepted as being euphonic (if low in level and low order), the latter will make the sound 'congested' or 'mushy', lacking 'clarity' and 'separation'. This is made more obvious with sonically complex music and large ensembles.

Loudspeakers also produce intermodulation distortion, with 3-ways performing better than 2-ways in this particular parameter, and 3-ways+sub even better:

lvxCSGC.png

source: Neumann
Interesting, thanks. For sure, more ways means less intermod.

We don't have 16-way speakers ... because crossovers have their own issues. My personal opinion is that the minimum number of crossovers needed with current tech is 2 (leading to a 3 way speaker, with mid covered by one driver without a crossover bestriding it).
 
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