speakers for 24/96 audio

Hi Bryon,

Interesting question, and an interesting paper, which I read through. It strikes me as very intelligently and knowledgeably written, and I see no obvious flaws in the details he presents. And intuitively it does strike me as plausible that our ability to resolve timing-related parameters might be somewhat better than what would be suggested by the bandwidth limitations of our hearing mechanisms.

However, looking at his paper from a broader perspective I have several problems with it:

1)He has apparently established that listeners can reliably detect the difference between a single arrival of a specific waveform, and two arrivals of that waveform that are separated by a very small number of microseconds. I have difficulty envisioning a logical connection between that finding, though, and the need for hi rez sample rates. There may very well be one, but I don’t see it.

2)By his logic a large electrostatic or other planar speaker should hardly be able to work in a reasonable manner, much less be able to provide good reproduction of high speed transients, due to the widely differing path lengths from different parts of the panel to the listener’s ears. Yet clean, accurate, subjectively "fast" transient response, as well as overall coherence, are major strengths of electrostatic speakers. The reasons are fairly obvious – very light moving mass, that can start and stop quickly and follow the input waveform accurately; no crossover, or at most a crossover at low frequencies in the case of electrostatic/dynamic hybrids; freedom from cone breakup, resonances, cabinet effects, etc. So it would seem that the multiple arrival time issue he appears to have established as being detectable under certain idealized conditions can’t be said on the basis of his paper to have much if any audible significance in typical listening situations.

3)More generally, it seems to me that there are so many theoretical, practical, recording-dependent, and equipment-dependent variables that would have to be reckoned with and controlled in any attempt to make a meaningful comparison involving hi rez vs. redbook sample rates, that reaching a definitive conclusion about the degree to which this particular factor may be audibly significant under real-world listening conditions is probably not possible.

All best regards,

--Al

I agree with Al.

This shows how good we are at hearing sounds and nothing to do with temporal resolution.

The wavelength at 7KHz is 5cm. Therefore in order to get the direct sound completely out of phase at the listener one need only move one speaker back by 2.5 cm (half a wavelength). This will result in the direct sound being Zero and will probably reduce the SPL level to be clearly audible. The fact that only a 2.9 mm movement was audible suggests that reflections may also have played a role here too.

The use of pure signal of a single tone with no (audible) harmonics can often gives surprising results! This is not reflective of musical instruments that have many harmonics so it is hard to draw any conclusion other than a test tone produces an audible result. Anyway my money is that there is enough of an amplitude difference here to make it audible in the case of a pure test tone. A pure test tone will fluctuate as you move around the room (you get peaks and suckouts depending on how it all adds up (reflection and direct sound).

"Irv, keep in mind that it is generally accepted that signal can be perceived at levels that are significantly below the level of random broadband noise that may accompany the signal. 15db or more below, iirc. So amplifier noise floor is not really a "floor" below which everything is insignificant."

Maybe, but it is very difficult to believe this is the case when listening to music or other complex sounds, like movie dialog or foley. I've always been leery of effects 70db or more below the music level, regardless of the component in question.

Hi Al and Shadorne.

Thanks for your thoughtful responses. Everything you guys said makes sense to me, but I do have some additional thoughts...

07-04-11: Almarg
1)He has apparently established that listeners can reliably detect the difference between a single arrival of a specific waveform, and two arrivals of that waveform that are separated by a very small number of microseconds. I have difficulty envisioning a logical connection between that finding, though, and the need for hi rez sample rates. There may very well be one, but I don’t see it.

I believe Kuncher addresses this in this document, in which he says:

For CD, the sampling period is 1/44100 ~ 23 microseconds and the Nyquist frequency fN for this is 22.05 kHz. Frequencies above fN must be removed by anti-alias/low-pass filtering to avoid aliasing. While oversampling and other techniques may be used at one stage or another, the final 44.1 kHz sampled digital data should have no content above fN. If there are two sharp peaks in sound pressure separated by 5 microseconds (which was the threshold upper bound determined in our experiments), they will merge together and the essential feature (the presence of two distinct peaks rather than one blurry blob) is destroyed. There is no ambiguity about this and no number of vertical bits or DSP can fix this. Hence the temporal resolution of the CD is inadequate for delivering the essence of the acoustic signal (2 distinct peaks).

In essence, I understand him to be saying that the temporal resolution of human hearing is around 6μs. But the temporal resolution of the 44.1 sampling rate is around 11μs. Since the temporal resolution of human hearing is better than the temporal resolution of 44.1 recordings, those recordings fail to accurately represent very brief signals that are both audible and musically significant. For example, Kunchur says:

In the time domain, it has been demonstrated that several instruments (xylophone, trumpet, snare drum, and cymbals) have extremely steep onsets such that their full signal levels, exceeding 120 dB SPL, are attained in under 10 μs…

He also suggests that the temporal resolution of 44.1 recordings might be inadequate to fully represent the reverberation of the live event:

A transient sound produces a cascade of reflections whose frequency of incidence upon a listener grows with the square of time; the rate of arrival of these reflections dN/dt ≈ 4πc3t2/V (where V is the room volume) approaches once every 5 μs after one second for a 2500 m3 room [2]. Hence an accuracy of reproduction in the microsecond range is necessary to preserve the original acoustic environment’s reverberation.

I’m not saying that these claims are true. I’m just trying to give you my understanding of Kunchur’s claims about the connection between human temporal resolution and the need for sampling rates higher than 44.1.

07-04-11: Almarg
2)By his logic a large electrostatic or other planar speaker should hardly be able to work in a reasonable manner, much less be able to provide good reproduction of high speed transients, due to the widely differing path lengths from different parts of the panel to the listener’s ears. Yet clean, accurate, subjectively "fast" transient response, as well as overall coherence, are major strengths of electrostatic speakers. The reasons are fairly obvious – very light moving mass, that can start and stop quickly and follow the input waveform accurately; no crossover, or at most a crossover at low frequencies in the case of electrostatic/dynamic hybrids; freedom from cone breakup, resonances, cabinet effects, etc. So it would seem that the multiple arrival time issue he appears to have established as being detectable under certain idealized conditions can’t be said on the basis of his paper to have much if any audible significance in typical listening situations.

I think perhaps Kunchur does his own view a disservice by emphasizing the deleterious time-domain effects of speaker drivers with large surface areas, e.g. electrostatic speakers. It seems to me that those deleterious effects might be offset to a large extent by the very characteristics you mention, viz., light mass, minimalistic crossover, etc.. But your objection does seem to cast doubt on the significance of the very brief time scales that Kunchur contends are audibly significant.

Having said that, the putative facts about jitter bear on this point in a somewhat paradoxical way. According to some authorities, such as Steve Nugent, jitter is audible at a time scale of PICOseconds. For example, Steve writes:

In my own reference system I have made improvements that I know for a fact did not reduce the jitter more than one or two nanoseconds, and yet the improvement was clearly audible. There is a growing set of anecdotal evidence that indicates that some jitter spectra may be audible well below 1 nanosecond.

That passage is from an article in PFO, which I know you are familiar with. I bring it up, not to defend Kunchur’s claims, but to raise another question that puzzles me:

If jitter really is audible at the order of PICOseconds, does that increase the plausibility of Kunchur’s claim that alterations in a signal at the order of a few MICROseconds are audible?

Again, I don’t quite know how to make sense of all this. I’d be interested to hear your thoughts.

Bryon

Well, Bryon, that was a very interesting article. I'm not sure what to think after reading it... is this yet another investigation into a micro-problem that doesn't really affect music reproduction, or is it a significant factor? I certainly don't know. I can't even venture a guess.

Anyway, Kunchur admits to listening to cassettes. I haven't heard cassettes for many years, but 16/44 CDs must sound like a revelation by comparison. ;-)