Why can I hear a difference between 32bit 384khz and 24bit 44.1khz?

[Frank Dernie]

A vinyl record, for example, can record up to 15kHz-17kHz. .. Because of the tip of the recording stylus is not small enough to make a clear 'drawing' of frequencies above this. But it does record harmonics of higher frequencies as -periodical- small notches (what they become when projected into the surface of a vinyl master) that when played back ad substance to the sound.

Tape does record these higher frequencies naturally and it does adds them during the playback.

Harmonics occur as part of a sound wave and audio signal in a different fashion (although explainable using the same principles). When a wave occurs it is because it carries certain amplitude which is sufficient to create the next cycle of the same in the longitudinal sense of propagation. Waves do not isolate their energy in a certain portion of the medium but rather they propagate in all direction. A secondary oscillation that occurs from a central one, will carry some of the energy of the main wavelength and will have a very similar frequency value. In the space of the medium between these two, if these carry enough amplitude, some superior harmonics will form. Most sound waves indeed carry higher harmonics. Lower harmonics, always. The difference and correlation of the phase and amplitude between any fundamental and it's harmonics is very specific and no current digital systems are able to capture it. Although this can be remediated to a point using higher sampling rates and more bit depth.

Another part of the story is that there's no feasible way for any current digital system to match their sampling clock or microprocessor clock to a certain value of phase for any audio frequencies. Let's say that you have a tone of a constant frequency coming out of a violin. .. This tone is A 440. This tone is going to acquire any 360 degrees of phase 440 times in one second. No digital system can tell if they started recording at any given value of phase or another. Or, in other words, when the ADC generates a code value, they are not delivering the phase value of the voltage of the audio signal.

(Certainly you can see in many DAW's a beutiful graphic which is somehow an approximation of the voltage value and phase of any signals. But the margin of error is huge, depending how you wanna calculate it, it can be as huge as 600-1200% or more).

Most sound fields are composed by different tones of different frequencies, at different amplitudes. When these frequencies travel together in the same medium and when they reach together the element of a microphone, they are going to superpose to each other 'transiently' at interger points and create numerous intermodulations.

Similarly to the effect of 2 equal tones cancelling or reinforcing each other, any two simultaneous frequencies will alter each other in some way even if they are far apart in the spectrum. And the values of the resulting intermodulations are very phase-specific (and not always a factorial of 2x2x3x3x5x5x7x7).

Oftentimes this transients harmonics as well as the natural harmonic could be 'seen' in a graphic as appearing for a couple of miliseconds every some five or six, ten or more miliseconds.

And the precise reproduction of this harmonic content is what makes a recording to sound more natural. Most digital systems will cut the frequency spectrum at 20kHz. No digital systems are able to deliver a neat reproduction of the harmonic content of a recording. Although, it does not means it's all lost. Loudspeakers and preamps will -to some extent- re-create artificially the lost harmonic content by reinforcement of fundamentals.

And anyways the sound quality is not everything. I prefer to listen to my favorite song at a listenable level of compression than a recording I don't like nevermind what level of resolution.

Almost everything you write here is completely and absolutely wrong.
Some of it is vaguely right but you haven't understood it fully. Where the royal f*ck did you "learn" this stuff?
I am gobsmacked

 

[threni]

Apoligies for entering in such a smug and arrogant domain. I hope the couple of comments I made help you break your cycle of self redundancy.

I do have things to do and no time for people being hard of each other.

Sweetest goodbye!

So you've found a problem in Nyquist Theorem? I look forward to reading your paper!

 

[SIY]

A vinyl record, for example, can record up to 15kHz-17kHz. .. Because of the tip of the recording stylus is not small enough to make a clear 'drawing' of frequencies above this. But it does record harmonics of higher frequencies as -periodical- small notches (what they become when projected into the surface of a vinyl master) that when played back ad substance to the sound.

Tape does record these higher frequencies naturally and it does adds them during the playback.

Harmonics occur as part of a sound wave and audio signal in a different fashion (although explainable using the same principles). When a wave occurs it is because it carries certain amplitude which is sufficient to create the next cycle of the same in the longitudinal sense of propagation. Waves do not isolate their energy in a certain portion of the medium but rather they propagate in all direction. A secondary oscillation that occurs from a central one, will carry some of the energy of the main wavelength and will have a very similar frequency value. In the space of the medium between these two, if these carry enough amplitude, some superior harmonics will form. Most sound waves indeed carry higher harmonics. Lower harmonics, always. The difference and correlation of the phase and amplitude between any fundamental and it's harmonics is very specific and no current digital systems are able to capture it. Although this can be remediated to a point using higher sampling rates and more bit depth.

Another part of the story is that there's no feasible way for any current digital system to match their sampling clock or microprocessor clock to a certain value of phase for any audio frequencies. Let's say that you have a tone of a constant frequency coming out of a violin. .. This tone is A 440. This tone is going to acquire any 360 degrees of phase 440 times in one second. No digital system can tell if they started recording at any given value of phase or another. Or, in other words, when the ADC generates a code value, they are not delivering the phase value of the voltage of the audio signal.

(Certainly you can see in many DAW's a beutiful graphic which is somehow an approximation of the voltage value and phase of any signals. But the margin of error is huge, depending how you wanna calculate it, it can be as huge as 600-1200% or more).

Most sound fields are composed by different tones of different frequencies, at different amplitudes. When these frequencies travel together in the same medium and when they reach together the element of a microphone, they are going to superpose to each other 'transiently' at interger points and create numerous intermodulations.

Similarly to the effect of 2 equal tones cancelling or reinforcing each other, any two simultaneous frequencies will alter each other in some way even if they are far apart in the spectrum. And the values of the resulting intermodulations are very phase-specific (and not always a factorial of 2x2x3x3x5x5x7x7).

Oftentimes this transients harmonics as well as the natural harmonic could be 'seen' in a graphic as appearing for a couple of miliseconds every some five or six, ten or more miliseconds.

And the precise reproduction of this harmonic content is what makes a recording to sound more natural. Most digital systems will cut the frequency spectrum at 20kHz. No digital systems are able to deliver a neat reproduction of the harmonic content of a recording. Although, it does not means it's all lost. Loudspeakers and preamps will -to some extent- re-create artificially the lost harmonic content by reinforcement of fundamentals.

And anyways the sound quality is not everything. I prefer to listen to my favorite song at a listenable level of compression than a recording I don't like nevermind what level of resolution.

There’s a remarkable amount of complete nonsense packed into this post. Seriously, you have significant learning to do.

 

[Ennis]

Great! Than surely you can tell us what a sound engineer does with ultrasonics? And how do you verify that what you did to them has any effect?

Oh boy...

 

[threni]

Where the royal f*ck did you "learn" this stuff?

I'm going with either Head-fi or SBAF.

 

[Ennis]

Or you could just read the thread to realise this had nothing to go with transients, supersonic content or golden ears, but with the oversampling process.

... It's about sound and audio quality...

 

[threni]

A vinyl record, for example, can record up to 15kHz-17kHz. .. Because of the tip of the recording stylus is not small enough to make a clear 'drawing' of frequencies above this. But it does record harmonics of higher frequencies as -periodical- small notches (what they become when projected into the surface of a vinyl master) that when played back ad substance to the sound.

Tape does record these higher frequencies naturally and it does adds them during the playback.

Harmonics occur as part of a sound wave and audio signal in a different fashion (although explainable using the same principles). When a wave occurs it is because it carries certain amplitude which is sufficient to create the next cycle of the same in the longitudinal sense of propagation. Waves do not isolate their energy in a certain portion of the medium but rather they propagate in all direction. A secondary oscillation that occurs from a central one, will carry some of the energy of the main wavelength and will have a very similar frequency value. In the space of the medium between these two, if these carry enough amplitude, some superior harmonics will form. Most sound waves indeed carry higher harmonics. Lower harmonics, always. The difference and correlation of the phase and amplitude between any fundamental and it's harmonics is very specific and no current digital systems are able to capture it. Although this can be remediated to a point using higher sampling rates and more bit depth.

Another part of the story is that there's no feasible way for any current digital system to match their sampling clock or microprocessor clock to a certain value of phase for any audio frequencies. Let's say that you have a tone of a constant frequency coming out of a violin. .. This tone is A 440. This tone is going to acquire any 360 degrees of phase 440 times in one second. No digital system can tell if they started recording at any given value of phase or another. Or, in other words, when the ADC generates a code value, they are not delivering the phase value of the voltage of the audio signal.

(Certainly you can see in many DAW's a beutiful graphic which is somehow an approximation of the voltage value and phase of any signals. But the margin of error is huge, depending how you wanna calculate it, it can be as huge as 600-1200% or more).

Most sound fields are composed by different tones of different frequencies, at different amplitudes. When these frequencies travel together in the same medium and when they reach together the element of a microphone, they are going to superpose to each other 'transiently' at interger points and create numerous intermodulations.

Similarly to the effect of 2 equal tones cancelling or reinforcing each other, any two simultaneous frequencies will alter each other in some way even if they are far apart in the spectrum. And the values of the resulting intermodulations are very phase-specific (and not always a factorial of 2x2x3x3x5x5x7x7).

Oftentimes this transients harmonics as well as the natural harmonic could be 'seen' in a graphic as appearing for a couple of miliseconds every some five or six, ten or more miliseconds.

And the precise reproduction of this harmonic content is what makes a recording to sound more natural. Most digital systems will cut the frequency spectrum at 20kHz. No digital systems are able to deliver a neat reproduction of the harmonic content of a recording. Although, it does not means it's all lost. Loudspeakers and preamps will -to some extent- re-create artificially the lost harmonic content by reinforcement of fundamentals.

And anyways the sound quality is not everything. I prefer to listen to my favorite song at a listenable level of compression than a recording I don't like nevermind what level of resolution.

If you want to appear maybe just a little less foolish you might want to watch this video. It's about 20 minutes long so watch all of it. I've bookmarked it at the point where it teaches you why your comments about "started recording at any given value of phase" are objectively wrong.

 

[tmtomh]

Our modern science does not know the exact efficiency of the air or any mediums for each frequency. Although a superior harmonic can be identified or in theory found as as x1.5 , x2 or x3 times the frequency but the same amplitude as the fundamental (in theory anywhere from 1.1875 and on ... times the frequency, and still you can find harmonic noise in the range from 1 to 1.1875).

And you don't hear the ultrasonic harmonics, what you hear is the harmonic modulation of the fundamental in the hearable range, making it sound wider and fuller.

I don't want to be rude, truly, but I don't know how else to say it: what you have written above is nonsense. Not nonsense as in, "this is a bad idea and I don't respect your view," but nonsense as in, "this does not make any sense."

"The efficiency of air" is a vague and unclear term, but in any event the way that soundwaves travel through air is well-understood, and the real-world impact of air circulation, turbulence, temperature, humidity, and so on is completely irrelevant to the question of digital sample rates.

"Mediums for each frequency" is either another way of referring to "the efficiency of air" or is a nonsense phrase. It is well-understood that different frequencies have different levels of absorption and reflectivity with different thicknesses, densities, and structures of material they come into contact with. Again, that has nothing whatsoever to do with digital sample rates.

As for harmonics, we do in fact hear "the harmonic modulation of the fundamental in the hearable range," and yes of course, harmonics alter the sound compared to if we just hear the fundamental. That has nothing to do with sample rate. A 20kHz 2nd harmonic of a 10kHz fundamental can be encoded and reconstructed by a 441kHz sample-rate system just as well as it can by a 192kHz sample-rate system.

 

[tmtomh]

... It's about sound and audio quality...

It is indeed - and nothing you are bringing up here has anything to do with the sound and audio quality produced by different digital sample rates.

 

[sq225917]

And vinyl records 'are' capable of holding sound above 20khz, if, it's there in the original instrument, and the microphones and recording chain aren't bandwidth limited electronically or physically due to the mass of the cutter head.

Some playback carts can comfortably reach 25khz if there was anything but noise there to playback. It's all down to tip mass.

 

[levimax]

And vinyl records 'are' capable of holding sound above 20khz, if, it's there in the original instrument, and the microphones and recording chain aren't bandwidth limited electronically or physically due to the mass of the cutter head.

Some playback carts can comfortably reach 25khz if there was anything but noise there to playback. It's all down to tip mass.

Quad LP's have rear channel information stored at over 40 Khz

 

[voodooless]

Oh boy...

That’s not a whole lot. Sounds like ultrasonics are not important after all..

 

[krabapple]

Apoligies for entering in such a smug and arrogant domain. I hope the couple of comments I made help you break your cycle of self redundancy.

I do have things to do and no time for people being hard of each other.

Sweetest goodbye!

As you apparently came here to promote pseudoscientific babble, I applaud your exit.

 

[krabapple]

Quad LP's have rear channel information stored at over 40 Khz

CD-4 LP carrier signal ranges from 30 to 50K. 'Shibata' styli can track it, and then a demodulator renders the quad output. But vinyl playback wear takes its toll, degrading the carrier relatively quickly, unless the vinyl is specially formulated (and even then...)

None of which has anything to do with the pseudoscience Ennis is spouting.

 

[tomelex]

I'm pretty skeptical myself, but honestly I can hear a difference! Is it my brain playing that much tricks on me?

If for one simple thing, you do not position your head at the exact same spot each time, a difference of a few inches can result in bass response changes that are audible, I am not denying you hear differences, but the science to the do the experiment correctly is pretty intense.

 

[Yuhasz01]

This guy is a bat in regards to hearing..

 

[Emlin]

This guy is a bat in regards to hearing..

Probably just bats.

 

[mansr]

Probably just bats.

In the belfry.