Ok, I understand the bit depth, signal to noise ratio and headroom when recording or math calculation .Still in common car stereos, home-theaters,my Pro Behringer and Sony near field monitors, JVC and Pro Yamaha headphones...well, is like I cant heard a difference between a " first generation" MP3 file encoded with Soundforge at real stereo 44k/192 kbs and the outed 24 bits original wave file, no matter the original higher sample rate!
I'm crazy????
One thing I know is...suppose my Neuman 4000 dls condenser mic register and record according to specs from 20 to 20000 hz, imagine my guitar is in a long line of processors (analog, digital) and the last one is a 24bits/ 48k delay..or let say I use real tape saturation in some recording tracks and tape give no more than 40khz according to what I know..resuming: most of what we input record to our daws (direct line, microphones, processors ect ect etc) never even reach 48khz in most cases,so... what the hell are talking those professional when they say that they LISTEN to something extraordinary different when they mix/export to 88.2 kHz, or 96 kHz or 192 kHz???? when they already KNOW the original substantial material (voice, guitar, whatever) never reached that frequency!!!!
First, your not a Whale or Elephant to hear that frequencies, 2nd, your microphone or instruments never gave you those higher frequencies in first place!!!
Some advice please, I think Im crazy!

That question is gonna turn into another 100 post thread easily :) There are a lot of interesting articles out and about, but this one might interest you, if you haven't seen it yet. http://mixonline.com/recording/mixing/audio_emperors_new_sampling/

I'm just waiting for a double-blind test where someone can reliably tell the difference among different sample rates. I'm not saying it's not possible; I just haven't seen any.
 
In some tests done at AES several years ago, the only technology where I could reliably tell an obvious difference was DSD compared to 44.1kHz PCM.

Neil Young's Porno video makes me chuckle:
http://vimeo.com/88705147

I agree that many very high frequencies seem useless or even detrimental to me (cpu load and storage space wise). However, you seem to mistakenly make the assumption that a sampling frequency of 48KHz equals the "audible" frequency of 48KHz. This is not the case, a sampling frequency of 48KHz means the converter measures 48000 times per second what the value is. If you were recording a true sine wave of 10KHz, at 48KHz sampling rate it would only be measured 4.8 times for each cycle of the wave, meaning it would become digitally represented by 4.8 "points", not a very accurate representation of a sine wave at all. This is the reason the sampling frequency is much higher than the audible frequency.

The endless discussion begins anew.....
 
Bob Bone

Sanderxpander
I agree that many very high frequencies seem useless or even detrimental to me (cpu load and storage space wise). However, you seem to mistakenly make the assumption that a sampling frequency of 48KHz equals the "audible" frequency of 48KHz. This is not the case, a sampling frequency of 48KHz means the converter measures 48000 times per second what the value is. If you were recording a true sine wave of 10KHz, at 48KHz sampling rate it would only be measured 4.8 times for each cycle of the wave, meaning it would become digitally represented by 4.8 "points", not a very accurate representation of a sine wave at all. This is the reason the sampling frequency is much higher than the audible frequency.

This is the myth in a nutshell. Mathematically you only need one sample to reproduce the wave. Two samples becomes redundant. People equate sample rate with resolution. More slices of the wave form. In fact its not needed to create and reproduce the sound. Its hard to get ones head around this concept but its how sampling works.

John
This is the myth in a nutshell. Mathematically you only need one sample to reproduce the wave. Two samples becomes redundant. People equate sample rate with resolution. More slices of the wave form. In fact its not needed to create and reproduce the sound. Its hard to get ones head around this concept but its how sampling works.

Respectfully, it is false to say that "Mathematically you only need one sample to reproduce the wave."  What you can represent with just one sample is the periodicity of a waveform, but not its shape (and this only works up to frequencies equal to half the sampling rate; i.e. the Nyquist frequency).  Beyond the Nyquist frequency you get aliasing.
 
The problem with using too few samples is that you are not able to accurately reconstruct the waveform shape.  The shape of a waveform is related to the spectrum of the sound, and this varies profoundly as you vary the wave shape.  This is the principle behind Fourier analysis/synthesis.  The relevant idea is that for a given waveform periodicity, you can have numerous different waveform shapes /spectra; and each of these is going to sound different timbrally.  At the Nyquist frequency you would not, for example, be able to discern a sine wave from a square wave.  Cheers...

Really! Alright, here we go.
 
Nope, im not going to do it..... Its a waste of time. total waste of time.
 
Bye
 
CJ

Anderton
I'm just waiting for a double-blind test where someone can reliably tell the difference among different sample rates. I'm not saying it's not possible; I just haven't seen any.
 
In some tests done at AES several years ago, the only technology where I could reliably tell an obvious difference was DSD compared to 44.1kHz PCM.

This