24 bit in and around, 16 bit out?

  • Thread starter Thread starter jedblue
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Here's some more food for thought. Take any actual recording anyone has ever made, low pass it at 22k (with no bump in the LP filter at the knee), Double-blind test the two versions with any group of people you want, and tell us which one their sixth senses or invisible organs or whatever tells them is which, and is correct more than 50% of the time when it does.

Look, miro, we can't worry about speculations in our racket. They're fun to talk about over a beer after the session or in Internet forums, and they can be interesting (sometimes) to investigate by those who's job it is to investigate such things. But for us who apply the technology, it's impossible to anticipate the unknown.

Let's say, just for the sake of argument, it is possible that frequencies above 22kHz or so are actually important for some reason. We don't *know* that, but we speculate it's possible. Fine.

Now, by definition, since we don't really know, it's just as possible that those frequencies mean nothing to us. That makes it a 50-50 gamble at best. But that's not the end of it...

There is also the equal possibility that such EHF signals do have an effect, but it's a negative effect, not a positive one. That now means that there is a 2 out of 3 chance that the effect is either negative or non-existent. Any bookie, stockbroker, scientist, pit boss or palm reader will tell you those are not odds to bet on.

And all this is before one even brings the reality into it; that even at higher sample rates, the energy at EHF freqs just ISN'T there and can't be reproduced by the gear your using anyway.

It's not a matter of whether one hears or feels them through some undiscovered mechanism or not, it's a matter of they just aren't THERE. This is not a matter of something beyond science, it's well within the realm of current science and experiment.

@ MYMorningstar; if you missed it, the answer to your question git buried at the end of the last page (I hate when that happens :p).

G.
 
Glen, the FFT picture you are talking about does not show in that thread. Instead, there is that red X. Perhaps the link to the pic is no longer valid? :confused:
This isn't the first time someone has said that my in-line images aren't showing up, when they work just fine for me. I have no idea what is wrong there. Are they showing up in this thread? Let me know, and I'll try something else if it's still not working.

G.
 
This isn't the first time someone has said that my in-line images aren't showing up, when they work just fine for me. I have no idea what is wrong there. Are they showing up in this thread? Let me know, and I'll try something else if it's still not working.

G.

I see them fine.
 
I see them fine.
Thanks Rami!

I just checked, and in the User CP (Control Panel) for this BBS software, under the "Edit Options" selection, there is a box called "Thread Display Options" that has a checkbox called "Show Images (including attached images and images in code)". That needs to be checked in order to see in-line images. Noisewreck (and others), you might want to check this option, that might be what's preventing you from seeing them.

G.
 
And all this is before one even brings the reality into it; that even at higher sample rates, the energy at EHF freqs just ISN'T there and can't be reproduced by the gear your using anyway.

Yeah, this pretty much sums up my situation: my mikes are rolling off some at 20K (somewhere between 5 and 15 dB) and probably heading south pretty quickly above that, though I don't have any response plots that go beyond 20K. So there probably wouldn't be much up there, even if I were tracking to tape at 30 ips. I really don't know how much my tweeters put out above 20K, but it will certainly start to get very directional. My headphones claim to be no more than 3 dB down at 30K and 10 dB down at 38K. So, at best the question is what is the difference between bandwidth limiting at 22K or having a little bit of response, probably down 15 dB at 22K and 30 dB or perhaps much more at 30K.

I also notice that when I was younger I could hear much higher and there were stores that my sister and I could not go into because of the pain of the high frequency noises in the lighting section that no one else seemed to take any notice of. Now I can barely hear a test tone at 16K.

Cheers,

Otto
 
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also notice that when I was younger I could hear much higher and there were stores that my sister and I could not go into because of the pain of the high frequency noises in the lighting section that no one else seemed to take any notice of. Now I can barely hear a test tone at 16K.
There's actually a company out there that makes devices that put out a constant screech at something like 17k or 18k (give or take, I forget the exact freq) in order to annoy younger people and chase them away without affecting us old codgers. The idea is to use them in places like Quik-E-Mart parking lots or mall atriums or places like that where the proprietors don't want them just hanging around and chasing away business. I won't argue whether it's a good idea or not, just pointing out that someone has used this fact of human nature as an actual tool.

Then there's the "Mosquito Ringtone" popular amongst school students who wish to not be students or to be in school, which is a HF ringtone that they can hear but their teachers often can't.

Even dog whistles, which are commonly touted as examples of sounds that can be heard or "felt" by some humans even though they are "beyond normal human hearing range" typically fall between 16k and 22k, with the ones that most folks can "sense" being in the lower half of that range. Not exactly as EHF or supersonic as they can be made out to be.

And yeah, your gear, and my gear, and most all gear begins rolling off at the top of the "human range". There are some monitors and audiophile speakers that'll claim 35k or better on the top end of their response, but it hardly matters when playing a 44.1 CD of songs recorded though microphones, does it? ;) . And while the charts on mics and most other gear may not go much above 20k, the physics of the situation pretty much tell us that the frequency curves are not going to suddenly reverse and climb again at EHFs.

G.
 
It's not a matter of whether one hears or feels them through some undiscovered mechanism or not, it's a matter of they just aren't THERE.

Not there because of….the limitations of some recording device...or not there naturally?

Since we're just kicking this around over beers.... :)

Again...if we just assume that everything outside of the 20-20kHz range is useless info, then the bar will never be raised higher, and we will just accept not needing recording gear that can capture outside of that range. ;)
And even if all the mics roll off outside of 20-20kHz...you may end up with a lot of tracks that individually may be in that limited ranage, but in combination could generate new/additional harmonics that go outside of 20-20kHz.
Of course...if all equipment is made to roll off at 20-20kHz because we've accepted those cut-off points as “good enough”, then anything outside of that range will certainly never be heard/sensed/felt... :D

There are some monitors and audiophile speakers that'll claim 35k or better on the top end of their response, but it hardly matters when playing a 44.1 CD of songs recorded though microphones, does it? ;)

OK, it may not matter for R&R (not to mention the current “preferred” medium of most listeners)...though the choice of 44.1 for CDs has little to do with that being the optimum sampling rate for music.
It was just a cobbled format that came out of the Sony-Philips' power struggle and the existing technology of the day.
I don’t think anyone back then said "this is it...no need for ever improving this format any further"...that's just the “best” they had at the time...so like now we've all decided there's no need to raise the bar higher...???..though I guess the prevalence of the MP3 format may be the answer to that question. :(



There were a couple of good points about bit depth and sampling rates raised on this web page that I posted previously:

http://www.macintouch.com/readerreports/audio/topic4467.html


David Charlap

There seems to be some misunderstanding about what the Nyquist frequency is. Some recent posts seem to imply that if you sample at this frequency, then you'll get a perfect reproduction of what you're sampling.
While this is true, it is not the whole truth.

If I am playing an "A" (440Hz) on a sine wave, then I can sample it at 880Hz and get a perfect reproduction.

If, however, that "A" is played on a piano, an 880Hz sampling will destroy it. The fundamental tone will be sampled accurately, but the overtones and harmonics (that is, the components that make a piano sound different from a sine wave) will get lost. You'll end up with an "A" that only sounds vaguely piano-like.

While the wave itself will be at 440Hz, the shape of that wave will be fairly complicated. From an engineering standpoint, it can be thought of as a composite of many waves at different frequencies and amplitudes (most of which will be integer multiples of the fundamental 440Hz.) In order to accurately sample the shape of the wave (and not just its fundamental frequency), you'll need a sampling frequency much higher than 880Hz.

So, although 44.1KHz can accurately sample fundamental (that is sine-wave) frequencies as high as 22.05KHz, a complex wave-shape at 22.05KHz will require a much higher sampling rate if you want an accurate representation.

96KHz and 192KHz sampling (as used in SACD and DVD-A discs) can often sound better because of this. Nobody can hear a 40KHz note, but the notes you do hear (especially the high notes, 8Khz and up) will be more accurately represented. Depending on the nature of the source material and the quality of your playback equipment, the differences can often be easy to hear.


..........................


Mark Bailey posts several reasons why (in his opinion) 24-bit and 96KHz audio interfaces are a pointless waste of money.
If all you ever do is rip CDs and play MP3 files, yes, it is a waste.

If, however, you do actual audio work (live recording, mixing, editing, etc.), it is necessary.

Every single time you perform an edit (mix, equalization, effects, etc.) you introduce errors into the data. This is because computers don't have infinite precision - every computation prduces a small amount of "round-off" errors in the least-significant bits and in the highest frequencies. The more you edit, the more the errors accumulate.

If your original source recordings are at 24/96, then the errors will occur in the bits/frequencies that nobody can hear. When you convert the audio to standard 16/44 (e.g. for burning a CD or making a file for internet distribution), the errors go away because they're all in the discarded bits/frequencies.

If, however, your source recordings start out at 16/44, then those errors will end up in your final product, and (depending on what kind of editing/mixing you've done), they can be audible.

There is a reason professionals always use 24-bit interfaces at 96 or 192KHz sampling, and it's not because of some pathological need to waste money.
 
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This isn't the first time someone has said that my in-line images aren't showing up, when they work just fine for me. I have no idea what is wrong there. Are they showing up in this thread? Let me know, and I'll try something else if it's still not working.

G.
I see it fine in this thread and thank you for the explanation that goes with it.
 
Thanks Rami!

I just checked, and in the User CP (Control Panel) for this BBS software, under the "Edit Options" selection, there is a box called "Thread Display Options" that has a checkbox called "Show Images (including attached images and images in code)". That needs to be checked in order to see in-line images. Noisewreck (and others), you might want to check this option, that might be what's preventing you from seeing them.

G.[/QUOTE]
I went to the CP and the box was already checked. I went back to the tread and there is just a red X where the pic should be showing. The X has the following properties which shows as not a valid address:

[url]http:///www.independentrecording.net/irn/file_xfer/reaktor_test.jpg[/url]

This is the properties for the one on this thread that I see fine:
[url]http://www.independentrecording.net/irn/file_xfer/reaktor_test_2.jpg[/url]

Hope that helps you narrow it down. Right click and choose properties to the the complete addy
 
Here's what I don't understand. Why is everyone worried about all those ultra-high frequencies that only your neighbor's dog will hear anyway?

I am here to defend and represent the ultra-low frequencies. I feel they have been vastly underrepresented in these parts. :D

I say we should try to extend the reproducible frequencies below 20Hz as much as possible. Why? Because those you can definitely feel if not hear. I mean, they will rattle your bones, make your head implode, make ladies' skirts blow over their head if speakers are positioned properly...

See? I think those low frequencies are much more fun.

I WANT ME SOME 5Hz RUMBLE!!!
 
If, however, that "A" is played on a piano, an 880Hz sampling will destroy it. The fundamental tone will be sampled accurately, but the overtones and harmonics (that is, the components that make a piano sound different from a sine wave) will get lost. You'll end up with an "A" that only sounds vaguely piano-like.

While the wave itself will be at 440Hz, the shape of that wave will be fairly complicated. From an engineering standpoint, it can be thought of as a composite of many waves at different frequencies and amplitudes (most of which will be integer multiples of the fundamental 440Hz.) In order to accurately sample the shape of the wave (and not just its fundamental frequency), you'll need a sampling frequency much higher than 880Hz.

So, although 44.1KHz can accurately sample fundamental (that is sine-wave) frequencies as high as 22.05KHz, a complex wave-shape at 22.05KHz will require a much higher sampling rate if you want an accurate representation.
We're going around in circles, here miro, because you keep ignoring some basic facts (and, yes, they are facts, and not just things we think are facts because of the limitations of our current science.) What David says above is all true, until the last paragraph. It would be truer if he added the words, "of any potential harmonics of that 22.05KHz signal". (Technically, less than 22.05 because of the aliasing filters, but we'll put that aside for a moment.)

But it still ignores a couple of basic points:

First; that there's nothing creating *fundamentals* at 20k or above, except for a couple of brands of dog whistles. Anything existing up that high is already going to be a multiple order harmonic of a much lower frequency fundamental. For example, the A440 he uses as an example would be at it's 45th harmonic by the time you got up to 20k, which just plain isn't going to be anywhere near audible - *if* a piano even created harmonics of that order. In fact, the highest fundamental a piano can create is a C8 at about 4.2k. 20k is almost 5 overtones above that, which again, is going to be a negligable amount of energy; even if we *could* hear or "feel" it, there'd be next to nothing to hear or feel.

OK, there are some pipe organs which can go slightly more than a half-octave above that on their fundamental to an A8, which is still only 7.1k. If anybody really wants to listen to a pipe organ at sibilant frequencies (even though we all know how grating sibilance sounds), we'll be limited to reproducing approximately the third overtone of that sibilance. Somehow I doubt anyone is going to miss the 4th overtone and above of a 7k fundamental which is already giving them headaches.

Other than that, even if we ignore the fundamentals, when it comes to harmonics of instruments, about as high as they go is the violin and some cymbals, which tend to end their audibility at around 16-17k (where dog whistles start.)

There is real physics involved here. The higher the order of harmonic, the less energy it carries, and that for anything that sounds even remotely pleasing to the human ear, by the time our harmonic order gets high enough to worry about Nyquist, they simply become virtually irrelevant to the timbre of the instrument or sound.

The second entry to physics is in the design of the human ear, as I said before. Dogs and cats can her much higher frequencies than we can, and the exact,specific reasons are measurable; first, it's because of the size and shape of the cochlea, which, like a french horn or tuba, is designed to channel certain frequencies in certain ways. The dog and cat cochleas are measurably designed to handle higher frequencies, just like the size and shape of the french horn handles higher frequencies than the tuba does.* Second is the characteristics of the cilia, the hairs within the cochlea, whose vibrations turn the sound into the nerve impulses which go to our brains. Their size and position within the cochlea determine which frequencies they can deal with. (It's the degradation of the finest of these cilia which are the main cause if tinnitus, BTW; they "malfunction" and send constant impulses to your brain which we hear as "ringing of the ears".) And the design and build of the human ear dictates that it just cannot send ultrasonic signals to the brain any more than a tuba can generate a 16k overtone

This is all known science which further science can build upon but will never repudiate. To try and sidestep this by going to the unknown future is just taking random pot shots that have a far greater chance of missing the side of a barn than they do of hitting any real factual target, now or in the future (I'm still waiting for my flying car and ESP-in-a-pill, which they told us when I was small we'd all have by now! ;) )

If, however, you do actual audio work (live recording, mixing, editing, etc.), it is necessary.

Every single time you perform an edit (mix, equalization, effects, etc.) you introduce errors into the data. This is because computers don't have infinite precision - every computation prduces a small amount of "round-off" errors in the least-significant bits and in the highest frequencies. The more you edit, the more the errors accumulate.
Incorrect. The kind of quantization errors he's talking about are NOT cumulative in the way he describes them. Refer back to Ethan's discussions re this earlier in this thread. Nor will the errors be concentrated in higher frequencies just because you have higher sample rates; the distribution of the digital noise remains even across frequencies and it's level is a function of bit depth, not sample rate.
There is a reason professionals always use 24-bit interfaces at 96 or 192KHz sampling
There might be a reason if they actually did. But this is a mis-statement; te majority of professionals that I have worked with and talked to on these forums use and recommend 44.1k/24 for pure audio and 48k/24 for audio for video post. Perhaps, as George mentioned, 96k/24 for SACD production, but that's not for the point of capturing phantom overtones, it's for the idea of avoiding extreme sample rate conversion factors.
Again...if we just assume that everything outside of the 20-20kHz range is useless info, then the bar will never be raised higher, and we will just accept not needing recording gear that can capture outside of that range. ;)
Yep. And there's nothing wrong with that. What's the point of dealing with useless information - or more to the point, information that doesn't exist?

It's not a question of us setting the bar too low, it's a question of Mom Nature setting the bar herself, and all we have to do with our technology is clear it. There's absolutely no point in jumping 50 feet over an 18 foot bar.

This is all sidestepping the real question, though, miro; why do you feel the inner need to discuss, accept and defend the fringe topics? No offense intended, here, but I notice that we seem to always wind up getting the biggest head bumps on subjects like MP3 encoding bit rates, sample rate and so forth which frankly mean so very little in the grand scheme of audio engineering.

*BTW, have you ever noticed that even though dogs and cats can hear up to maybe 40k-50k, that they NEVER find the ultrasonic stuff to be pleasing? They are not attracted to dog whistles any more than we are attracted to air raid sirens, and I have never met a dog that actually enjoyed the ultrasonics put out by the suction motor of a vacuum cleaner ;)

G.
 
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AH HA! I don't know why I didn't see this earlier; the third slash after the "http". it must be that my browser can deal with that but some browsers choke on that (it's not supposed to be there, my browser is a bit too lenient, I guess ;) .)

Thanks, 'Star. I'll fix that. Sorry for the problems.

EDIT: Aargh! The post is too old and the BBS won't let me edit it any more. Maybe if there's a bored mod reading this, they might be kind enough to remove that slash from IMG tag url in this post.

G.
 
*BTW, have you ever noticed that even though dogs
Now that's really scary that we both started talking about dogs at the same time. Eery.

Now as for psychoacoustics that Miro is referring to (I think) don't necessarily depent on what you can hear (or indeed what your ears do with) those frequencies, but rather whether you can feel them in some sort of way... perhaps tingling through your skin, :D I dunno...

However, I do agree with your refutations of that guy David Charlap. If you have an audio source creating fundamentals at even 15k... or even at 10k, you got some serious issues.

Miro, don't make me go and pull Reaktor out and create some more ugly Saws at those frequencies... Don't worry, I'll record them at 384kHz (yeah, Reaktor allows for it) so you get all those nice overtones :D
 
Now that's really scary that we both started talking about dogs at the same time. Eery.
Well, i can't speak for you, but I think with me it's because I'm in heat ;) :D

Kinda like when I start using cooking/food analogies whenever I get hungry :D.

G.
 
miro; why do you feel the inner need to discuss, accept and defend the fringe topics? No offense intended, here, but I notice that you are not defending the points yourself, but rather choosing the opinions of others who agree with what you want to believe.

Well...some of the "fringe" topics are interesting, especially where there is still some room for thought and where some juries are still out as demonstrated by views of others…not just my own, but I’m certainly NOT the single “hold out”! :)
I’m just trying to view the whole thing from as many angles as possible.
I witnessed this same basic discussion about bit depth and sampling rates on a forum moderated by George Massenburg, and the thread went on for like 150 pages or something! There were a LOT of opposing views, and if I recall, even GM didn’t really stand firm on any one side….though maybe that was just his way of letting the discussion continue on.
So I'm not sure if it's really a "fringe topic" as I've seen it come up time and time again over the years, not just on these forums, but also on other forums and on websites like the ones I posted links to, by folks who apparently have done some deeper experimentation and/or investigation.

And no, I have not done extensive double-blind experiments myself, however, I think I can digest what others are doing/saying and then finding points to believe in or at least points worth questioning further.
I didn't think that before we can discuss or agree/disagree on a particular subject or point we must have firsthand experience and be directly involved with the testing and proving of that subject...do we? ;)


So the question I have is why you present the appearance of needing to hold on to beliefs on the fringes of the subject, and why the idea of there being a natural limit or bar to some things is so repulsive? I think there is great importance in the true answer to that question.

I'm not really holding on to any specific beliefs, rather I am looking for the most absolute, definitive answers possible and this is the best way I know how to go about getting it…by asking questions, maybe even playing devil’s advocate on some points, and maybe even looking or hoping to find a little more than just what the current science has presented.


Don't get me wrong, I like you, and find you to be one of the smarter guys on this board, and when you first came on here I supported you when you thought you may have mis-stepped a bit. But I notice that we seem to always wind up getting the biggest head bumps on subjects like MP3 encoding bit rates, sample rate and so forth which frankly mean so very little in the grand scheme of audio engineering.

And don’t get me wrong…I’m really not trying to butt heads with you (or anyone for that matter).
You seem eager enough to explore the questions I’m asking and the opinions (even of others) I am presenting…and just like those other opinions that I reflect on and come to conclusions from....I am also listening to yours.
If my only ticket into this debate would be that I first go and conduct a lot of serious experimentation and prove to myself all these things…then I wouldn’t have the need to hear anyone else’s opinons! :)
If it’s any more palatable…we can talk about the origins of the universe and black holes, but I say it up front, I’m not a physicist nor have I done any experiments on those subjects ( but I do watch a lot of Discovery and the Science channels!).
I mean…it’s more fun kicking these “fringe” cans around here….then hanging out in The Cave!!! :D
But I like talking about more fundamental, “hands-on” recording issues too…when they come up.
Seems like the last few threads have been about bits and sampling rates. *shrug*

We can stop talking about it…that’s fine too…I wasn’t looking to change/adjust/impove anyone else’s beliefs other than my own.
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IME, one of the reasons these topics go on for so long is because the differing opinions of many are based upon the differing opinions of many others, and very few are actually working from first principles and actually figuring things out.

As I have said before, the biggest problem with topics like sample rate is that they are by their very nature *very* difficult topics to comprehend - digital information theory makes topics like compression and gain structure and so forth look like first grade math, and is on a level of intuitiveness almost as bad as quantum mechanics (the two are in fact quite related in many ways).

I have also said that I don't claim to fully understand all the math and technical details of it; I remember back in college heading to the school library and pulling out a book on telecommunications and information theory, and within an hour I actually felt so light-headed and goofed out that I literally had to just sit down fro a while and decompress because I was actually afraid to get behind the wheel of a car lest I wind up crashing it because I just wasn't feeling right. True story.

But, I have done enough study and wrapped my head around enough of it since then to know that half the stuff that's taught in the derivative consumer books on audio are oversimplified to the point to cause many who read them to come to wrong opinions about how it all works. Add in the culture of the Internet, where derivative opinions and untruths spread like contact highs in an opium den, all opinions are considered equally valid truths, and "long posts" discussing complex subjects are derided as being too hard to read, and the bullshit just piles up so high you need a snorkel just to breathe.

The more difficult and complex the subject, the worse an idea it is to get one's information from other people in Internet forums. By definition, that includes me :o.

But it also includes even people who are even more knowledgeable, like GM and BK. Stick with the subject instead of what others think about the subject. Easier subjects it's OK, but when it actually IS rocket surgery, there's no substitute for first-hand learning :).

G.
 
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To echo Glen's points about difficulty with/lack of understanding all this digital mess is because the ones that truly understand how digitizing and analog reconstruction works are the people who are not musicians. They are programmers and DSP specialists who don't necessarily have "musical ears". The ones that actually have musical ears, don't have the necessary DSP knowledge to understand how the whole process works. So you get all these opinions, half-truths and such from otherwise well meaning people who themselves don't quite get it, and unfortunately for them, most of the time they don't know that they are not quite getting it.

I remember having a conversation with a guy who had built some impressive stuff in Reaktor (just looking at the structure made my head spin). We were talking about the sound quality of the Low Pass filter on various synths, and I was telling him that I found the filters on OSCar to have a certain three-dimentional feel to them, where the filters in Reaktor while effective had a certain shallowness to them. His response was "I don't get it, they just make the sound dull" :D

It is this disconnect between the musical types and the DSP engineering types that I believe is part of all this misunderstanding.

I am a person that has a relatively good grasp of physics and calculus but the DSP formulae used in things such as FFT, sample-rate conversion and such make my head spin, especially once you move away from the basic recursive summing and sample-delay.

The other complication is as has been discussed and reiterated many times is that the seemingly basic act of AD/DA conversion involves so many steps, processes, hardware designs, software designs that it is almost impossible to pinpoint the even audible sound differences on one thing alone. You cannot take sampling rate independently of a given filter desin or clock being used or the design of analog stages themselves to make any valid arguments for or against one thing or another.
 
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If it hasnt already been said... I think there are at least two separate issues being discussed.
1. Is 20khz a high enough upper limit for audio reproduction?
2. Will 44.1khz adequately reproduce up to 20khz?

For the second question, no, 44.1khz sample rate will most definitely not reproduce a complex 20khz waveform because that's not what the specification means. The specification for audio normally refers to sinewaves, at least for the upper limit.That applied long before digital came along.

So an analog piece of gear that is specified to be flat from 20hz to 20khz is only flat for sinewaves. Sure, it will handle complex waveforms but only up to the equivalent of a 20khz sinewave. A complex 20khz waveform is simply out of the specified range.

Similarly, (in reference now to 1.) when we speak of human hearing extending no higher than 20khz we are referring to 20khz sinewaves. A complex 20khz waveform is not even considered. We humans have a hard enough time hearing a pure 20khz sinewave. Add harmonics to it and I think most would agree, all bets are off.

I used to be able to hear 15khz, just. Ten years later, I can only hear up to about 13khz. Has this diminished my love of and appreciation of music? Not in the slightest. In my judgement, most of the "meat" of music, at least for humans is usually in a frequency range well below 20khz.

FM radio shelves everything above 15khz. Return that extra 5khz to the broadcast and I suspect only a tiny fraction of listeners would even notice, if at all. It's good to be practical about these issues.

Cheers Tim
 
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