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In Reply to: RE: 44.1 kHz shown scientifically to be inadequate posted by Tony Lauck on July 26, 2009 at 19:26:14
that compared SACD/DVD-A/and redbook?
http://theaudiocritic.com/plog/index.php?op=ViewArticle&articleId=4&blogId=1
There findings are quite different than the musings here.
Follow Ups:
You might as well walk into a full church on X-mas eve and proclaim god's death long ago, or proclaim yourself a child molester at the entrance to Central Park. Both those options are easier.
FWIW, don't introduce any material here that uses scientific principles to test audiophile shibboleths. You'll be treated as if you just grew a bong out of the top of your head.
Thank you for the warning. I don't really have an opinion one way or another, it just seemed worthy of mentioning.
Problem is that the only accepted and allowable viewpoint is subjectivism; objectivism and scientific method is, perversely, ironically, treated as junk science, as voodoo, and with the same disdain as doggy doo on new Ferragamos. Thing is, objectivism and scientific approach has the potential to eradicate the subjectivists' reason for being--constant comparison of equipment, or audiophilia for equipment's sake. Pure subjectivists' primary public service is that they keep Audiogon full of fine gear at cheap prices.
I rarely, if ever, wade into any such debates; pointless.
Actually the Meyer/Moran work was heavily critised by objectivists, too.
But obviously for different reasons ;-)
bring bac k dynamic range
Kristan wrote:Problem is that the only accepted and allowable viewpoint is subjectivism; objectivism and scientific method is, perversely, ironically, treated as junk science . . .
The audio world is a bit like the “environmental” milieu except that the latter not only endorses “junk science” but lobbies politically on its behalf.
Common to both is that many who aggressively extol the supremacy of “scientific method” prove unable to subject their science to the most rudimentary scrutiny. They prefer to cherrypick papers on the basis of lightweight press reports and use them as intellectual cudgels.
I am wondering if this is the case with some posts in this thread.
deskducker wrote:
[Their] findings are quite different than the musings here.
I’ll reply, if I may, to Werner's points later but I’d like first to look at the Meyer & Moran paper as people are suggesting it refutes Kuchner’s work. (Werner reports that it has already been criticised by “objectivists” - whatever they are - but I haven’t seen that so apologies if I go over old ground.)
Sadly, (a) some of those who cite the paper appear not to have read it and (b) while it is on a related topic, it is nevertheless on a different one. Point (a) may explain why (b) gets missed.
The link that deskducker and Phelonious Ponk give is not to a journal paper but to a note on a blog about tests by “two veteran audio journalists who aren’t professional engineers” written by a third audio journalist, The Audio Critic editor, Peter Aczel.
As polemics go, it’s harmless, knockabout stuff but it tells you little about what the paper says and less about how to get hold of it.
Deskducker’s question, “Have any of you read the Audio Critic Double/Blind study?” is thus confusing: the answer can only be “No” because there isn't a pertinent study by The Audio Critic .
What there is is a September 2007 article in the Journal of the Audio Engineering Society written by AES members E Brad Meyer and David R Moran under the auspices of the Boston Audio Society (BAS) of which they are members and former officers. I eventually found their paper at:
http://drewdaniels.com/audible.pdf
The BAS has a follow-up note describing the test equipment used at:
http://www.bostonaudiosociety.org/explanation.htm
Kuchner’s papers report a five-year experimental program assessing “the human discriminability of temporal convolution” with tests that “employed either lowpass filtering or delays due to spatial misalignment” using “special ultrahigh-fidelity equipment” and showing “discernment at a ~ 5 microsecond timescale” which proved to be “much shorter than found previously”. (Editors demand this turgid stuff.)
Moran and Meyer, on the other hand, report “a series of double-blind tests comparing the analog output of high-resolution players playing high-resolution recordings with the same signal passed through a 16-bit/44.1-kHz ‘bottleneck’. [snip] The test results show that the CD-quality A/D/A loop was undetectable at normal-to-loud listening levels, by any of the subjects, on any of the playback systems.”
In short, their subjects were unable to distinguish the sound of SACDs played on a range of more or less high-end audio setups with the sound of the same SACDs after converting their output to 44.1 KHz using a “well-regarded professional CD recorder”.
However, unless I’ve missed something, their experimental procedure was so casual as to undermine the reliability of their results. That doesn’t, of course, mean that they were wrong, only that you can’t tell either way from their study.
Note that the paper was asking if SACDs sounded better than CDs, not whether 44.1 KHz recording could capture the finest nuances of “temporal convolution”. If it was, it would have had to demonstrate (as Kuchner did with his) that their apparatus was capable of reproducing such detail. It didn't do that.
1. In all, 554 trials were conducted but it is not reported how many subjects were used or how many trials each performed. Sixty were members of the Boston Audio Society but what proportion of the total they comprised is not known. They are likely to have known the authors. If the tests were non-judgmental and the subjects unaware of their purpose, this might be acceptable but neither condition is reported as being met.
It is the ABC of psychological experiment that subjects tend to provide “good” results and subject selection and procedure design must allow for that. This was not done.
2. The trials were conducted at four sites with different audio setups though most were done at the first, a moderately high-end domestic setup. Unfortunately, the first “well-regarded professional CD recorder” used proved to be faulty and the second, for whatever reason, unsuitable:
The first of these trials was done with the Pioneer player, and the fadeup of the room tone at the beginning of the Hartke disc revealed a slight but audible nonlinearity in its left channel decoder. We did some tests with the Sony, which sounded clean at any gain setting, and then switched to the Yamaha DVD-S1500, which was used for the remainder of the tests at this site.The text implies (it’s not clear) that the Pioneer was used for three months. It is not reported that results obtained with the faulty device were discarded or what proportion of the total they represented.
Details of the second site seem to have been lost:
I do not currently have a detailed equipment list for this venue, but the speakers were very large and capable high-end monitors, approximately 7 feet tall, and the power amps were sufficient to drive the speakers to very high levels without audible distortion.3. It is not clear that the test discs used actually reproduced the quality claimed for SACD (though the recordings were reported as musically very fine).
. . . we did a trial using the only disc we came across with an acoustic/electronic noise floor lower than our CD link [snip]. We advanced the gain beyond our nominal setting by 20 dB and used the player’s A-B repeat to loop through a short segment containing only room tone and a couple of extremely quiet musical notes. The noise of the CD loop was easily audible at the listening chair.4. Over twenty SACDs were used for the trials (some but not all are listed). Subjects could choose which tracks to listen to but the choices are not recorded.
Although the authors note that “The two variables that determine whether differences would emerge, in our experience, were the source material (of which very little is quiet enough) and the system gain”, there is not only no record of who listened to what or when but subjects were allowed to alter gain settings in ways that are not recorded.
5. Some trials were sighted, some unsighted but the procedure used is not recorded. There was no control set.
All of these are serious omissions but the last lot are showstoppers. At the end of his or her first year, a psychology student making such howlers might be asked tersely by their tutor, “Have you considered Media Studies at all?”
In summary:
1. The Meyer and Moran paper reports very sloppy experiments. It doesn’t follow that their thesis is wrong but only a lay reader would claim that they provide competent proof either way.
2. Even if they were robust, the results are irrelevant to an assessment of Kunchar’s work and most certainly do not refute it.
3. The point most at issue in this thread is whether Kunchar's results mean that 44.1 KHz recordings are (or are not) “perfect sound forever”. That is a proper debate to hold but Meyer and Moran’s paper has no more part to play in it than cheap abuse about dog dirt posturing as scientific insight. Let’s move on.
Dave
Edits: 08/01/09
"I’ll reply, if I may, to Werner's points later but I’d like first to look at the Meyer & Moran paper as people are suggesting it refutes Kuchner’s work. (Werner reports that it has already been criticised by “objectivists” - whatever they are - but I haven’t seen that so apologies if I go over old ground.)"
-- I'm not suggesting that the Meyer & Moran study refutes Kuchner's work, I'm suggesting it is relevant to the audibile quality of redbook vs. hi res, and that Kuchner's work is not.
The Meyer & Moran study conducted controlled listening tests using music, multiple reference audio systems and hundreds of trials with a variety of listeners, and concluded that the listners, who included audiophiles, mastering engineers, recording arts students, civilians, etc., could not hear the difference between hi-res files and exactly the same files run through A/D/A conversion and reduced to redbook specs.
Kuchner's study conducted controlled listening tests using two ribbon tweeters playing square waves, moved physically out of alignment with each other, and concluded that the listeners in that study could hear the physical alignment variations.
"Sadly, (a) some of those who cite the paper appear not to have read it and (b) while it is on a related topic, it is nevertheless on a different one. Point (a) may explain why (b) gets missed."
-- I have read it, I do understand that it is on a different topic. I also understand that the topic it addresses is related to the audibility of redbook vs hi res and Kuchner's study is not. Some people in this discussion, however, do not seem to understand that.
"The link that deskducker and Phelonious Ponk give is not to a journal paper but to a note on a blog about tests by “two veteran audio journalists who aren’t professional engineers”
-- I provided a link to the actual PDF of the paper in another post. Or you can search Meyer & Moran on Google and I'm sure you'll find it yourself. On the subject of redbook fidelity, I'll take the two journalist's study over Kurchner's work and the huge leap of faith and logic required to relate it to digital audio any day.
P
Phelonious Ponk wrote:
I'm sure you'll find it yourself.
Obviously, I did (eventually) as I posted a URL and quoted it at length.
On the subject of redbook fidelity, I'll take the two journalists' study over Kurchner's work . . . any day.
Such is, of course, your right but my scientific training convinced me many moons ago that an experiment with faulty kit, bad sampling technique, shoddy procedures and the like is inherently flawed. It is no more a partial truth than a near miss with a rifle is a partial hit.
. . . and the huge leap of faith and logic required to relate it to digital audio . . .
Maybe it is difficult for some though it seems fairly straightforward to me. Perhaps that's because it's my field. Whatever, Kurchner's experimental work is unarguably very fine and in a different league from Meyer & Moran's "study". We shall just have to disagree on how we interpret his results.
Meanwhile, to tidy up loose ends, if you explain to the Hi-Res aficionados on this list why they're deluding themselves, I'll drop a line to Sony's SACD people making it clear how they've been wasting their time. They'll appreciate it.
Dave
I'm not saying the high-res people were deluding themselves. That's not the point. The point is that Kuchner's study say absolutely nothing on the subject. I'm not sure where you get the faulty kit, bad sampling procedures, etc. There is nothing in the report that would lead to that conclusion. And you can, of course, choose to accept the "fine work" if you choose. It will, no doubt, be very useful the next time your setting up a system based on misaligned ribbons to play back your favorite square waves. It won't tell you much about the mysteries of music reproduction, though.
Audiophiles confuse me. "Trust your ears," they say. "Measurements aren't important." Until, of course, the most obscure and loosely relevant measurements appear to support their pre-conceived notions.
P
Phelonious Ponk wrote:
I'm not saying the high-res people were deluding themselves. That's not the point.
If Meyer and Moran are right to say that it is not possible to distinguish 44.1 KHz from media such as SACD, then the high-res wallahs are deluding themselves. I’ve yet to hear either SACD or 192/24 recordings but there have been enough reports from competent people on this list and elsewhere that there is a difference to prod me into examining the Meyer & Moran paper with a critical eye.
The point is that Kuchner's study say[s] absolutely nothing on the subject.
Kuchner demonstrated that humans can detect differences in percepts that cannot be resolved by 44.1 KHz recordings. That would affect inter alia the perception of timbre, especially on fast transients. Whether, in practice, on the kit most of us use, this matters (or even whether he is right) is a proper debate but it is nonsense to say his work is irrelevant. If, after all this debate, you can’t see that, we must simply agree to disagree.
I'm not sure where you get the faulty kit, bad sampling procedures, etc. There is nothing in the report that would lead to that conclusion.
There’s plenty - if you choose to look. An experienced eye helps but a willingness to scrutinise the text is a good start. The authors, whose integrity is not in question, deserve no less.
Faulty kit :- I got this from, well, Meyer and Moran’s description of faults in their kit. See text.
Bad sampling :- No details are provided of the size and composition of the subject group except that some were students on a recording course, others were “interested parties” and “about 60” were BAS members. The latter group especially was likely to be well aware of the views of Meyer and Moran on the issue of 44.1 KHz and “perfect sound forever”.
Competent research must therefore either exclude it from the trials or design (and report on) procedures to evade the problem. Given the circumstances, the first precaution might have been impracticable. The second is therefore critical. There is no report that either was taken so I concluded, not unreasonably, that neither was.
Shoddy procedures :- No details are provided of the procedures used though, as some tests were “sighted” and some “blind” and as there was no control set, this matters.
If Hi-Res playback precedes RBCD, the subject has prior knowledge of any difference. This will colour his/her perception. All we are given is a picture of an ABX box and “A-X-B” display. How they were used or even what they do, we are not told. There may be reasons for telling subjects which version is under test - but I’d like to know what they are.
They don’t give any so I concluded, again not unreasonably, that the authors (who report no pertinent experience in their biographies) were simply unaware of these very basic design issues.
The über-howler :- I am embarrassed to own up to this but I have only just noticed that Meyer and Moran obtained their 44.1 KHz signal by digitising the analogue output of their SACD player . (See “Tests”, para 1 and Fig 1).
They have just assumed that the DACs on the SACD players used are capable of resolving the differences between SACD and 44.1 KHz. Though they suspect the marketing claims of the SACD lobby, they take their kit on trust.
If the SACD players used cannot adequately resolve the signal, Meyer and Moran can run 1,000 subjects through 100 trails on each of 25 SACDs and do so sighted, unsighted and helped by any guide dogs in the BAS. They will never find a difference whether it’s there or not. The trials are not merely “flawed”, they are worthless.
In summary, it’s no wonder they didn’t find a difference: they used subjects likely to give them the answers they wanted, in all likelihood used procedures that did not prevent that and were, in any case, testing for the wrong thing. Otherwise, it’s fine work.
And you can, of course, choose to accept the "fine work" if you choose.
Thank you. I look forward to your critique of it. The only point I’d make is that he is a bit vague as to his subject base though he (a) evades bias by design and (b) reports that he adhered to campus selection guidelines so my beef would be pretty muted.
It will, no doubt, be very useful the next time your setting up a system based on misaligned ribbons to play back your favorite square waves. It won't tell you much about the mysteries of music reproduction, though.
No one has suggested I should use mis-aligned ribbons to set up my audio. I don’t use spectrum analysers, distortion meters and the like either but I’m glad that equipment designers do. Nor do I hire apparatus to determine the limits of my hearing but I’m glad that scientists do the work and that equipment designers read what they write.
There is much that is mysterious about hearing (and music) but to conclude from that that all of it is forever impenetrable is anti-scientific philistinism posing, in this case, as empirical rigour.
Kuncher demonstrated that the extreme limits of auditory perception were higher than previously thought due to the ear/brain’s ability to perceive the modulation of sounds by suprasonic frequencies (i.e. not themselves directly perceptible). Audiophiles have argued the point for decades, sometimes sensibly, sometimes not and many equipment designers have tacitly acknowledged it for almost as long.
Whether Kuchner proved his point using speaker alignment, low-pass filtering or a poker up his backside is irrelevant to how I set up my audio kit but it could influence the debate on using higher resolutions for distributing recordings.
Audiophiles confuse me. "Trust your ears," they say. "Measurements aren't important."
Audiophiles (and philistines) can say what they like. I’m writing as an experimental psychologist who would never say that measurements aren’t important, who has some knowledge as to how perception both deceives and informs and who gets angry at the energy with which ill-informed people persist in talking anti-scientific nonsense on both sides of the debate.
I have no personal interest in the outcome of the argument. My kit, which I’m stuck with for the mid- to long-term future, is aimed at the optimal reproduction of a large RBCD library; I don’t use upsampling and I can’t listen to hi-res recordings. But when a scientist peels back a corner, however small, of the “veil of mystery” that is great music, I’m intrigued.
I’m sorry that you and others can’t seem to appreciate what Kuncher has done and genuinely disappointed that you can’t debate the points at a higher level. That’s partly what forums like this ought to be all about.
Until, of course, the most obscure and loosely relevant measurements appear to support their pre-conceived notions.
If you find you’re in a hole, just stop digging.
Dave
"Kuchner demonstrated that humans can detect differences in percepts that cannot be resolved by 44.1 KHz recordings."
Please go back to the Kunchur papers. He did NOT demonstrate that.
He demonstrated two things about auditory perception, things that may
be new (then again, maybe not). However, there is no experimental
evidence, formal proof, or any founded reasoning in the papers that this
indicates a problem with 44.1kHz sampling. There is only, in the
conclusion of one paper, Kunchur's *claim* that this is so. That's all,
and that's *not* sufficient.
bring bac k dynamic range
werner wrote:Please go back to the Kunchur papers. He did NOT demonstrate that.
Thank you for the advice. By the same token, do please read what I wrote slightly more carefully.
Immediately after the phrase you quote (one, from a lengthy, perhaps over-lengthy, post), I said "Whether, in practice, on the kit most of us use, this matters (or even whether he is right) [emphasis added] is a proper debate".
By that I meant (and I'd have thought it pretty obvious though I apologise if it isn't), "I am not at this point saying either that he was right or that he was wrong - I am now dealing with other matters".
I've already said that I'd reply to your points later and, if you stop jumping down my throat, I will. I was suggesting here only that it was silly to say the Kunchur study is irrelevant.
But my post was essentially about the Meyer and Moran paper. Due to a sloppy oversight, I had previously argued that it was not totally worthless. Having "gone back to the paper" and spotted the howler I described, I now think it is.
Am I right or am I wrong? As the paper is regularly used to discredit Kunchur and undermines the case for publishing hi-res recordings, it matters.
Dave
Edits: 08/02/09
"Whether, in practice, on the kit most of us use, this matters (or even whether he is right) [emphasis added] is a proper debate".
We're slowly getting there, but i'd personally revise the above to say "whether, in practice, on any complete audio playback system, playing any actual music, this matters, is a proper debate."
P
Phelonious Ponk wrote:
We're slowly getting there . . .
As you wish. I've explained at (tedious) length why the Meyer and Moran paper is worthless. Am I wrong? If so, where?
Dave
Yes, you're wrong to argue that listening to square waves on misaligned ribbons is more relevant to the perception of music reproduction than listening to music unsighted, regardless of the tedium of your objections. You are going to great lengths to discredit, perhaps justifiably, a series of listening tests that attempted to get to the actual comparative quality of 44.1 vs hi-res, while putting forth as evidence a study that made no such attempt and has no relevance to the subject whatsoever. It is, as you say, "your field." Perhaps you're right that the Meyer and Moran paper is useless. I'm sure it is, at least, inconclusive. So we have a debate in which the only data is the inconclusive and the irrelevant. I think we're done.P
Edits: 08/03/09
You seem to have scant idea of how science works and less of how to participate in discussion so I see no further purpose in debating with you. But thanks for your time anyway.
To be fair, your inability to defend the Meyer & Moran paper helped me to see it as the pseudo-science it is and I'm grateful for that.
I could well be wrong in my reading of the Kuncher papers - but they are at least science (i.e. Truth or Error, not Ignorance, which is indifferent to both).
Dave
Snarky, but interesting response. I haven't attempted to defend the Meyer and Moran paper. I've merely used my total lack of understanding of science to conclude that regardless of the methodology, a study still needs to be related to the subject you're addressing if you expect your references to it to be considered relevant. And I'm pretty sure I'm not the only scientifically-ignorant participant here who has made that point.
Tim
To be relevant it has to be RELATABLE not necessarily explicitly related. He thinks it is and so do many others who've responded here. Others don't see it as actual evidence in regard to what it is being related to, which is fair but far from meaning it is not suggestive of greater discrimination by music listeners than what accords with the currently held limits. At the least, it makes hypothesizing about this greater discrimination reasonable.
I get that he, and a few others here, think it is related to the subject. But unless I misunderstand, the subject is the viability of 16/44.1, a digital recording format, relative to higher-resolution formats (or were we thinking that square waves are superior to Redbook?). And yet the study they think is not only related, but provides evidence of the unacceptability of 16/44.1, contained no 16/44.1 (or higher res) recordings, no complete speaker or playback systems, no music. Perhaps the audibility of misaligned ribbons playing square waves is related tangentially to digital timing in some theoretical way we non-scientists are incapable of grasping, but it says nothing about what people actually do or do not hear when listening to real music on real systems. I am, once again, a bit stunned that anyone can even argue that it does.
P
It says that there is some basis for thinking (hypothesizing) that people might be able to hear/distintuish more than what currently held beliefs about the limits of this ability are. It means that you cannot just dismiss claims by people who believe they, in fact, do. In other words, religoid dogma aside, it should humble those of your persuasion into saying that however unlikely it seems to you it just may be so but needs more valid demonstration and testing to establish whether in fact it is in the complex actual music listening situation.
No further humbling is necessary. No ribbons or square waves are needed either. And I don't even think it's particularly unlikely that hi-res files sound different than redbook (though I believe it is quite unlikely that the difference is anywhere near as dramatic as is often reported by breathless audiophiles following sighted personal listening "tests"). But I definitely think it needs more valid demonstration and (blind) testing to establish just what can be perceived in an actual music listening situation. On that, we completely agree.
P
I'll believe you need no more humbling when it's clear you don't respond to reports of hearing differences with a statement implying it just has to be some psychological distortion, placebo, or whatever. If you respond with 'maybe so, but we can't be sure until we test..." then you have the requisite humility. I'm not holding my breath.
Well, I'm trying, riboge, I'm trying. Sometimes it's difficult not to respond that way. I've been involved in audio - pro and domestic, personal and professional, for a long, long time and I am absolutely convinced that psychological bias is one of the strongest influences on perception of quality. I have seen much of the "high-end" abandon the pursuit of high fidelity in favor of jewel boxes filled with attractive colorations (or simply nothing special). And I have seen so many audiophiles follow them down that path throwing money after "upgrades" and "hearing" ever higher and higher levels of warmth, musicality, euphonics, PRaT and other terms they and their vendors have invented to substitute for real specifications and meaningful language. When a roomful of audiophiles hear huge, paradigm-shifting improvements from something that shouldn't make any difference and can't make much difference, I'm reminded of rooms full of engineers responding to things like the Aphex Aural Exciter. And they actually DID hear a big difference. They just needed more time to understand that it was not good.
But I'm trying, riboge, I'm trying.
P
Riboge wrote:
"Maybe so, but we can't be sure until we test ..."
A fair point but, though my mathematics would struggle, surely it's possible for those more numerate than me to say unequivocally whether the changes Kuncher has detected can or can't be resolved by 44.1 KHz sampling.
If they can, then it is a question of testing whether, in practice, they are. If they can't, there seems little purpose in testing anything though no doubt someone will demand it.
On other forums where the issue has been discussed (such as HydrogenAudio and Stereophile, with the latter as unpleasant as Werner reports) there are several posts saying "This man is wrong", "Call the firing squad", "Who does he think he is?" and all the usual tosh but I can't find any explaining why he is wrong. It seems to be taken as self-evident.
Any takers here? Did I miss something?
If the sums do suggest he is wrong, it should be possible, using his apparatus, to record test tones at 44.1 and, say, 192 KHz and see what listeners can or cannot detect at either sampling rate. Good Grief! - you could even use a double-blind test (assuming you knew how to design it).
BTW, I don't get Tony's argument that the tests should have used a frequency more in line with RBCD parameters. As I see it, Kunchur was researching human hearing, not RBCD technology, and presumably (and certainly should have) chose his tones accordingly.
That said, I agree that using more than one frequency could have provided useful data with the obvious hypothesis being that discrimination would tend to be more acute in the mid-band and taper off at the extremes. How feasible it would have been in practice I can't say.
Dave
" BTW, I don't get Tony's argument that the tests should have used a frequency more in line with RBCD parameters. As I see it, Kunchur was researching human hearing, not RBCD technology, and presumably (and certainly should have) chose his tones accordingly."
I made my comment in response to an AA poster, not as a criticism of Kunchur's work. As you correctly stated, Kunchur designed his experiments and reported his results to a scientific community interested in human hearing. Had he been an engineer rather than a scientist or had he been publishing in JAES then perhaps he would have designed the tests differently.
Kunchur's choice of a 7000 kHz square wave was no error. But he did make an error by jumping into a snake pit on the two forums. I doubt that he will repeat that mistake! There is little point in debating with a bunch of characters who criticize statements taken out of context and then go on to admit they haven't even read the scientific papers that were the substance of the work. Some of the people on those forums have the smarts and knowledge to appreciate Kunchur's work, but unfortunately they appear to be suffering from overpowering egos that prevent proper use of their mental faculties.
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
"BTW, I don't get Tony's argument that the tests should have used a frequency more in line with RBCD parameters. As I see it, Kunchur was researching human hearing, not RBCD technology, and presumably (and certainly should have) chose his tones accordingly."I agree and would add that the only real problem here is concluding what humans can hear when listening to music, based on a study of the theoretical limits of human hearing when listening to test tones.
P
Edits: 08/04/09 08/04/09
To further Tony's point, let me add that besides the breadth of what can be considered music in the continuum from a single tone to highly complex combinations of tones and rhythms, is the issue of training. While the complexity and possible aesthetic 'divertingness' of music might make it impossible for the untrained to do with music what they do with test tones, training might very well enable listeners to overcome this and eventually make the discriminations that tests with simple tones more readily demonstrate. That is one of the great shortcomings of most feasible double blind testing in this area, it is very difficult to control for the level of training and sophistication of the subjects. This is because we are not seeking to determine just what people DO distinguish but more importantly what they CAN distinguish. The maximum of what people can distinguish in music is only manifest with the most expensive training/experience along with the most innate talent for it. Each level of experience and of talent should be expected to have different results, no?
Sorry, I don't get it. I can see the point in training myself to recognize middle C, to hear and identify complex harmonies, to understand the pattern of a composition and be able to learn it by ear. I really don't see the point in training myself to hear the the otherwise inaudible limitations of recordings when I could, instead, be listening to the music. And if I did see the point and did find the time to train myself to hear such things, I don't know how that could possibly do anything to enhance my enjoyment of music. In fact, I think it would have the opposite effect.
And you're wrong about AB/X. Of course you can test the trained; it is no less feasible than testing the untrained and has been done often. The study I referred to in this thread used audiophiles, professional recording engineers, recording engineering students, and civilians. Sean Olive's studies at HK often use a similar cross section of listeners.
P
It's not that you SHOULD train yourself thus, it's that many audiophiles DO to one degree or another and as a result can distinguish what others don't. Some studies may have used more experienced people but that is a far cry from determining how trained each is and matching result to level of training. The point is that you doubt people distinguish with what you consider music what is distinguished with test tones. I say it depends on who, with what training, experience of that kind of music, etc, in addition to whether the test psychological set and conditions are conducive to it.
On a different level, you seem to foolishly claim you are the arbiter of what in the sound presented is the music and what is something else you find worthless or even negative. It's hard to see how hearing more can ever be worse than hearing less of what is presented. What you enjoy is an even more idiosyncratic matter. Some people enjoy the deafening most of all and dislike lower volumes that actually allow one to hear what's playing.
I think we're going to just have to agree to disagree. What you're talking about is, in my view, hand-picking your test subjects to pre-determine the result. And even though I believe you would fail to accomplish that goal, the goal is the antithesis of good research. And while I know you can train yourself to hear artifacts and distortions that usually go unnoticed, even by the most discriminating listeners, it doesn't just happen, it is an active process, and it is, in my opinion, the opposite of "audiophile" in the true sense of the word -- a lover of sound. It is crossing over to be a lover of gear, a lover of the technology that produces sound.
In my opinion.
And no, I don't consider myself an arbiter of what constitutes music and never implied that. We haven't discussed my opinions of music and what I think deserves that name in this thread or any other. We have only discussed what listening to physically misaligned tweeters playing square waves says about the relative quality of digital sampling rates when listening to music - type undefined - on an actual sound reproduction system.
I believe they are two very different things, that the former says little about the latter and that the introduction of actual systems, digital files and music into the testing would have completely changed the outcome, but we can agree to disagree on that as well.
P
I didn't speak of hand-picking subjects so much as noting differences among subjects or of matching the type of subject to what is being tested. If you are studying what people in general and on average hear, that's one thing. If you are testing about what people at the the upper limit can hear, then you should use subjects who are among the best at hearing the type of material you are presenting.Your distinction between square waves thru misaligned speakers and music on a 'real' system producing completely changed outcome is an hypothesis itself which hasn't been tested, and I don't see how it could be generally. First there is a lot of synthesized music that might have elements much like the square waves. That is, the distinction may be quantitative but surely undefinable qualitatively. And if you tested the difference with a square wave, you would not know how that pertained to one with other waveforms, frequencies or whatever versus music, to say nothing of how heterogeneous that latter category is. I can think of no basis for your belief in this 'distinction' other than desire to discount what is discordant to you.
But the most absurd assertion is that audiophile listening isn't or should not be an active process. Coupled with that is the non-credible claim that hearing artifacts and distortions that usually go unnoticed somehow is a different skill from hearing finer detail, subtle differences in timbre, etc, that also usually go unnoticed--especially if you try not to try to hear them which is what you seem to be advocating for audiophiles. I recall as a young music lover taking a similar position about listening to classical music. I believed bringing knowledge of music theory and music history to bear when listening interfered with a preferable direct, less active taking the music in holistically. I am taking it that this is what you are advocating in relation to trained listening with heightened perception of the finer aspects of the musical sound. The fallacy with this position of mine earlier or yours is that listening is ALWAYS AND IRREDUCIBLY AN ACTIVE INTERPRETIVE PROCESS (read Music and Memory by Snyder to find out about this.). The more you are trained and the more you know the better you do at this interpreting.
Edits: 08/06/09 08/06/09
I don't accept the distinction between music and test tones. If one can hear differences with test tones, then someone is going hear differences with music, even if it in a "Concerto for square waves and orchestra". We all have our tastes in music. :-)
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
I don't accept the distinction between music and test tones."
Ok then.
P
"to say unequivocally whether the changes Kuncher has detected can or can't be resolved by 44.1 KHz sampling."
Yes, they can be resolved.
The theoretical limit on the time resolution of a sampled quantised system is the sampling period divided by the number of quantisation steps (I may be off with a factor of 2).
Applications like cellphones, modems, and optical attitude sensors exploit this phenomenon.
"I don't get Tony's argument that the tests should have used a frequency more in line with RBCD parameters"
The tests were done with 7kHz square waves. Both Tony and I wonder about what would happen with 8kHz. For two reasons.
1) The third harmonic of 7k is 21k. If the auditory system is sensitive
(in some unknown way) to sound above its nominal cutoff frequency, then surely this 21k would figure heavily in the detection process. But the thing is: 21k can be passed by 44.1kHz sampled systems.
Re-doing the test with 8k would put the third at 24k, which cannot pass
a 44.1k system.
So the 7k test cannot prove (contrary to K's claim) that 44.1k is insufficient. On the other hand, if people pass an 8k test (which was not tried), and if the detection mechanism involves the third harmonic (which we don't know), then 44.1k is proven to be insufficient.
2) The first intermodulation product of 7k and 21k is 14kHz. Many people still hear 14k. The first intermod prod of 8k and 24k is 16k. Many people don't hear 16k anymore. If the auditory system uses the level and phase of this intermod product for detecting the differences experimentally tested by K, then the outcome of such test would differ drastically between 7k and 8k.
I hope this clarifies things a bit.
bring bac k dynamic range
werner wrote:I hope this clarifies things a bit.
Thanks for your note and sorry for the late reply but, no, I’m afraid it doesn’t entirely clarify things for me.
Yes, they can be resolved . . . The theoretical limit on the time resolution of a sampled quantised system is the sampling period divided by the number of quantisation steps (I may be off with a factor of 2).
Unfortunately, I cannot translate that into the “μsecs” on which Kunchur bases his argument. Can you help?
The third harmonic of 7k is 21k. If the auditory system is sensitive (in some unknown way) to sound above its nominal cutoff frequency, then surely this 21k would figure heavily in the detection process. But the thing is: 21k can be passed by 44.1kHz sampled systems.
1. Your argument seems to assume that RBCD recordings are made to (and DACs operate at) these theoretical limits. This seems optimistic: as I understand it, upper limits are, in practice, significantly lower than 22 KHz because the Nyquist filters are not, by design, “brickwall”.
In other words, are you sure that 44.1 KHz systems can, and in practice do , resolve (pass) 21 KHz? If they do, your point is fair. If they don’t, it isn’t. My guess is that they very often don’t.
2. In any case, Kunchur’s data seem to have undermined the argument already.
In his 2008 paper ( Temporal resolution . . . ), he measured the HF threshold of a group of subjects with the two best coming in at 17.7 and 17.8 KHz. The poorest, on the other hand, only got to 9.4 KHz and would thus have been struggling a bit with the 8 KHz tone, never mind overtones (see below).
In spite of that, he/she scored 100 per cent for t = 5.6 μsec, the second lowest value to provide a significant result over the group as a whole. (See Experiment 1 and Table 3.) In other words, he/she was pretty well up there with the best of them in spite of significant hearing loss. That suggests, and Kunchur’s text makes it explicit, that the results are not down simply to the audibility of harmonics.
So the 7k test cannot prove (contrary to K's claim) that 44.1k is insufficient. On the other hand, if people pass an 8k test (which was not tried), and if the detection mechanism involves the third harmonic (which we don't know), then 44.1k is proven to be insufficient.
As I read the text, Kunchur demonstrated that all the harmonics of the test tone were inaudible (Table 1 & p 597).
2) The first intermodulation product of 7k and 21k is 14kHz. Many people still hear 14k. The first intermod prod of 8k and 24k is 16k. Many people don't hear 16k anymore. If the auditory system uses the level and phase of this intermod product for detecting the differences experimentally tested by K, then the outcome of such test would differ drastically between 7k and 8k.
As I have explained, this argument does not seem to accord with the data.
If you were to say that Kunchur makes a conceptual leap when he argues that his experiment conclusively proves that 44.1 KHz sampling rates are inadequate, I’d probably agree with you. But the argument that using an 8 KHz test tone in preference to a 7 KHz one would have proved the argument unequivocally does not, it seems to me, hold much water.
And if it does, it leaves the task of explaining why so many competent observers are adamant that higher resolutions provide better reproduction. (Higher resolutions for recording is, of course, a different thing.)
Dave
Edits: 08/05/09 08/05/09
"Unfortunately, I cannot translate that into the “μsecs” on which Kunchur bases his argument"
The (theoretical) temporal resolution of a 16-bit 44.1kHz sampled system is 22us/2^16 = 346 picoseconds. In practice it is bound to be worse, but still vastly better than 5 microseconds. What this means is that if you have two waveforms that are identical except for a temporal feature of, say, 5us, that the sampled system maintains this temporal feature, even when other aspects of the waveforms, like their bandwidth, are obviously modified when passing through the sampled system. Or even simpler: take a waveform (A), then copy it with a delay of 5 us (B). Then sampled and decode them. What you end up with are two waveforms than are delayed 5us relative to each other.
Or in the specific (but absurd) case of two fast pulses separated by 5us versus one single pulse of the same total energy, the post-sampling waveforms would 1) merge the two pulses (obviously), but 2) still keep a spectral distinction between the two signals as such (so that they can be recognised).
In satellite attitude control this principle allows digital cameras (i.e. spatial sampling) to determine the position of stars with an accuracy that is much much smaller than the image sensor's pixel pitch. That's the one non-audio application I'm quite familiar with. There are others in cellphones and modems, which are after all pretty scary sampled systems, but then I'm not into RF at all...
"This seems optimistic: as I understand it, upper limits are, in practice, significantly lower than 22 KHz because the Nyquist filters are not, by design, “brickwall”."
Digital filters can be arbitrarily brickwall, especially in software sample rate convertors. But that's even not required: in ADC and DAC chips the filter type most often used (for reasons of economy) is the half-band FIR. These have a -6dB point at 22.05kHz for CD. Their response at 21kHz is between 0 and -6dB, so almost level. So yes, CD systems can and do pass 21kHz with quite some ease.
"The poorest, on the other hand, only got to 9.4 KHz and would thus have been struggling a bit with the 8 KHz tone, "
That one 9.4kHz person is indeed something of a sore thumb. I discussed that case already at Hydrogen Audio.
BTW a limit of 9.4k does not imply struggling at 8k. Hearing is pretty much on/off at these elevated frequencies: the cutoff is very abrupt.
"As I read the text, Kunchur demonstrated that all the harmonics of the test tone were inaudible (Table 1 & p 597)."
As pure tones. But fact remains that the subjects did distinguish between the two test signals, and that if the discrimination is not based on level differences of the fundamental, then it *must* be based on something else, which leads to the lowest-order non-linear distortion products of the signals involved, namely the second harmonic of the fundamental, and the first intermodulation between the fundamental and its third. That's all there could be for the ear to work with ...
"If you were to say that Kunchur makes a conceptual leap when he argues that his experiment conclusively proves that 44.1 KHz sampling rates are inadequate, I’d probably agree with you."
Well, that's what I said at the start of this thread.
bring bac k dynamic range
werner argued that:
[the changes Kunchur has detected] can be resolved [by 44.1 KHz sampling].
The theoretical limit on the time resolution of a sampled quantised system is the sampling period divided by the number of quantisation steps . . .
Again, sorry for a late reply. I’m well out of my field here and haven’t found much by way of useful background. I’m not saying you’re wrong only that what you say seems directly to contradict what Kunchur says. See e.g. his “FAQ”:
Optical example: A binary star system is imaged through a telescope with a CCD. First, there is the analog optical resolution that is available, which depends on the objective diameter, the figure (optical correctness) of the optics, and seeing (atmospheric steadiness). This optical resolution is analogous to the "analog bandwidth". Because this resolution is limited, a point source becomes spread out into a fuzzy spot with an intensity profile governed by the point spread function or (PSF).
Next we are concerned with the density of pixels in the CCD. To avoid aliasing, the pixel spacing L must be finer than the optical resolution so that the optics provides "low pass filtering". If the pixels and their separation are larger than the separation of the centers of the two star images, the two stars will not be resolved separately and will appear as a single larger merged spot. In this case the essential feature (the fact that there are two separate stars and not an oblong nebula) has been destroyed. This is usually what is meant by "resolution" or the lack of it.
The number of bits N that can differentiate shades of intensity ("vertical resolution") has little to do with this – no number of vertical bits can undo the damage. However, details of the fuzzy merger do indeed depend on N: if the star images are moved closer together, the digital data of the sampled image will be different as long as the image shift exceeds L/N. This L/N definition of resolution applies to the coding itself and not to the system's ability to resolve essential features in the signal as described above . . .
and, in the next section (Digital Audio Recording):
However this lack of temporal resolution regarding the acoustic signal transmission should not be confused with the coding resolution of the digitizer, which is given by 23 microseconds/2^16 = 346 picoseconds. This latter quantity has no direct bearing on the system's ability to separate and keep distinct two nearby peaks and hence to preserve the details of musical sounds.
[Emphasis added - DB] You, OTOH, say that The (theoretical) temporal resolution of a 16-bit 44.1kHz sampled system is 22us/2^16 = 346 picoseconds .
I accept that Kunchur seems to confuse the notion of distinct temporal events with that of temporal resolution as investigated in his experiments and that that is unhelpful - but I don’t see it as critical to the argument.
What is more clear to me is that he says that the sampling rate is the sole critical parameter whereas you say that it is the sampling rate divided by the bit depth.
The two formulae cannot both be right - and the difference (here, nearly four orders of magnitude) is the nub of the argument as to whether RBCD is "perfect sound".
Comments?
Dave
Oh, Kunchur agrees:
"details of the fuzzy merger do indeed depend on N: if the star images are moved closer together, the digital data of the sampled image will be different as long as the image shift exceeds L/N"
Which just says that depending on the ADC resolution a sampled system can encode spatio (or temporal) features smaller than the pixel pitch (or sample period).
But then come:
"the two stars will not be resolved separately and will appear as a single larger merged spot. In this case the essential feature (the fact that there are two separate stars and not an oblong nebula) has been destroyed."
"applies to the coding itself and not to the system's ability to resolve essential features in the signal as described above "
"This latter quantity has no direct bearing on the system's ability to separate and keep distinct two nearby peaks and hence to preserve the details of musical sounds."
Above three lines state as much times that according to Kunchur it is mandatory that a system preserves "essential features", and what are these features? Well: keeping two 5-us-separated impulses separate.
That's what K claims: that an audio system has to pass two impulses less than 5 us apart.
Where is the proof that such is required for audible transparancy?
There is none. All we get are the experiments followed by an "and hence ...".
By the way: the optical analogy Kunchur cites is only a valid analogy to CD audio when the pixel display is viewed from a distance so that the
angle subtended by a single pixel is slightly smaller than half of
the angular resolution of the human eye. Only this reflects fully the
case of a 22kHz-limited system with an assumed <20kHz aural bandwidth.
And you know what? Under such circumstances would the eye merge the
two stars somewhat before the camera+display would.
But I'm fed up with this. Let this thread die. If anything more comes of it then I'm sure it will be posted at Hydrogen.
bring bac k dynamic range
"I rarely, if ever, wade into any such debates; pointless."
Ah...then you are a wiser man than I.
P
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