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.
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.
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 toono 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.