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Sampling rates higher than 44.1Khz?
Grand Dizzy
post Feb 5 2006, 01:10
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I recently met a musician who claims he can quite easily hear the difference between 44.1KHz and 96KHz.

This shocked me a little because I'd always been told that the human ear cannot hear any higher quality than CD (44.1KHz) quality.

So... was this guy just lying (or fooled by his senses), or was I being lied to when I was told the human ear cannot hear any higher quality than CD?

This post has been edited by Grand Dizzy: Feb 5 2006, 01:11
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AndyH-ha
post Feb 5 2006, 03:34
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The most profound differences are not higher frequency response but the effect of the anti-aliasing filters on the frequencies that can be heard. At 96kHz sampling rate the filter effects at 20kHz are somewhat different than when sampling at 44.1kHz. This is a measurable result but whether or not many, or any, people can truly hear it does not seem to have been established by impartial studies. Or maybe it has but the results don't please the people who promote higher sampling frequencies.

The results of those filters are not exactly a given anyway. There are a variety of ways, and many graduations of these, to accomplish the filtering, with different final effects. There are those who claim to prefer digital recording and playback with no such filters, regardless of the images. There are some who have developed special (non-conventual) processes to produce CD standard files from high sample rate masters without anti-aliasing filters, but also without the stronger images that would result from the more normal downsampling minus the filters. Most or all of this is too subtle for anyone to really hear it if they don't know to which form they are listening at the moment.

There are some limited studies that indicate people may be able to respond to frequencies well above 20kHz (maybe as high as 30-35kHz) under very limited conditions. The mechanisms seem to be something other than hearing, such as direct conduction of vibrations through parts of the body. Such high frequencies are very rarely strong enough to accomplish this.
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gameplaya15143
post Feb 5 2006, 03:38
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call him/her on it... make em prove it to you

it is possible that they can hear a difference, but i think it is highly unlikely

record something at 96khz... and make a copy at 44.1khz (resample with something good like SSRC) keep the bitdepth the same


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Grand Dizzy
post Feb 5 2006, 21:48
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Andy, I didn't realise antialiasing filters were applied to music. Do all audio players use antialiasing?

If no one can hear higher than 44.1KHz, what is antialiasing needed for? Surely it would be too fast to tell the difference between the original aliased sound and the smoother antialiased sound?
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AndyH-ha
post Feb 6 2006, 03:17
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MOST audio players (as part of the DAC) use anti-alaising filters. The image is reflected back down from the Nyquist limit. That means it gets mixed into the music. You generally can't detect it as something separate, on its own, it just adds stuff that should not be there.

It comes in reverse order. The lower the signal frequency, above the Nyquist limit, the higher the frequency of its image. Which also means, the higher the frequency above the Nyquist limit, the lower the frequency of its image.

Nyquist limit = 22,050 Hz at 44.1KHz sampling rate
image of audio at 24kHz appears at
(22,050 Hz - (24,000 - 22,050) = 1950Hz) = 20,100 Hz
image of audio at 28kHz appears at
(22,050 Hz - (28,000 - 22,050) = 5950Hz) = 16,100 Hz
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Grand Dizzy
post Feb 6 2006, 13:49
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Duhh... sorry, that all went completely over my head!
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enry2k
post Feb 6 2006, 14:06
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I know that oversampling in A/D and D/A converters are employed to both spread the same amount of noise over a wider spectrum (noise shaping) and to avoid aliasing, even with 44.1 khz.

Digital Finite Impulse Responce filters can be used to filter the signal.

Enrico
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Hollunder
post Feb 6 2006, 15:53
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QUOTE (Grand Dizzy @ Feb 6 2006, 01:49 PM)
Duhh... sorry, that all went completely over my head!
*


It's not half as hard to understand as it sounds.

the Nyquist-thing:

you need double the frequency (samples per second here) to record a certain frequency (frequency of the signal wave)

If you want to record a Signal that's 800 Hz you need 1600 Hz (practicaly a bit more afaik, but this is theoretical) to record it

If you'd record a Signal of 800 Hz with a samlingrate of 1500 Hz (1500/2 = 750) there would be 50 Hz too much, those "come back down" and appear at 750 Hz in this case. This is a not really recorded signal and unwanted, the so-called Alias-signal.



So if I'm right a Anti-alias Filter is a low-pass filter (lets signals below a certain frequency go through) that cuts off every frequenzy above half the sample frequency to avoid alias-signals.

The Problem is that those filters can't be "perfect" but if you apply them at a higher sampling-frequency the result will be closer to perfect and might sound a tiny bit better, but that's unaudible for nearly everyone.

hope I didn't tell something wrong and also hope that it's a bit clearer now,
and sorry for my bad english
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Grand Dizzy
post Feb 6 2006, 23:40
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Oh I think I get it.

It's a lot like picture resolution. In order to resolve a certain level of detail, you need ideally more than double the number of lines than the intended resolution. Ideally as many as possible. If your resolution is too low, the detail blurs together to form new colours that were never part of the original image.
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Hollunder
post Feb 7 2006, 12:58
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right, it's principialy the same

I found a nice explaination somewhere a few days ago, but I think there's no need for searching since you know how it works.

The main reasons for the higher rate are quite good explained (I think) in AndyH-has post below your original question.
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krabapple
post Feb 7 2006, 17:48
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QUOTE (Grand Dizzy @ Feb 6 2006, 07:49 AM)
Duhh... sorry, that all went completely over my head!
*


simply put:
frequencies so high that you can't hear them, produce digital conversion artifacts in the range you *can* hear. This phenomenon is called 'aliasing'.

Antialiasing filters block out those artifacts.
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hdante
post Feb 7 2006, 18:06
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QUOTE (krabapple @ Feb 7 2006, 02:48 PM)
QUOTE (Grand Dizzy @ Feb 6 2006, 07:49 AM)
Duhh... sorry, that all went completely over my head!
*


simply put:
frequencies so high that you can't hear them, produce digital conversion artifacts in the range you *can* hear. This phenomenon is called 'aliasing'.

Antialiasing filters block out those artifacts.
*



Greetings !

I was just Googling about this right now. There's a site that says that anti-aliasing filters are already good enough at 44 KHz. You shouldn't probably hear the difference because of filter problems. The matter seems to be simpler than that. For example, take a violin and a cello. They may produce faint harmonics at ~ 30 KHz (let's say they the former has one higher peak at 30 KHz and the latter, at 32 KHz). If you listen to them (that is, nothing to do with recording), you may hear a 2KHz beating. However, when you record them, you would do that separately. Record the violin at 44 KHz and you'll lose that important peak. Then record the cello and you'll lose that other important peak. Now mix them together: there's no 2 KHz beating ! If you recorded them together you could sample at 44 KHz and still get the beating. Since you don't, then you'll have to record at least at ~65 KHz. 96 KHz would then be a convenience sampling rate (ie 2x48 KHz).

That's what I read. I'm no speciallist on that. You may google for it also.

Henrique Dante de Almeida
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SebastianG
post Feb 7 2006, 19:29
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QUOTE (hdante @ Feb 7 2006, 06:06 PM)
[...] For example, take a violin and a cello. They may produce faint harmonics at ~ 30 KHz (let's say they the former has one higher peak at 30 KHz and the latter, at 32 KHz). If you listen to them (that is, nothing to do with recording), you may hear a 2KHz beating. [...]
*


Why may I hear something like that ?

Sebi
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RockFan
post Feb 7 2006, 20:38
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QUOTE (Grand Dizzy @ Feb 4 2006, 04:10 PM)
I recently met a musician who claims he can quite easily hear the difference between 44.1KHz and 96KHz.

This shocked me a little because I'd always been told that the human ear cannot hear any higher quality than CD (44.1KHz) quality.

So... was this guy just lying (or fooled by his senses), or was I being lied to when I was told the human ear cannot hear any higher quality than CD?
*


Hi,

this is a 2Khz (stereo) square wave, represented in 16/44.1 PCM



A square wave is actually composed of a sine-wave fundamental (of 2KHz in this case) with an infinite number of it's odd order harmonics folded back into it (3rd, 5th, 7th etc). In fact a'perfect' squarewave doesn't exist, it would have an infintely short rise and decay for each cycle, requiring an infinite number of harmonics, but the more (higher-frequency) of those odd-orders you add, the closer you get to one. This is how the 'edges' needed for digital data transmission are created on such things as analogue phone lines.

This waveform obviously doesn't exist in 'nature', there's no way of producing it acoustically, transmitting it through the air and capturing it with a microphone, it has to be synthesized.

So, this sythesized 2KHz sqaurewave actually has harmonic components extending to 100's of KHz and beyond. Strange but true. You can't 'hear' them, but they're there, they create theis waveform by reinforcing or attenuating the original 2KHz sine.

To actually reproduce this wave 'perfectly' in the analogue domain as the output of a DAC (that is, downstream of it's anti-aliasing filter) is as 'impossible' as the waveform itself is. Filter ringing and phase-shifting between frequencies will produce various effects such as rippling which can be seen graphically if the output is re-captured digitally or monitored in real-time on an oscilloscope.

Now as it happens almost *all* musical instruments produce sound swith harmonic components extending to 40KHz, 50KHz and beyond. Some, such as muted brass produce very substantial pressure levels indeed at these frequencies.

Can we hear them, or sense them in any way? Doubtful (even if you go with the putative non-aural mechanisms some suggest).

BUT they are nonetheless intrinsic to the waveform which results when they are captured - it is *irrelevent* that we cannot 'hear' them, or that the recording hardware or digital protocol is 'band-limited'.

On playback of a recording, the same digital-filtering effects which can be seen graphically in the output of the simple, mathematical square-wave will affect the ultrasonic components of musical instruments and *will* at the very least have an effect on timbre, from innocuous to possibly ear-shredding.

Please don't anybody tell me they *havn't* at some heard point heard a recording of violin or trumpet playing on a CD-based system that didn't make them want to clap their hands over their ears!

I'm not at all surprised to hear that a musician says he/she can hear their instrument reproduced more faithfully with higher sampling rate PCM.

Higher sampling rate = much more benign filtering and more realistic music.

R.

>>edits - yptos as usual.

This post has been edited by RockFan: Feb 7 2006, 20:52
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krabapple
post Feb 7 2006, 22:11
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QUOTE (RockFan @ Feb 7 2006, 02:38 PM)
Now as it happens almost *all* musical instruments produce sound swith harmonic components extending to 40KHz, 50KHz and beyond. Some, such as muted brass produce very substantial pressure levels indeed at these frequencies.

Can we hear them, or sense them in any way? Doubtful (even if you go with the putative non-aural mechanisms some suggest).


OK., so far so good, though whenever someone brings up square waves in a discussion of digital, I expect the worst.

QUOTE
BUT they are nonetheless intrinsic to the waveform which results when they are captured  - it is *irrelevent* that we cannot 'hear' them, or that the recording hardware or digital protocol is 'band-limited'.


Wrong. If we can't hear them -- or their effects in the audible range -- then they are indeed irrelevant to our audio experience.

QUOTE
On playback of a recording, the same digital-filtering effects which can be seen graphically in the output of the simple, mathematical square-wave will affect the ultrasonic components of musical instruments and *will* at the very least have an effect on timbre, from innocuous to possibly ear-shredding.


At the very most, that is *possible*, but not *certain* to happen, nor is it at all certain that whatever effect you hear on timbre you hear, is due to the sampling rate. You'd have to rule out lots of other causes. Generally the biggest 'hit' the accuracy of a digital recording takes is when the signal passes through the mic and the speakers -- the electromechanical parts of the chain. These are by far the least linear.

QUOTE
Please don't anybody tell me they *havn't* at some heard point heard a recording of  violin or trumpet playing on a CD-based system that didn't make them want to clap their hands over their ears!



Please tell me that that you don't consider this proof that Redbook standard *necessarily* affects the timbre of a recording. (I've heard all-analog recordings that make me want to cover my ears, btw.)

QUOTE
I'm not at all surprised to hear that a musician says he/she can hear their instrument reproduced more faithfully with higher sampling rate PCM.


I'm not surprised that pepoel claim all sorts of things. I'm far more surprised when they've actually tested those claims properly. Because that;s so very rare.


QUOTE
Higher sampling rate = much more benign filtering and more realistic music.



It can mean that. Doesn't necessarily mean that. It's down to how well the filtering is implemented.

.

This post has been edited by krabapple: Feb 7 2006, 22:17
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hdante
post Feb 7 2006, 22:27
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QUOTE (SebastianG @ Feb 7 2006, 04:29 PM)
QUOTE (hdante @ Feb 7 2006, 06:06 PM)
[...] For example, take a violin and a cello. They may produce faint harmonics at ~ 30 KHz (let's say they the former has one higher peak at 30 KHz and the latter, at 32 KHz). If you listen to them (that is, nothing to do with recording), you may hear a 2KHz beating. [...]
*


Why may I hear something like that ?

Sebi
*



It was just an example. If you were talking that the 2 KHz was a mistake, I'm sorry, it should be 1 KHz. If not, it's because of the following. I supposed that there would be an instrument which would produce a significant harmonic at 30 KHz (actually this is true, for example, for violins and flutes), and another that would produce it at 32 KHz. Since those frequencies are actually a pressure in the same medium (that is the air and then your ear), the expansions and compressions generated by the instruments will add to each other some times and cancel each other some other times at a rate of 1 KHz. Mathematically, cos(32KHz)+cos(30KHz) = 2*cos(31KHz)*cos(1KHz). Add the time in equation and you have a 1 KHz harmonic with a variable intensity of 2*cos(31KHz*t). In practice, you should have a few harmonics for every instrument in this region. For most of them, they will be so faint, that you won't ever notice it. The already cited instruments, however, are known to cause audible beating which enrich the listening experience. Unfortunately I have no link to that, except one that also claims this is true, but doesn't cite sources either :-/.

http://www.dvdsoftwareguide.com/all-about-dvd-4-guide.html

One should hope, thus, that every recording is made at very high sampling rates. After they are mixed and filtered with high quality equipment, they may be safely downsampled to human listening limits again.
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hdante
post Feb 7 2006, 22:37
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QUOTE (RockFan @ Feb 7 2006, 05:38 PM)
On playback of a recording, the same digital-filtering effects which can be seen graphically in the output of the simple, mathematical square-wave will affect the ultrasonic components of musical instruments and *will* at the very least have an effect on timbre, from innocuous to possibly ear-shredding.

Please don't anybody tell me they *havn't* at some heard point heard a recording of  violin or trumpet playing on a CD-based system that didn't make them want to clap their hands over their ears!

R.

>>edits - yptos as usual.
*


Again, I don't think that filtering artifacts are relevant. The issues only happen with bugged filtering. Recent equipment shouldn't cause audible artifacts. Concerning the bad violin or trumpet, see the other discussion.

http://www.digitalprosound.com/Htm/SoapBox/soap2_Apogee.htm

Henrique Dante de Almeida
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sven_Bent
post Feb 7 2006, 23:04
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@gangran dizzy

i hear alot of audiophiles around me claiming alot.

They do have a hard time proving it... atctually they have failed all there claims when doing probally scientific correct testing.


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RockFan
post Feb 7 2006, 23:21
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QUOTE (krabapple @ Feb 7 2006, 01:11 PM)
Wrong.  If we can't hear them -- or their effects in the audible range -- then they are indeed irrelevant to our audio experience.

*


I don't want to get into yet another nit-picking session on this.

You simply havn't grasped the point I'm making.

If the intrinsic ultrasonic content of our squarewave (of whatever frequency) cannot be maintained with 'time-domain coherency' on playback (within a band-limited playback-system), then QED - neither can any other *captured* sound/waveform which is defined by it's ultrasonic components.

The timbre of many, if not most, musical instruments *is* defined by this ultrasonic content. I'm amazed how few people haven woken up to this.

As I said - the ability to perceive/hear ultrasonic frequencies or to capture them discreetly with recording equipment is completely *irrelevent*.

Time-domain coherency is the key to realistic reproduction of music - instruments, voices, whatever.

Of course, if we're lsitening to a Stratocaster and an overdriven Marshall stack, this might all be moot.

R.
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WmAx
post Feb 7 2006, 23:39
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QUOTE (RockFan @ Feb 7 2006, 03:38 PM)
Please don't anybody tell me they *havn't* at some heard point heard a recording of  violin or trumpet playing on a CD-based system that didn't make them want to clap their hands over their ears!

I'm not at all surprised to hear that a musician says he/she can hear their instrument reproduced more faithfully with higher sampling rate PCM.

Higher sampling rate = much more benign filtering and more realistic music.

R.

>>edits - yptos as usual.
*


By reading your statement(s) here, one would think you have not paid any attention to the last several discussions on hi-rez audio on hydrogenaudio.org. Please go back and reference these prior discussions.

-Chris
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RockFan
post Feb 7 2006, 23:40
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QUOTE (hdante @ Feb 7 2006, 01:37 PM)
Again, I don't think that filtering artifacts are relevant. The issues only happen with bugged filtering. Recent equipment shouldn't cause audible artifacts. Concerning the bad violin or trumpet, see the other discussion.

http://www.digitalprosound.com/Htm/SoapBox/soap2_Apogee.htm

Henrique Dante de Almeida
*


It really isn't a matter of opinion.

Oversampling filters serve as a panacea for the limited resolution of RB CD (16/44 PCM).

But many people are building completely filterless/non-oversampling DACS. Why, one should be bound to ask?

Here's the rub; OS DACs do sine waves pretty well up to (insert freq; 10KHz?) but are utterly incapable of resolving a squarewave at anything close to this freq.

Non-OS DACs make an unholy mess of sines above 10KHz, but (at least some of them) can do squares at this frequency and beyond.

In a previous discsussion here at HA, someone siad that the the Non-OS DAC's inability to reproduce HF sines meant they were "broken".

Why, then, does the OS DAC's inability to reproduce HF squares not mean they are *edit >* NOT "broken"?

R.

This post has been edited by RockFan: Feb 8 2006, 00:22
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RockFan
post Feb 7 2006, 23:44
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QUOTE (WmAx @ Feb 7 2006, 02:39 PM)
By reading your statement(s) here, one would think you have not paid any attention to the last several discussions on hi-rez audio on hydrogenaudio.org. Please go back and reference these prior discussions.

-Chris
*


I do actually pay attention, and not just to HA.

Do you have a point?

R.
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WmAx
post Feb 7 2006, 23:45
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QUOTE (RockFan @ Feb 7 2006, 06:21 PM)
You simply havn't grasped the point I'm making.


I have grasped your point, as I'm sure the other post has as well: You think that ultrasonic information has some relevance to the audible waveform, though such has not ever been shown in a credible peer-reviewed study. But your premise as presented here is in error. 1. Assuming that analog filters are used for the anti-alias filters, no one has shown the phase distortion that occurs as a result to be audible. 2. Most systems today should be using linear phase digital filtes, in which pase distortion is not even an issue. 3. Since you can only hear frequencies <X, the waveform content >X is not relevant to audibility. You can not hear square waves, for example. You can only hear the sine waves composing the square waves which are <X frequency. The content >X is only relevant to a pretty looking graphical respresentation of the waveform, not to audible parameters. How you think supersonic content has to do with time domain coherancy of the audible waveforms, I have yet to figure out.

-Chris

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RockFan
post Feb 7 2006, 23:51
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QUOTE (RockFan @ Feb 7 2006, 02:21 PM)
Of course, if we're lsitening to a Stratocaster and an overdriven Marshall stack, this might all be moot.
*

I take that back. It might well be just as important.
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mandel
post Feb 7 2006, 23:51
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QUOTE (hdante @ Feb 7 2006, 10:27 PM)
QUOTE (SebastianG @ Feb 7 2006, 04:29 PM)
QUOTE (hdante @ Feb 7 2006, 06:06 PM)
[...] For example, take a violin and a cello. They may produce faint harmonics at ~ 30 KHz (let's say they the former has one higher peak at 30 KHz and the latter, at 32 KHz). If you listen to them (that is, nothing to do with recording), you may hear a 2KHz beating. [...]
*


Why may I hear something like that ?

Sebi
*



It was just an example. If you were talking that the 2 KHz was a mistake, I'm sorry, it should be 1 KHz. If not, it's because of the following. I supposed that there would be an instrument which would produce a significant harmonic at 30 KHz (actually this is true, for example, for violins and flutes), and another that would produce it at 32 KHz. Since those frequencies are actually a pressure in the same medium (that is the air and then your ear), the expansions and compressions generated by the instruments will add to each other some times and cancel each other some other times at a rate of 1 KHz. Mathematically, cos(32KHz)+cos(30KHz) = 2*cos(31KHz)*cos(1KHz). Add the time in equation and you have a 1 KHz harmonic with a variable intensity of 2*cos(31KHz*t). In practice, you should have a few harmonics for every instrument in this region. For most of them, they will be so faint, that you won't ever notice it. The already cited instruments, however, are known to cause audible beating which enrich the listening experience. Unfortunately I have no link to that, except one that also claims this is true, but doesn't cite sources either :-/.

http://www.dvdsoftwareguide.com/all-about-dvd-4-guide.html

One should hope, thus, that every recording is made at very high sampling rates. After they are mixed and filtered with high quality equipment, they may be safely downsampled to human listening limits again.
*



That's really interesting actually. I just created a 96khz wav file with a 30k and a 32k tone and could hear a beat frequency as you say. Though at 2khz not 1khz

Why do you say the hi-res mix may be safely downsampled to 'human hearing limits'? If I resampled the above wav file to 44.1khz I ended up with silence!

This post has been edited by mandel: Feb 7 2006, 23:56
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