Sampling rates higher than 44.1Khz? Options

[Grand Dizzy]

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

[RockFan]

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

[hdante]

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

[RockFan]

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

[krabapple]

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).

Wrong. Resolution is a function of word length (bit depth), not sampling rate.
Oversampling filters were a solution to the difficulty (not impossibility) in implementing excellent filtering at 44.1.

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

Because there's no silly idea that some audiophile won't embrace. There are belt-driven CD players out there too. And no, it's not *many* people doing this. It's a relatively tiny cult, as with most audiophile tweaks. One question to ask is *who* is doing it.


I looked up your posts here and I see you were arguing for cryogenic treatment of wires elsewhere.


I suggest interested parties check out Nika Aldrich's comments on quare waves,
before they disappear from the the Web (they're now cache-only). Or buy his book on 'Digital Audio Explained' -

http://72.14.207.104/search?q=cache:stHyib...us&ct=clnk&cd=1

The bottom line with the point above is that, depending on the quality and design of the filter, 44.1 or 48k are perfectly adequate to COMPLETELY represent any signals under 20k. Thus, for the sake of what is commonly accepted as our ears' hearing ability (from 20Hz to 20kHz) 96kHz recording is totally unnecessary. For further information on the theories of the potential validity of 96kHz, see the topic that I referred to above in my original post. There are indeed some theories that are worth exploring, only one of which is the "psychoacoustic" theory that we can percieve information that our ears are not attributed to being able to hear. I must tell you that this is, I believe, the weakest of all of the theories.

As for the notion that things that we can't hear can affect the things that we can hear, the answer is "no". Defiantly "no". Your ear acts as the same type of filter that we discussed above. If we take a 1kHz sine wave and then add all kinds of processing to it of very high frequencies (50k and such) so that in the end it doesn't really look like a 1kHz sine wave at all, and then put a filter on it that filters out everything over 2kHz, all you'll be left with is your 1kHz sine wave. I don't care how much junk you added. Once you add that 2kHz filter, it's right back to a 1kHz sine wave.

The thing is that your ears work this way also. If you take a clarinet note and add all kinds of garbage at ultra high frequencies to it for the sake of who-knows-what and whatnot, by the time you listen to it, all you'll hear is the clarinet.

If, however, you take the same clarinet and add some eq to it at 2.5k which also induces some wacky 50kHz stuff to happen, it would be incorrect to say that the 50kHz signal is CREATING differences that you can hear. What would be more correct is to say that you are processing the signal at 2.5k and some side effects at 50k, but that your ear won't hear the results of what happened at 50k. All that you'll end up hearing is the clarinet and the change of it at 2.5k. The 50k didn't CAUSE the change. It is a BIPRODUCT of the change, and an inaudible one that will be filtered away.

This post has been edited by krabapple: Feb 8 2006, 00:11

[RockFan]

You bore me.

The last word is yours, please do savour it.

R.