How Science Got Sound Wrong

It is an interesting article, and I certainly will not fault his credentials w.r.t neurobiology, though it sounds like his knowledge w.r.t. the auditory processing system is 2-skin layers deep but no doubt still deeper than mine. But, even that I will not fault.

What I will fault is his knowledge of signal processing and how that relates to analog/digital conversion and analog signal reconstruction. He seems to process the same limitations in his knowledge as Teo_Audio illustrates above with his record example, that Millercarbon alludes to, and whoever did not calculation w.r.t. bandwidth.

I will start off with the usual example. Records, almost all of them made in the last 2 decades (and longer) were recorded, mixed, mastered on digital recording and processing systems. Therefore, whatever disadvantages you think apply to digital systems w.r.t. this timing "thing" absolutely and unequivocally apply to records recorded in digital.

So back to the paper, Teo’s error in logic / knowledge, miller’s interpretation. The most recent research shows that us lowly humans can time the difference of arrival of a signal to each ear to about 5-10 micro-seconds. Using that mainly, and other information, we can place the angle of something in front of us to about 1 degree. 5usec =~1.5mm of travel. Divide the circumference of the head by 1.5mm and you get about 360, or 1 degree of resolution. Following?

So how does the brain measure this timing? By the latest research, it appears to have 2 mechanisms, one, that works on higher frequencies, higher than the wavelength of the head’s size, that is based on group delay / correlation, i.e. the brain can match the same signal arriving to both ears and time the difference and another mechanism for lower frequencies, that can detect phase, likely by a simple comparator and timing mechanism. The two overlap. Still following? You will not this happens with relatively low frequencies, i.e. still frequencies within the range identified for human hearing. I know know ... but the timing, what about the timing. So let’s talk about that.

First a statement: In a bandwidth limited system (as digital audio systems are), any signal on those two (or more) channels will be time accurate to the jitter and SNR limit of the system, and NOT the sampling rate. Let me state that another way. Any difference in timing captured by a digital audio system, assuming the signal is within the frequency limits of that system, will be captured. Let me state that a 3rd way with an example. We have a 96KHz ADC with 10 pico-second jitter. We have two identical signals, bandwidth limited to say 10Khz. One signal arrives at the first ADC 1-microsecond before it arrives at the other ADC. We then store it and play it back. What will we get? ... We will get 2 signals, essentially exactly the same, with one signal delayed by 1-microsecond.

So, all those arguments the neurobiologist made in that extensive article, all his knowledge, are all for naught because he does not understand digital signal processing and ADC systems and analog reconstruction. If he did, he would have known that digital audio systems, within the limits of bandwidth, are not limited in inter-channel timing accuracy to the sample rate, but to the jitter. Whether the signals leave both channels at time A or time B does not matter, as long as the relationship in timing between the two channel is accurate .... which it is in digital analog systems.

.... and if you are reading this GK, not once did I need to consult wikipedia :-) ...

terry9,

Are you familiar with Shannon-Nyquist theorem? I provided the rather long-winded wikipedia article link below.

In a bandwidth limited system, if the sampling rate is 2x the bandwidth, you can capture all the information, including relatively timing information. I.e. with a 100KHz sample rate, you can capture everything in up to a 50Khz bandwidth limited signal. For practical reasons of analog filters, you normally want to sample 4x or more the target analog bandwidth so by 1/2 the sample rate there is no more signal.

Within the realm of signal capture and reconstruction, I would consider this established fact, though many, without the requisite knowledge, do not understand (or at least accept) the premise.
https://en.wikipedia.org/wiki/Nyquist%E2%80%93Shannon_sampling_theorem

terry9,   No worries on being confused about this. I find that many audio writers, many people in the audio industry period, and certain many (most) on audio forums do not get this concept. When you do the math (no literally go through the math), which I have not done in years, it becomes quite obvious how it works (after the 3rd of 4th reading).

Let me do a more real world signal. We have a 24 bit audio system, so it captures with a resolution of about 1/16.7 million, though practically will be closer to 1/1-2 million. Let’s say the system is sampling at 100Khz, and the system is bandwidth limited to 20KHz. Now let’s say we have 10KHz signal.

One key concept in a bandwidth limited system is that you cannot have just a pulse 1 waveform long, i.e. you can’t have a 1Khz waveform that last exactly 1 cycle. That would violate the bandwidth of the system because in a bandwidth limited system you cannot start and stop instantly. You can’t start and stop instantly in the real world either.

Here is where it gets harder. So these two signals, both 1KHz tones, separated by 1 microsecond arrived at these two ADCs. Let’s assume that Signal B arrives at Channel 2, 1 microsecond before Signal A arrives at Channel 1. To make the math easy for me, let’s assume that Signal A arrives at exactly 0 phase. Here are the digital outputs for the first 10 samples at 1KH and 20KHz. This is a DC offset AC signal, so the numbers go from 1 to 2^24.

You can easily tell these numbers do not represent the same signal, there is definitely something different about them. Your next question may be about accuracy / resolution. Jitter will obviously impact the inter-channel timing accuracy. I have not looked at the math in a while, but as you approach the SNR, I remember there is an increase in the inter-channel timing uncertainty.

• 1KHZ
  
• Ch1 / Ch2
• 8,388,608 / 8,441,314
• 8,915,333 / 8,967,925
• 9,439,979 / 9,492,249
• 9,960,476 / 10,012,218
• 10,474,769 / 10,525,779
• 10,980,830 / 11,030,906
• 11,476,660 / 11,525,605
• 11,960,303 / 12,007,923
• 12,429,850 / 12,475,958
• 12,883,448 / 12,927,862
  

We can do it at 20Khz as well

• Ch1 / Ch2
• 8,388,608 / 9,439,979
• 16,366,648 / 16,628,630
• 13,319,308 / 12,429,850
• 3,457,907 / 2,646,210
• 410,567 / 798,368
• 8,388,607 / 9,439,979
• 16,366,648 / 16,628,630
• 13,319,308 / 12,429,850
• 3,457,907 / 2,646,210
• 410,567 / 798,368

This is 20KHz, 90db down from full. As you can see, there are still substantial differences between the channels. This is 20-30db above the noise floor of a good ADC.

• Ch1 / Ch2
• 265 / 298
• 281 / 314
• 298 / 331
  
• 314 / 347
• 331 / 362
• 347 / 378
• 362 / 393
• 378 / 407
• 393 / 421
• 407 / 434

terry91,067 posts11-15-2019 12:29amThe nature of the signals is irrelevant. It is the relative timing of the encoding that matters. If the sampling rate is not high enough, or the jitter rate not low enough, then two signals differing by 1 microsecond will be encoded as identical.

Perhaps an example will help you to understand my confusion. It seems to me that if sampling is done at a frequency of 1Hz, and two signals differing by 1 us are detected, they will be encoded in the same pulse about 999,999 times out of 1,000,000. Which logically implies that sampling rate is intrinsic to the issue.

Perhaps you could point out the source of my confusion.

A system, with a 20Khz bandwidth, can still respond/detect a signal in microseconds. A real world impulse may last only a few microseconds, however, as your ear is bandwidth limited, you won't perceive an impulse as only lasting a few microseconds, you will perceive it lasting 10's of microseconds or longer, just as when you hit a woofer with a short impulse, it does not stop moving at the end of the impulse.  In those respects, you can reconcile both a 20khz bandwidth limited system and microsecond impulse timing.

akgwhiz1 posts11-15-2019 12:32amNew here but I found his points, yes his science, very intriguing. To the point I thought, heck, hes got it right. But I cant help but wonder, even a fully digital stream/source/path ultimately has to be reproduced through a vibrating speaker. It seems that this is a massive integration or smoothing, each connected (albeit complex) peak and trough lasting way longer than the neural timing. Accepting his points, maybe this is digital's way to get by as well as it does. BTW, I'm not picking sides, just the way I stated it.  

terry9,

Excellent catch on infinite series, but also easily addressed. As we are dealing with audio, there is effectively no information below 10Hz, and some would argue 20, but let’s say 10Hz. For that reason, any real single data set, i.e. a song file, can be modelled as an infinite series as there is a maximum rise time and minimum fall time at beginning and end, hence you can "set" all data outside to 0 (whatever your 0 is) for all points when applying the theorem. Any "errors" in bit level would be in the silence at the beginning and end of the track. In some ways, this is like a natural windowing function.

There are lots of papers, proofs, course books, material, etc. that goes into detail, including size of error when you don’t have an infinite series, which in a practical audio case, would be much smaller than other error sources.

If you want to play with "math", GNU Octave is a free-ware version of Mathcad (not as graphical) and would let you simulate any of these concepts.

I had Mathcad on the brain as I use it pretty regularly. GNU Octave is a freeware version of Matlab, not Mathcad.

The problem is GK, is that he is Not a world expert, not even remotely on the underlying topic of this whole article. He is an expert on physics and and neurobiology. He is absolutely not an expert on digitization, digital signal processing and reconstruction. Everything he says about human hearing and perception we can assume is 100% right and it makes no difference as the whole premise of his article is underlying flaws in timing in a multi channel audio system that frankly are not there. No expert in signal processing would have ever made the fundamental flaw(s) he did.


I find it disappointing that once again you have made posts that carry absolutely no relevance or information and add nothing to the discussion but appear to be only attempts to hear yourself talk. Feel free to use your obviously extensive free time to find a scientifically relevant paper ( i.e. something published and reviewed) that shows what I said to be false. If you can’t do that, then please go troll elsewhere. There are people here that actually want to learn.

Your brain measures the phase difference of the same signal reaching both ears. Errors in phase of a sound primarily from one speaker will not affect the measurements. Errors in phase from a sound from both speakers only matters if the phase shift is significantly different between the speakers, I.e. mfg variation.


And yes it is a red herring that keeps being raised by people who don’t understand how how digitization and analog reconstruction works and the math behind it.

terry9,

Here is a paper, it is about ultrasonic imaging, but that is simply a scaling issue w.r.t. frequency. It has nice graphs that clearly illustrate the ability to extract timing information and shows them as a function of sample rate. Even at a relatively low SNR, 30db, the error in extracting timing is very small. The oversampling in this case is 20x the frequency of the waveform:  https://www.diva-portal.org/smash/get/diva2:995652/FULLTEXT01.pdf

Once again, GeoffKait enters the argument, makes personal insults and jokes, adds absolutely Nothing to the argument, and hijacks the thread making it useless. Unlike Geoffkait’s posts, which are nothing but personal attacks, deflection, and obtuse comments, mine are filled with real information, directly related to the post, and I even provided links to relevant information and downloadable experiments that can be run that show exactly what I am claiming. If you have any, I mean any value at all to add to this thread, you would disprove well anything that I have written .... but no, just more personal attacks because you have ... nothing.   This discussion and the basic premise of the article have little to do with "sound" at all, but whether a digitized system has relative sub-sample timing information. That you attempt to make it about "sound" shows you don't understand the premise of the article and were confused by the title.

Is There A Moderator In The House

terry9, here is an example that someone created that shows an example of what I am discussing w.r.t. subsample timing:.https://www.dsprelated.com/showcode/207.php  


https://www.dsprelated.com/showarticle/26.php


This is fairly simple paper that looks at impacts of noise and distortion on time measurements:.   
https://www.google.com/url?sa=t&source=web&rct=j&url=http://www.ajer.org/papers/v4(04)/S...


There are literally thousands of articles and papers on subsample timing measurement.

It's a full time job keeping up with the misinformation being spread :-)

Thank you for further contributing to the spreading of misinformation clearthink. When You can prove anything that I have said is wrong, instead of just throwing ad-hominems, maybe I will take your posts seriously. Until then, you are just bloviating in the wind.

It’s really weird why so many people fight so hard to discredit someone, and spend so little time discrediting what they say. Is that because you can’t?  I really must think that is it.

clearthink971 posts11-18-2019 1:46pm
atdavid"It’s a full time job keeping up with the misinformation being spread "

With more than 375 posts in less than 30 days of membership hear you are doing an excellent job of contributing to the misinformation even if you are correct about 15% of the time.

ahofer,

To be clear, you mean the assertions in the linked article are not remotely accurate?  I only ask that because people will read only the few words you wrote and hear what they want to think ;-)


I looked up the author and sent an email directly showing where he was wrong. I also invited a former colleague to contact him/the website as he has more academic clout than I do. Nice, but unfortunate that I am not the only one that saw through the fallacies of the article.

Okay hotshot, Mr. "clear"think.  I am correct about 15% of the time huh?   That is called an ad hominem. Look it up if you don't understand what the word means. It is an ad hominem as it is an attack on me, but not on anything that I have actually posted. You will be hard pressed to show even once where I am incorrect, let alone 85% of my posts. I will never claim to be perfect, but I know what I know.   Oh, and please do learn what the word, "Evidence" means before trying to make some silly claims of where I am wrong. 

clearthink972 posts11-18-2019 3:53pm
"You have repeatedly and consistently shown to be factually wrong and materially deceptive in many posts. You have only been hear a short while so it is easy to review your posts and identify those where you are in error I understand why you are uncomfortable with you’re mistakes being so frequently disproven and rendered uselss but as I kindly pointed out you are correct about 15% of the time!

See what I mean. Even when faced with absolute evidence, that he does not have a degree in theoretical physics, he triples down and tries to make excuses instead of admitting error. Geoff may be the only person on the planet that claims their Batchelor's level aerospace engineering degree is a degree in theoretical physics .... I am thinking the Real theoretical physicists would take exception with that. Last time 3 of the 4 courses he listed were just typical undegrad level engineering courses. You would think someone claiming to be smart would know the difference between taking electives and having a degree in something. Are you ready for quadrupling down? Get out the popcorn, this should be good!

geoffkait18,341 posts11-18-2019 6:21pm
Aerospace Engineering curriculum IS theoretical physics, Mr. Bluster. Remember? Theoretical propulsion, theoretical fluid dynamics, statistical thermodynamics, things of that nature. I can see stream coming out of your nose again, Mr. Bloviator in Chief. 😤

Thanks you for linking that ahofer, I was rather appalled at the article, especially from someone espousing expertise, but the whole premise of the article is obviously outside his area of the article. Unfortunately, as you can see on this thread, while certainly there are people who understand this, and others who want to learn, others are just content to throw ad-hominems at anyone who pokes their bubble.

Here is an example of being wrong. "Geoffkait claims he is a theoretical physicist .. it’s right in these forums. Then geoffkait claims he never said he had a degree in theoretical physics". atdavid proves that geoffkait actually has a degree in aeronautical engineering (check UVA records), then proves geoffkait actually wrote that he had a degree in theoretical physics (he wrote that in a forum). See, that is called addressing the content and proving a claim to be wrong.


atdavid makes a "claim". geoffkait’s response, "you’re wrong" and "you cut and paste from wikipedia", etc. No proof that atdavid’s claims are wrong, and usually an ad-hominem.


Notice a difference?  geoffkait and a clique of others like to tell me (and others) that they are wrong, but unsurprisingly, can never show how we are wrong.

And even when faced with overwhelming evidence of being wrong, the clique will never admit it and just doubles down on lies ...

English majors vs Tweakers

just another example from Jgossman of the problem with English majors getting into the fray., at least from a technical standpoint. I have a degree in theoretical physics (fluid dynamics and propulsion) from the University of Virginia, actually now that I think about it I accumulated the most credits ever recorded by an undergraduate, 203. I was selected to present my undergraduate thesis to the AIAA national conference on a design of a low thrust rocket engine for interplanetary space travel using highly magnetized metal crystal bombarded by highly accelerated Xenon ions. I designed the FAA's first satellite system twenty five years ago. i wrote the definitive explanation for how the intelligent chip works quantum mechanically ten years ago and have been designing quantum chips for many years. However, I can certainly understand how English majors would be rubbed the wrong way.

Cheers,

geoff Kait
Machina Dynamica

• Log in or register to post comments

September 28, 2014 - 7:34am

Long after the game is over, the stands are clear, and the lights are turned off, geoff will still be screaming at the empty umpires chair about the call that the video replay clearly shows was out. 

ieales486 posts11-18-2019 6:19pmGame, set and match

How can he be on to something if the whole premise of his article is wrong?

geoffkait18,360 posts11-19-2019 6:12amFrankly I think the author might be onto something and I’m only judging by what he wrote in some other articles I located somewhere in cyberspace, including this excerpt from one of them. It just sounds right.

The amount of effort some people will put into supporting their delusions and dishonesty is really astounding. aerospace ENGineering. Batchelor's level.  The basis of most of semiconductor physics is in theoretical physics ... likely far more than aerospace engineering. That does not mean I studied "theoretical physics", nor does it mean that I have a degree in "theoretical physics". There is a big difference between applying concepts of theoretical physics, which while theoretical, provide more than enough accuracy for engineering tasks, and being an actual theoretical physicist (i.e. the person working on those underlying theories).


For quadrupling down gk, without providing anything viable to refute your claim to be a theoretical physicist, nor to refute you claiming to have a degree in theoretical physics, all you have proven ... is what is wrong with many audiophiles.

Ms K. seems to quote Wikipedia, word for word, several times a day. I think they protest too much.

Sorrry artemus_5, you did post a very valid question, and unfortunately I got carried away responding to ad hominem attacks and not addressing your question. I reposted what I think was your question below. For my answer, I want to define "Information", i.e. what the recording engineer intended to commit to media, whether vinyl, CD, streaming whatever, and "sound" namely what comes out of the speakers and gets to our ears, and perception ... how we interpret what reaches our ears. From that:

• Unintentional colorations of the "information" that may result in a sound we like (or don’t like), i.e. cross-talk on vinyl, aliased noise on a NOS-DAC, certain harmonic distortions, etc.
• Intentional colorations of the "information", i.e. intentional voicing of the frequency response wherever that happens.
• Unintentional colorations of the "information", that we typically don’t like, like IMD in poorly implemented solid-state amplifiers, dielectric absorption cross-over capacitors.

I see it repeated regularly that we can’t measure everything. There are three aspects of that in audio, going back to my premise, information, and sound. We obviously cannot measure all aspects of perception, we are all built differently so the best we could hope is to measure individuals. Even measuring the sound that reaches the ear we cannot, or at least from a practical aspect cannot measure "everything". We would need to measure the incoming sound arriving in a spherical pattern at the ear. However, information, i.e. how well the original "signal" is faithfully reproduced in the digital/electrical domain, we can measure with great accuracy (think of all the amazing measurements we can make today).

So, in a long about way, we can objectively and quantifiably say whether "something" provides a more true communication of information, but what we can’t say is whether you will prefer that truer communication or not. I really don’t understand the heated argument w.r.t. CD or Vinyl, some prefer one, some prefer the other. Who cares. Where it gets dicey is when claims about their "information" ability get tossed around willy nilly .... which leads to all the mud-slinging w.r.t. things like cable and interconnects, things that directly impact information, something we can measure very very well, especially when you take the next point into account ....


Adding a final thought, there has been a lot of work done in the fields of psychoacoustics, neuroscience, neurophysics, neurobiology, etc. that has put bounds around "audible". i.e. we have not been able to show conclusively any audibility beyond about 20KHz. There were some experiments that suggested we may "perceive" higher frequencies, but later it was shown it was as likely there was harmonic modulation either in the transducer or the environment. Unfortunately, you also see misapplication of studies, a common one being that since we can detect arrival times between the ears to the microsecond level, then we must be able to detect frequencies beyond 20Khz, and/or we need >44.1KHz sample rate, neither of which is true. The other misuse of this fact is to suggest phase response is super important, but this properties is the ability to measure the time difference of a signal, any signal, arriving to two ears. Screwing up the phase does not matter, though consistency between left and right channels on the screw up would be important. If the "bound" is felt to be fuzzy, we can always add tolerance when comparing it to "information". I.e. if we are never able to show 0.1% THD is audible, under any testable condition, then a device that has 0.01% THD or lower is likely inaudible. Similarly, i.e. we cannot detect a 1db difference in level at 15Khz, then any product making a less than 0.1db difference is likely to be inaudible. There are many similar measures.

The biggest question I have is this. How can an objective quantitative answer be given to such a subjective subject as music, its reproduction and one’s interpretation of what they hear? Oh sure, we can give some ideas or thoughts about it. But our knowledge only goes so deep. One may look at figures and speculate what should be heard. But can we absolutely know what IS heard by 100 different people listening to the same music on the same equipment? I don’t think so. My $.02 worth.

Timing resolution of digitized bandwidth limited signals.

The author came up with an explanation that was wrong by trying to assert expertise in an area he does not have. In many ways that is worse than creating no explanation at all, because it deflects from understanding possible real reasons ... just look at this thread, and all the people trying to grasp onto an explanation that is simply wrong. What value is there in that ???


Many people prefer vinyl (even though much of the time the source material was digital). Many people prefer all digital chains.  It is a good thought exercise to try to determine if anything is wrong with digital, if you stick to things that are factual.  It is an equally good thought exercise to determine what is "wrong" with vinyl and what makes that sound attractive to many.

taras22301 posts11-21-2019 8:12amActually there is a problem for many people....for them digital does not sound as good as analog, and highly compressed is even worse. And the author was taking a shot at explaining that. And who knows he may not have explained himself adequately but that doesn’t mean that problem does not exist.

Nope, an article by an actual scientist ... just not an actual scientist on the technology most applicable to the article.

mapman16,341 posts11-20-2019 8:22pm

How Science Got Sound Wrong

Let me guess.  People who know nothing about science trying to take over?

The author makes a claim w.r.t. timing of digitized signals, i.e. the timing limitation is the sampling rate, that is not at all accurate for a bandwidth limited signal.


The author creates a problem that literally does not exist .... there is no problem.

taras22300 posts11-21-2019 7:30amSo what exactly is the problem that this technology is being applied to?

Cool, I also participate on DIYAudio and AudioScience, and you know what, there are many many people on both those forums who would never make the mistakes that the author of this article and you have made.

I provided several links in this forum that prove what I am saying is true. I could provide many many more. Your name and background shows you clearly unqualified to comment on this topic, so just what are you trying to suggest? Are you a published researcher on digitized systems? Nelson Pass, John Curl, Demian Mark are not published researchers in digitized systems, so why even bring them up (Scott, if it is the Scott I am thinking of may have). Are you suggesting you are in the same class as them?

Teo,

You are displaying the same technical ignorance as the author of the article. Just "stating" something without understanding the underlying mathematics does not make your statement true, not even remotely, all it says is you do not understand the underlying concepts.

Bandwidth, signal to noise, and jitter absolutely are the functions that define relative timing resolution in a digitized system and high quality audio is not some magic bullet that magically changes how math works. Even in high end audio, 1+1 = 2.

teo_audio1,245 posts11-21-2019 8:20am

Timing resolution of digitized bandwidth limited signals.

Of course, timing resolution and bandwidth limiting not being the same thing at all, except in some given limited mathematical applications.

High quality Audio ---not being one of them.

And I could bring forth many musicians who would tell you that digital recordings are more truthful to the sounds they are creating, even though "analog", with its limitations may have lovely euphonics. Who is right, them or you?

I could also bring forth many audiophiles who also prefer digital to analog, whether vinyl or tape or otherwise. Are you more right than they are?

And yes, throwing math is appropriate. Humans perceive sound, they don’t perceive electrical signals in a wire, or bits on a disk. You can call the science of human perception weak, but the science of what happens w.r.t. the storage and reconstruction of information, namely what is captured by that microphone and played back through the electronics, up to the speaker, that science is exceptionally strong, bounded by very robust science and math.  As per your statement below, models change .... math does not. The math behind sampling, reconstruction, does not change because you add a human observer to the equation.  Intentional alterations of a signal may change to suit preferences, but the math does not change.

In such a situation the factual bits are fuzzy but it may be a good idea to maybe look at that end of things to get a better idea of how to optimize the digital math.

The article throws no new light on the argument at all. We have known for some time that we can detect time of arrival differences in the ears to the microsecond level, and that neuron fire rates are fast. Literally no one is questioning. The whole premise of the article is that "timing" is limited to the sample rate of a digitized system .... a premise 100% false within the confines of a bandwidth limited system and no one has ever shown that our ears/auditory system is anything but a bandwidth limited system, and this article did absolutely nothing to disprove that it is not bandwidth limited.

I feel you are not being honest with me or yourself by accusing me of a double standard. I have done nothing but address the actual technical content of the original article, and the posts made against what I said, that as opposed to actually addressing what I wrote, essentially only attack me.

As opposed to attacking me, perhaps you could address what I have actually wrote and show me and everyone else how I am clearly wrong. I  have posted data simulations and links to several papers (written by people that understand the topic) that clearly show that a digitized system can carry within it relative timing information that is well beyond the sample rate. The whole premise of the article is that the timing is limited to the sample rate. That is false. That makes the whole premise of the article also false.


teo_audio1,249 posts11-22-2019 11:55amcareful with those double standards... and your penchant for putting words in others mouths that they have not said.... and then using those false premise to attack their view or position.

It is fine to say that I am wrong, it is another to prove that I am, or even show some logic to justify that I am.


The initial aspects of the auditory system is a mechanical system. Heck, the transfer of sound is a mechanical system. But even if not, all systems are bandwidth limited at some point, it is just that mechanical systems are often bandwidth limited at lower frequencies.


Neurons firing in a microsecond does absolutely nothing to prove that the bandwidth limit of the auditory system is not 10's of KHz. Similarly, being able to time arrival based on phase difference between two points (our ears) in space to microsecond timing does nothing to prove that the bandwidth limit of the auditory system is not 10's of KHz.

teo_audio1,247 posts11-22-2019 11:02am

a premise 100% false within the confines of a bandwidth limited system and no one has ever shown that our ears/auditory system is anything but a bandwidth limited system, and this article did absolutely nothing to disprove that it is not bandwidth limited.

Say what?

Your reading comprehension is way way off....which indicates a multitude of other ......

Feel free to show me how I misinterpreted what you wrote:

teo_audio1,249 posts11-22-2019 11:02am

a premise 100% false within the confines of a bandwidth limited system and no one has ever shown that our ears/auditory system is anything but a bandwidth limited system, and this article did absolutely nothing to disprove that it is not bandwidth limited.

Say what?

Your reading comprehension is way way off....which indicates a multitude of other ......

Are you seriously suggesting that we cannot measure electrical signals, to very high bandwidths, and with exceptional resolution?  That is definitely what you appear to be saying?  Is that what you are saying?

I really don't need to say much more than that. You can call my statement ridiculous, but you just claimed that our ability to measure electrical signals, to very high bandwidths and with high resolution does not exist. How I am supposed to take that seriously.  I guess all those amazing scientific instruments we have really don't work at all?

Maybe you were just so quick to call me wrong that you misinterpreted what I said?   I am willing to give you the benefit of the doubt.

taras22303 posts11-22-2019 11:30am

but the science of what happens w.r.t. the storage and reconstruction of information, namely what is captured by that microphone and played back through the electronics, up to the speaker, that science is exceptionally strong, bounded by very robust science and math.

As the Brits would say, brilliant, just brilliant, its absolute comedy gold....in fact that has to be one of the funniest things I have read in quite a while. I can only thank the powers that be I wasn’t drinking coffee at the time because I surely would have lost a laptop....or I didn’t faint and fall and crack my skull ’cause I was laughing so hard I got very very light headed.

And delivered with such certitude / straight face delivery...to paraphrase the great and grand Zaphod, brilliantly brilliant.

It is rather cute Teo and Miller how you two bloviate with flowery language that would make it look like you know what you are talking about but to someone who actually understands this stuff you just sound ... Funny. Yes funny.


Teo. Jitter on recording is in the 10’s to 100’s of picoseconds. Ditto on playback. Easily modelled as phase errors or distortion. Distortion 100+ db even on real music signals. I know you don’t want to believe it but audio is easy. We do 100+db on complex signals day in day out in comms.


Not my first rodeo and I know the usual arguments that is why many many posts ago I posted a paper (that included actual experiments) that showed timing accuracy as a fraction of bits with very low signal.to noise ratios, ie like 30db, and the resolution is several orders of magnitude finer than the sampling rate.


Believe vinyl/current analog is superior ... No issues. But don’t hang your hat on something you are going to fail miserably on. I can post many experiments that clearly show high subsample timing accuracy in low SNR environments of which digital audio is not one.


All two channel does is increase the relative jitter mapped as higher effective SNR but since it is already very low the timing resolution is very high. Feel free to argue points you don’t have the background to argue, but at least back that up with some solid work by real experts to support what you are saying. Otherwise it is just hand waving.

You are confused because you do not understand the science, hence you put it down.

One of the papers Again clearly show orders of magnitude subsample timing at 30db SNR. Never will a digital audio system ever get anywhere near having as little as 30 db SNR, and if you say when the signal level is low, well then in your analog system it is all noise at this point. That paper looks at low SNR situations because the noise can be really bad, Unlike audio! Absolutely nothing I posted says the opposite of what I said and that you would state that suggests you really didn’t try to understand them. One only discusses difficulty in the presence of noise and other sources .... With examples where the SNR is 10dbs of db, not 90+, and they did not bandwidth limit the signal.

You post excerpts, not links to articles, why is that? The parts you link to show a lack of understanding and the same flawed thought process as the author. Audio Is Not a Real Time System. It is a recorded and played back system. There is 0 concept of real time in Audio. There is 0 concept of absolute time. Everything is audio as we are discussing is relative. That is why I clearly and distinctly use the term relative timing in most of my posts. That is the difference between actually understanding a topic and cutting and pasting things that match your world view.

Stop clutching at straws to attempt to justify a point of view that is wrong. I read the article. I understand it. That is why I know it is flawed. You want it to be true, but wanting something true and it being true are not the same.

Everyone who understands signal processing and digitized systems will instantly pick out the flaw in the premise of the article which is exactly what others online have done wrt this article, to the point of contacting editors because it is such a gross misrepresentation of reality.

That was dishonest of you posting those excerpts without a link to where they come from. Turns out, at least one of those excerpts was nothing more than a post from someone on an audio forum. Not an expert. Not a paper. Just someone with an opinion just like you.  He was quickly shot down in flames ... As others pointed out he did not understand bandwidth limited systems.  It is clear from his continued writing he does not understand the math (nor science). His assertion is continually proved wrong. Here is where you got that quoted text from:


https://www.google.com/url?sa=t&source=web&rct=j&url=http://forums.stevehoffman.tv/threa...


Here is another thread where people who understand the subject have a discussion about how the person you linked had no clue what he was talking about:


https://www.google.com/url?sa=t&source=web&rct=j&url=https://hydrogenaud.io/index.php/to...

Not only did I take the time to read what you wrote, I researched your uncited quotes to find they were just musings on a forum by someone with fundamental holes in their knowledge, not an actual expert, and not remotely reviewed piece of work. The papers I posted do not all support that forum posters hypothesis or yours.


Shannon Nyquist theorem is highly understood and not doubted. The links I posted are highly relevant. I specifically picked a paper with wickedly bad SNR to illustrate that for audio where SNR is high, that it is pretty much non issue wrt providing well beyond subsample resolution. That you think they are not relevant either shows you don’t understand the underlying concepts or your argument is a red herring. Which is it?


Your Fermi quote is a red herring. When Fermi says contrary to hypothesis, he means he reran it 100 times, reviewed all his equipment, talked to trusted colleagues, and then published a paper fully detailed so others could replicate the experiment. He did not mean a pop science paper by someone who does not understand the science.


I provided actual scientific papers. You quoted a post on a forum. That I think says it all.

So basically the system didn’t filter out high frequencies like it should have bean designed to? Bad example.


yes I am a guy on the forum who is not quoting other people on other forums I’m actually providing links to real papers. I am also not saying anything that conflicts with well-established Shannon Nyquist.

I don’t need to channel Fermi to understand how scientific Labs work or engineering labs for that matter. If you get a measurement but doesn’t correspond to what you expect your first inclination is not to think you’ve discovered something new it is to assume that you made a mistake in your measurement. At that point you will review your equipment redo the measurement try to do the measurement a different way look at how you may have made the measurement wrong or what is wrong with your hypothesis. Only after exhausting all other options will you assume you have made a discovery. Even then you were probably wrong.

Basically because you have ignorance about this topic and refused to accept the fact that you are wrong you are expecting me to prove Shannon Nyquist that is essentially what you are asking.


the one paper I did link to that shows timing under severe signal to noise restrictions says everything that is essentially needed on this topic.


You on the other hand have provided absolutely nothing to support your position. Your position is essentially that Shannon Nyquist is wrong and you’re not qualified to say that. And because you were trying to say Shannon Nyquist is wrong you can’t find anything that will support your position because it doesn’t exist.


given that you have illustrated you don't have the technical acumen on this topic how would you know a good paper if you saw one?

I ... Ya me, did provide articles. Who do you think you are fooling with this childish game asking for more? You ...what did you provide? ... You cut and pasted from a forum post, but didn’t provide the link to that post. That is bad form. Is that because the poster was getting roasted due to his inaccurate claims. He did not even understand something as basic as impulse in a bandwidth limited system. That is your "proof" you are right. All my posts in this thread are easily researched. My examples easily understood.

The one that needs to provide supporting articles is not me, it is you Taras22. You continually attack but can’t back it up. It is your credibility on this topic that needs help.

The articles were very cogent for those with the knowledge to understand them. Both were fairly simple actually as papers go.


If you need it applied to music, then again, you clearly don’t understand the science behind sampled data systems. Reality doesn’t suddenly change for the hifi industry no matter what some would wish. When a 30db SNR system achieve timing on the order of 100th a bit, audio systems at 90+ are not going to have an issue.


The whole premise of the article is laughable and you keep beating the dead horse. And yes, you did you the quoted and linked text as attempted proof that I was wrong. Why you didn’t cite it, only you can answer.

falconquest,

I would like to think that no one is suggesting we don’t all perceive sound differently. I think that is a given. This is not even a discussion of whether 44.1/16bits is enough bandwidth/resolution. It is about whether audio sampled at 44.1 khz has sub-sample timing resolution. Well really it is not even a discussion, no more than 1+1 = 2, or whether the earth is round or flat. No one who understands sampling theory and digitized systems thinks that audio timing resolution, monaural or binaural, or any bandwidth limited system is limited to the sampling rate. The author of the paper flat out states that timing resolution is limited to sample rate. That is just not true. Unfortunately, things like this pop up every few years, then get smacked down, but people have short memories and don’t do the required due diligence.

If the author had just stuck with 20KHz is not enough bandwidth, then he would have had a fairly supportable position (no matter how many scream Redbook is enough). There is a pretty strong case to be made that 20KHz is not enough. There appears to be no case for beyond 96ksps (40KHz).

Unfortunately, people from both sides of the argument are more interested in emotional positions than ones based purely on facts.

For a drive-by shooting, his aim was incredibly accurate.

True, but it is always "spinning" so that does increase the level of difficulty.

mahlman126 posts11-26-2019 1:25p
  Well it is a big slow moving target to hit so there is that to consider too.

GK,
It is nice to see you finally admit you are an angry pseudo-skeptic.

Arguably rarely right. The naysayer is almost always wrong. Sometimes they are right, and they end up with a Nobel prize, not selling a "Teleportation Tweak" to gullible customers.

teo_audio1,258 posts11-27-2019 9:10am

I’m the most successful one selling things on this site.

Is Teleportation Tweak considered a "thing" or it is some other state of non-being?

The problem for the naysayers is that the science and the physics has always supported what he is doing and what he is saying. It has no method of disproving him -- rather the opposite, in fact.

Similar to the family dinner table, it is probably best to leave religion and politics out of this discussion.

mahgister804 posts11-27-2019 11:04amQuestion: how is it that some scientific discourse, for example darwinism, is kilometers away, parsecs away perhaps, from scientific thinking process ?

At the University of Toronto, many of the Engineering disciplines like Electrical and Computer, Biomedical, Aerospace, and Engineering Physics were split out from some of the more "traditional" engineering disciplines into a sub-Department called Engineering Science.


This is the background on Engineering Physics, which is specifically administered within the Department of Engineering Science, in the Faculty of Applied Science and Engineering.

"The Department of Physics at the University of Toronto, together with the Faculty of Applied Science and Engineering, created the Engineering Physics program in 1934 (called Engineering Science since 1965). The Physics Major continues to attract students with a keen aptitude for physics who see the creative potential for combining this with an engineering degree. Graduates appreciate the high degree of flexibility provided to them in terms of the design of their program across a wide spectrum of theoretical and experimental physics courses."