Focus on 24/192 Misguided?.....


As I've upgraded by digital front end over the last few years, like most people I've been focused on 24/192 and related 'hi rez' digital playback and music to get the most from my system. However, I read this pretty thought provoking article on why this may be a very bad idea:
http://people.xiph.org/~xiphmont/demo/neil-young.html

Maybe it's best to just focus on as good a redbook solution as you can, although there seem to be some merits to SACD, if for nothing else the attention to recording quality.
128x128outlier
Also agree with the above, there is only so much on a recording, mastering is where it's at.

My Dac is a custom built NOS AD1865K 16/44 with short signal paths, silver internal wiring and V-CAP OIMP output caps fed by a modded CEC transport, if there's anything missing it's the recording.

I have some (not all) quality redbooks that equal and beat some of my friend's LP counterparts from his Walker T.T. and we compare all the time.

This is a good thread so far.
Poorly recorded and mastered music doesn't sound good at any sample/bit rate. Is that really surprising?

The referenced article is nonsense. Essentially he's saying the math is perfect and therefore nothing else is needed for better sound quality. Sony/Philips said the same thing back in the mid-1980s. Out of bandwidth induced noise is a problem in both digital and analog. It is something engineers are well aware and there are a multitude of solutions with proven track records. If an audio amplifier manufacturer said his new amp filtered out all audio above 20kHz would you consider it a serious high end oriented design? If you really take the article seriously we should all be using 32kHz/12bit digital because the math works perfectly at that level too.
I'm sorry Onhwy61 the referenced article is not nonsense. We are dealing with a pure digital signal until the output of the D/A. So, there are plenty of DSP techniques available to make this work without oversampling the heck out of the digital signal. We need to oversample just enough to ease the specifications of the analog reconstruction skirt so that it's not brickwall. That's where 96KHz sampling comes in.
I also bet that most people's systems (including yours & mine) do not have 96dB of dynamic range after all the sweat that we have put in to isolate & reduce noise.

If you really take the article seriously we should all be using 32kHz/12bit digital because the math works perfectly at that level too
no it does not. The Fs/2 freq would be 16KHz which would be less than 20KHz.
And, 12-b would be insufficient because one would add too much noise when going thru the mastering process & you'd effectively get 9-10 bits of music.
And what's wrong with mathematically perfect response out to 16kHz? Most people, and certainly most middle age and older audiophiles' hearing doesn't go beyond 16kHz. FM doesn't go beyond 15kHz and at its best it's pretty damn good sounding. 12bit audio has a theoretical dynamic range of 72dB. That exceeds all but the most dedicated of audiophile rooms and systems. And if you increase the sampling rate above 32kHz won't you be introducing ultrasonic IM distortion?

44.1kHz/16bit is not needed. Done!
I agree with Onhwy61 - article is a nonsense. First, motion that 16/44 is perfect if meets Nyquist criteria is first nonsense. Nyquist criteria applies only to continuous waves. Short high frequency bursts like cymbals will suffer the most of distortion. Second nonsense is that digital filter can suppress 96dB within 4kHz without any problem. There is no filter in the world, digital or analog, that can do that (no matter how many poles) with even group delays (or linear phase if you prefer). Uneven group delays will cause poor summing of harmonics (delayed differently) and change in sound. Reducing suppression won't help since low level signals above 24kHz will "fold" into audible band starting at 0Hz. Next nonsense is that ultrasonic frequency is harmful to the ear and modulate tweeter. Not only that 192kHz is WAY easier to completely filter out than 44kHz but also modulation can only happen on nonlinear element and for this to happen membrane has to move - not likely at 192kHz (even if your amp and CDP have such bandwidth). Then he claims that higher resolution does not increase dynamic range because of ambient noise floor forgetting that it is still improving resolution for louder signals. He claims that oversampling can increase resolution and sampling rate - true, but it is done with interpolated samples while 24/192 contains real samples. I agree that we might have hard time to hear better above certain resolution/rate, for instance 20/96 but claim that 24/192 is harmful is complete nonsense.