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

yes, I seem to concur that 24/192 makes no sense. I never was a believer of the hi-rez scam that pervades the industry.
There is also a nice article written by Dan Lavry on Lavry Engineering's website on why sampling upto a max of 24/96 makes sense & anything beyond that is bogus. Dan Lavry wrote this article back in 2004! This article is called "Sampling Theory" when you go to this link:
http://www.lavryengineering.com/index_html.html

Insert a quantity of "$" in the space between "1" and "0" and adjust until it sounds good to you.

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.