Sonar & other workstations, the sampling frequency is a myth?

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!

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.

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.

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.

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

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

Yep - this too

abb

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...

Well that is just pure nonsense. If it were true than sampling wouldn't work at the CD level CDs are 16 bit and 44.1 kH sampling. We should hear all that aliasing why don't we?  Could it be because its not there? 

Two samples are needed. Remember the Nyquist freq is 1/2 the sampling rate.
These two samples make a square wave at the Nyquist and after filtering make a sine wave. All a human can hear at 20Khz is a sine wave (even tho the waveform may have higher harmonics) . The higher harmonics that make the square wave are NOT audible. And yes mathematically all that is needed is two samples at 44.1Khz. The key here is that a complex waveform with higher harmonics above the Nyquist is NOT accurately reproduced but the waveform's harmonics up to the Nyquist ARE reproduced. The freqs below the Nyquist is ALL that we care about.
Typically the sample rate is a little more than 2x the highest freq to be reproduced since the ADC and DAC filters are not perfect.
Michael

Sander,
You are exactly right. the highest freq your rig can produce is slightly less than 1/2 the sampling rate. Even with higher sampling rates most codecs will digitally filter the input/output pass band to audible freqs. Each time the sample rate is doubled, it pushes the sampling artifacts down by 6db which reduces the filter complexity on the DAC output. So if you are working at 96Khz it does not mean your rig can output 40+Khz. Only a bat could hear it even if it did.
BTW 44.1Khz  was chosen by Philips for the CD because of some video related freq that was easily divided and still above 20Khz x 2. 

bitflipper
I was going to give a long, detailed, annotated and cross-referenced response to the OP, but after some consideration I decided to go have a sandwich instead.

Wise move Bit, I'm considering coffee too :-)
 

Here are the facts:
 
1. If you know in advance that a signal is a sine wave, you only need 3 samples to know everything about that sine wave - the frequency, the amplitude and the phase at which each sample was taken. It doesn't matter if it's a low frequency with thousands of samples per cycle or a high frequency with only 3 samples per cycle and it doesn't matter where the samples are taken (i.e. you don't need the samples to be at a zero crossing or at a 90 degree peak or whatever). 
 
2. A sampled waveform is not reconstructed by drawing lines between adjacent samples despite what many think.
 
If a sampled signal contains no frequencies greater than half the sample rate, everything between the samples is in fact stored in the samples. That's the essence of the sampling theorem. 
 
Note that the stuff between the samples is not just stored in the adjacent samples, but in a long stream of samples. If you can't fathom how that could possibly work, lets just say it involves non-trivial math.

Which is precisely why this is an endless debate....
 
This debate has existed since the mammals rose to power, and will be continued by the cockroaches that survive our collective passing of the planet to them.
 
People think what they wish, and in all of the thousands and thousands of posts on this subject I have seen, not ONCE did I see a single person ever change their mind, one way or the other.
 
There are multiple positions on the mathematics, on the range of human hearing, and actually any other aspect of this that anyone has floated.
 
It is as it is, and will always be what it has been and never was.
 
All of it is inaudible to me anyway, having grown up with a close and scratch phonograph.
 
Bob Bone
 

microapp
drew,
There ARE lines being drawn between the samples. The DAC is interpolating with typically an oversampled sinx/x function and/or a lowpass filter is smoothing the output. The lines between the samples are just not straight lines.
 
Michael

Yes. I was attempting to give a less technical explanation and talking about the common perception that the between the samples stuff is reconstructed by linear interpolation.
 
And we should also note the values between the samples can be greater or less than the adjacent samples.

And the short, non-technical explanation the impact of higher/lower bit depth and sample rate for playback (not talking about processing inside a DAW here, which is more complicated):
 
Assuming dither is properly applied, more bits just equals less noise.
 
Higher sample rates equals higher frequencies can be present and potentially any artifacts near the top of the frequency range are moved to a higher frequency, and also potentially less audible noise.
 
 
The key question in all of this is at what point of these things are any improvements in noise/artifacts/higher_frequencies no longer audible.