[RRod]

   
Fourier transform is how digital signal processing works:
 
http://en.wikipedia.org/wiki/Digital_signal_processing
 
Fourier analysis is at the heart of most analogue (waves you get from your mic or out of your headphones) and digital (0 and 1s) convertions. 
 
Your mic records the signal in the time domain, then you take samples of it to convert it digitally, the sampling rate is chosen and limited by 

Nyquist–Shannon sampling theorem​

(which is closely linked to fourier transforms)​

 
"My point is we want a higher sampling rate therefore we need to increase the frequency range" <- this means we choose sampling rate which results in a higher frequency spectrum.
 
"NOT hearing extra sounds at 20+ kHz that has been suggested" <- clearly said what you have just said "And you can't hear those frequencies, so ask yourself how you are possibly sensing something like these "better" transients."

 
I suggest you read more on the subject. Nyquist-Shannon discusses the necessary sampling rate to fully capture an *already band-limited* signal. In a PCM recording chain, this bandlimiting is accomplished by an anti-aliasing filter at the front of the ADC. Once this band-limiting is accomplished, the signal can then be digitized at the required sampling rate. Note that so far no actual DSP has happened (the anti-aliasing filter is an analog device).
 
My point on audibility is that the ONLY differences between PCM formats are the bit depth and sampling rate. If we hold the bit-depth constant, as in 16/96 vs 16/48 PCM, then all perceivable differences are due to the difference in sampling rates, and thus to the effects of the additional frequencies of the higher rate.
 
Anyway, we're in full sound science mode, so I'll see you there if you want to continue.

 

[milford30]

   
I suggest you read more on the subject. Nyquist-Shannon discusses the necessary sampling rate to fully capture an *already band-limited* signal. In a PCM recording chain, this bandlimiting is accomplished by an anti-aliasing filter at the front of the ADC. Once this band-limiting is accomplished, the signal can then be digitized at the required sampling rate. Note that so far no actual DSP has happened (the anti-aliasing filter is an analog device).
 
My point on audibility is that the ONLY differences between PCM formats are the bit depth and sampling rate. If we hold the bit-depth constant, as in 16/96 vs 16/48 PCM, then all perceivable differences are due to the difference in sampling rates, and thus to the effects of the additional frequencies of the higher rate.
 
Anyway, we're in full sound science mode, so I'll see you there if you want to continue.

you apply anti-aliasing filter so that the Nyquist-Shannon theorem works and the sampling rate was determined (44.1kHz) because an engineer decided humans can't hear anything over 20kHz.
 
i.e human hearing range leads to sampling rate dictated by Nyquist-Shannon theorem and filter is used to 'cut the ends off' so the ADC works! We if there was no Nyquist-Shannon theorem limit, the filters won't be needed.
 
If you don't understand the logic that band limiting is just a consequence of the N-S theorem and in turn fourier transform to convert D>A or A>D, then you need to do some thinking. Band limiting is only required because of fourier transform, not limiting to cut out high freq for no reason.
 
I suggest you do less reading and more thinking

 

[RRod]

  you apply anti-aliasing filter so that the Nyquist-Shannon theorem works and the sampling rate was determined (44.1kHz) because an engineer decided humans can't hear anything over 20kHz.
 
i.e human hearing range leads to sampling rate dictated by Nyquist-Shannon theorem and filter is used to 'cut the ends off' so the ADC works! We if there was no Nyquist-Shannon theorem limit, the filters won't be needed.
 
If you don't understand the logic that band limiting is just a consequence of the N-S theorem and in turn fourier transform to convert D>A or A>D, then you need to do some thinking. Band limiting is only required because of fourier transform, not limiting to cut out high freq for no reason.
 
I suggest you do less reading and more thinking

 
You're not contradicting anything I said. Instead you seem mad that 20kHz was picked as the cutoff, despite the fact that audiology tests show that this is perfectly reasonable. But as I said, this is all sound science now so feel free to move on over there, where everything I've said will be verified for you.

 

[Music Alchemist]

I honestly think this thread should have been posted in the Sound Science (or at least Computer Audio) section, since the topic has little to do with portable source gear.
 

  We'll agree to disagree.

 

 
If you can link me to documented cases of people verifying that they can hear a difference between Red Book PCM and high-res PCM, I would be very interested in seeing it. The first prerequisite is that the Red Book files must have been converted from the high-res files; otherwise, it's likely that two different masters of the recording were being played, in which case differences would be easily audible.

 

[goodvibes]

 

[musikaladin]

... here a DSD64 LP rip... and I also love it on my portable AK100II... so funny, reminds me of the times when we recorded LPs on tapes and listened via walkman; just now with portable high-res the sound is so much better...

 

[RRod]

  ... here a DSD64 LP rip... and I also love it on my portable AK100II... so funny, reminds me of the times when we recorded LPs on tapes and listened via walkman; just now with portable high-res the sound is so much better...

 
Referring to that thread, I'd happily make a 24/384 version of the DSD64 file, and then various downsamplings that would be inaudibly different, but that would mean you'd have to be willing to accept that 24/384 can be as "stress-free" as DSD64.

 

[goodvibes]

I might supply a few files at some point so folks can try for themselves. Problem is that I even have some issue with file hosting sites once you get 24/192 after a particular file transfer over a well known bit perfect site, I've stayed away from testing others so don't know if this is universal. Not going to get into how or why here but there's a lot of places in a chain where things can go less than right. Once you're convinced those aren't there, you tend to stop looking for them. I think most hear their equipment or file quality limitations and why I hate this particular argument. Absolute format determinations need to be done on the absolute best equipment with the absolute best files under the absolute best conditions. The rest becomes personal and valid for those doing it but not universal truths. Why my earlier post as to the circle argument as it involves too many variables to get anywhere significant or convincing to an opposing point of view on a forum.

 

[Music Alchemist]

  I might supply a few files at some point so folks can try for themselves. Problem is that I even have some issue with file hosting sites once you get 24/192 after a particular file transfer over a well known bit perfect site, I've stayed away from testing others so don't know if this is universal. Not going to get into how or why here but there's a lot of places in a chain where things can go less than right. Once you're convinced those aren't their, you tend to stop looking for them. I think most hear their equipment or file quality limitations and why I hate this particular argument. Absolute format determinations need to be done on the absolute best equipment with the absolute best files under the absolute best conditions. The rest becomes personal and valid for those doing it but not universal truths. Why my earlier post as to the circle argument as it involves too many variables to get anywhere significant or convincing to an opposing point of view on a forum.

 
Variables are irrelevant when you absolutely cannot hear anything above 20 kHz (or perhaps slightly higher, for some people) and no recording in existence has more dynamic range than 16-bit resolution can handle. At any rate, if you perceive a difference and want to document whether it is real, you can do so at my thread dedicated to this topic.

 

[goodvibes]

LOL

 

[Music Alchemist]

  LOL

 
I see nothing funny about people who insist they can hear a difference, yet refuse to back up their claims.
 
No one with even a basic understanding of audio has reason to take high-res seriously. Not a single argument made has stood up to scrutiny.

 

[goodvibes]