[gregorio]

HOW are you determining that?

Roughly 130 years of scientific/controlled testing countless thousands of subjects. In fact, the average limit for an adult human is 16kHz.

Sampling Theorem requires PERFECT filters.

The Sampling Theorem just requires a band limited signal. In practice we cannot create an absolutely perfect filter with a zero transition band. That’s why the CD standard was set at a sampling rate of 44kHz rather than 40kHz, it allows 2kHz above the 20kHz hearing limit for a filter. The filter still does not need to be absolutely perfect, it just needs to exceed human hearing.

G

 

[eq1849]

Roughly 130 years of scientific/controlled testing countless thousands of subjects. In fact, the average limit for an adult human is 16kHz.

The Sampling Theorem just requires a band limited signal. In practice we cannot create an absolutely perfect filter with a zero transition band. That’s why the CD standard was set at a sampling rate of 44kHz rather than 40kHz, it allows 2kHz above the 20kHz hearing limit for a filter. The filter still does not need to be absolutely perfect, it just needs to exceed human hearing.

G

You are now arguing the Sampling Theorum allows an imperfect bandwidth limited signal?

 

[gregorio]

You are now arguing the Sampling Theorum allows an imperfect bandwidth limited signal?

No, I am arguing that the upper limit of the sampling theorem, EG. 22.05kHz with a 44.1kHz sampling rate cannot be achieved. In practice only up to about 20kHz can be achieved, to allow roughly 2kHz for the filter’s transition band.

G

 

[71 dB]

Hi-Res is closer to the original analog audio signal when compared to CD (even the difference is small)

Hi-res can be closer to original analog audio signal if there are stuff above about 20 kHz in the analog signal (not so clear there is anything else than noise). Even then it doesn't matter, because people don't hear above 20 kHz (apart maybe some extremely rare cases). CDs sound bad when the music is produced, mixed and mastered badly. CDs sound awesome when the music is produced, mixed and mastered very well. That's it. Hi-res is just a marketing gimmick.

 

[71 dB]

Sampling Theorem requires PERFECT filters.

Fortunately human ears are somewhat less demanding than the sampling theorem.

 

[sunjam]

Let's define a few items here one more time to help us to discuss :

original audio signal <=== Let's call it A_original

digitized input from A_original in CD format <=== Lets' call it D_CD. i.e. D_CD=ADC_CD(A_original) where ADC_CD( ) is the 44.1/16 ADC process

digitized input from A_original in Hi Res format <=== Lets' call it D_Hi_Res. i.e. D_Hi_Res=ADC_Hi_Res (A_original) where ADC_Hi_Res ( ) is the 768/32 ADC process

reconstructed audio signal from D_CD <=== Let's call it A_from_CD,

reconstructed audio signal from D_Hi_Res <=== Let's call it A_from_Hi_Res,

=================================================================

The Monty's article/video claim "The analog signal can be reconstructed losslessly, smoothly, and with the exact timing of the original analog signal" ("fact 2"):

His statement means clearly that A_from_DAC_CD = A_original <=== this is wrong. It is not factual, hence "fact 2" is not factual

Of course, you can attempt to re-interpret his claim as "sounds similar" or "sounds close to" the original audio signal.
However, most people know the meaning of "losslessly", "smoothly" and "exact timing". People know he doesn't mean "sounds similar" or "sounds close to"

=================================================================

Due to the lack of the perfect filter, the audio signal reconstruction process is imperfect for both CD and HiRes

However, Hi-Res has more sampled points to help to recreate the orignal audio signal during the imperfect reconstruction process and less quantization noise due to higher bitdepth.

As a result, the difference between A_from_Hi_Res and A_original is smaller than the difference between A_from_CD and A_original <=== Hi-Res is better

You can argue A_from_CD is good enough as it is close to A_original but A_from_Hi_Res is better

 

[71 dB]

You can argue A_from_CD is good enough as it is close to A_original but A_from_Hi_Res is better

We can say hi-res is better for dogs, cats and bats, but I do NOTHING with that advantage, because I am a 53 years old human and even CD offers frequencies beyond my hearing...

 

[eq1849]

=

No, I am arguing that the upper limit of the sampling theorem, EG. 22.05kHz with a 44.1kHz sampling rate cannot be achieved. In practice only up to about 20kHz can be achieved, to allow roughly 2kHz for the filter’s transition band.

G

With 24/44.1 I think the point is on more solid ground...though I think up to 44.1khz it's not going to be infinitely, mathematically perfect even in 24bits.

16/44.1...given the errors actually produce noise artifacts (due to lack of steps, as there are only ~65,000 quantization steps to measure the amplitude of the voltage with) I think that the point would harder to defend.

Fortunately human ears are somewhat less demanding than the sampling theorem.

I don't think we should assume small differences do not matter regarding this stuff.

The brain has to be fooled, or the illusion is off.

 

[eq1849]

We can say hi-res is better for dogs, cats and bats, but I do NOTHING with that advantage, because I am a 53 years old human and even CD offers frequencies beyond my hearing...

Equating high res to just having inaudible frequencies beyond 16bit is basically a False Equivalence Fallacy.

 

[71 dB]

Equating high res to just having inaudible frequencies beyond 16bit is basically a False Equivalence Fallacy.

Sure, since bit depth has nothing to do with bandwidth.

 

[71 dB]

I don't think we should assume small differences do not matter regarding this stuff.

The brain has to be fooled, or the illusion is off.

Many audiophiles get fooled by snake-oil sellers. For me the illusion is off thou...

 

[gregorio]

Let's define a few items here one more time to help us to discuss :
original audio signal <=== Let's call it A_original
digitized input from A_original in CD format <=== Lets' call it D_CD. i.e. D_CD=ADC_CD(A_original) where ADC_CD( ) is the 44.1/16 ADC process
digitized input from A_original in Hi Res format <=== Lets' call it D_Hi_Res. i.e. D_Hi_Res=ADC_Hi_Res (A_original) where ADC_Hi_Res ( ) is the 768/32 ADC process

Why don’t you answer the question, do you really not know what 120dB below 90dBSPL is? If not, how can we take anything you say seriously?

While waiting let’s deal with your latest BS: Why do you want to define two items, one of which hasn’t been standard practice for about 30 years and the other that doesn’t exist? But OK, let go with it.

The Monty's article/video claim "The analog signal can be reconstructed losslessly, smoothly, and with the exact timing of the original analog signal" ("fact 2"):His statement means clearly that A_from_DAC_CD = A_original <=== this is wrong. It is not factual, hence "fact 2" is not factual

Not it is NOT wrong, just repeating over and over that it’s wrong does not make it wrong. Monty actually demonstrated it was correct and even the evidence you yourself have cited demonstrates it’s not wrong. How is your false claim even vaguely logical, let alone critical thinking, it’s just completely nuts??

Due to the lack of the perfect filter, the audio signal reconstruction process is imperfect for both CD and HiRes
However, Hi-Res has more sampled points to help to recreate the orignal audio signal during the imperfect reconstruction process and less quantization noise due to higher bitdepth.

No it does not! Oversampling in a typical modestly priced DAC has way more sampling points than your 768/32 imaginary hi-res, so even using your warped understanding digital audio you are wrong!! This is just funny now!

As a result, the difference between A_from_Hi_Res and A_original is smaller than the difference between A_from_CD and A_original <=== Hi-Res is better

No it is not. The difference between the original and A_from_Hi_res is absolutely massive because you’ve defined the hi-res as the result of a 768/32 ADC process. There are no pro/studio audio ADCs that support that format, so A_from_Hi_res is infinitely worse than A_from_CD because it’s no signal at all, it does not exist! Talk about nuts!!

Now answer the simple question put to you, what is 120dB below 90dB SPL?

I don't think we should assume small differences do not matter regarding this stuff.

We don’t assume that, in fact quite the opposite.

G

 

[theveterans]

What is more accurate?
A) a 44.1khz signal upsampled by the DAC to its max 16fs of 705.6khz
B) a native 705.6khz signal fed into the DAC, bypassing the DAC's interpolation algorithm/up sampling

Depends on the digital filter used on the 705.6 KHz file. HQPlayer can use Sinc-M (brickwall filter) to feed 705.6 (16fs) to a DAC that uses a gentler FIR filter and in this ONLY that B is more accurate than A theoretically. In practice, most likely you won't be able to tell the difference unless you have goldensound ears and lots of practice

 

[sunjam]

Why don’t you answer the question, do you really not know what 120dB below 90dBSPL is? If not, how can we take anything you say seriously?

Certainly! Let’s break down the concept of Sound Pressure Level (SPL) and calculate the value you’re interested in.
(all the log are base 10)

1. Sound Pressure Level (SPL):
   • SPL is a logarithmic measure of the effective pressure of a sound relative to a reference value.
   • It is expressed in decibels (dB).
   • The commonly used reference sound pressure in air is 20 μPa, which is often considered the threshold of human hearing.
2. Calculating SPL Difference:
   • You mentioned “120 dB below 90 dBSPL.” Let’s find out what this means:
     • The difference between two SPL values is given by:
       SPL difference (dB)=20⋅log(P1/P0)
       where:
       • P1 represents the measured sound pressure (in pascals).
       • P0 represents the reference sound pressure (typically around 20 μPa).
3. Given Information:
   • We want to find the SPL difference when the measured sound pressure is 120 dB below the reference level of 90 dBSPL.
   • Let’s convert the reference level to actual sound pressure:
     • P0 = 20 * 10^(90 / 20) μPa
     • P0 = 0.632455532 Pa
4. Calculation:
   • Let’s find the measured sound pressure P1:
     • P1 = 0.632455532 * 10^ (-120 / 20)
     • P1 = 0.632455532 μPa
5. Result:
   • The measured sound pressure is 0.632455532 μPa when it is 120 dB below the reference level of 90 dBSPL.

=================

I am not an audio science expert (but I know what is pseudo science). Please let me know if my calculation above is not correct.

I am curious why you insist to have me to do the calculation? It is done. What you want to show/prove?

To me, our discussion has nothing to do with the correctness of my calculations above.

If you want to show that I don't know what I am talking about if I cannot calculate it correctly, then I would imagine you don't know the sampling theory as you don't know it works only under ideal situation.

Real scientist knows the limitations of their experiments.
Not real scientist (aka pseudo scientist) would take 0.632455532 μPa = 0 μPa and claim they are the same, or in their language "exactly the same"

==================

To be honest, I really what to know your defintion of the word "perfect" and "exactly"

Question 1:
Do you agree that 0.632455532 μPa and 0 μPa are exactly the same?

Question 2:
Based on your reply regarding "perfect", it seems that your defintion of "perfect" is not perfect (pun indeed). People would consider perfect is perfect. There is nothing known as "absolutely perfect" and "perfect but not absolutely". To my trained eyes, these "absolutely / not absolutely" are the "art of pseudo science".

You said "absolutely perfect" below. Does it mean that there is something known as "perfect but not absolutely" in your mind?

The Sampling Theorem just requires a band limited signal. In practice we cannot create an absolutely perfect filter with a zero transition band.

Did you apply the concept of "perfect but not absolutely" in the following reply?

Yes, we did discuss it earlier and you are ignoring that discussion, how is that logical? Perfect can be defined as no audible artefacts, Ie. “Audibly perfect” or “audibly transparent” when using the term “perfect” that is what is typically meant.

Feel free not to answer if you feel unease about these questions. It is just some food for thought for me (and anyone who are interested in critical thinking)

 

[sunjam]

Sampling Theorem requires PERFECT filters.

Most people who believe in "Hi-Res is useless" have a different definition of PERFECT. They have the concept of "absolutely perfect" and "perfect but not absolutely".

I believe they also have concept of "absolutely exactly" and "exactly but not absolutely".

Hmm... I would not be surprised to find out if they have a different definition of "absolutely" too.

Oh... man... what I am talking about? Am I mentally blocked?

I think I need to check if my mind is "absolutely mentally blocked" or "mentally blocked but not absolutely"

Seems the best should be going back to listen to my "absolutely better" Hi-Res music while I am thinking this topic