I looked at an old thread, Nyquist Frequency Phase Shift, but was left wondering still about oversampling.
I learned Shannon 40 years ago, then worked 30 years in (s/w) engineering, but now retired I have only begun using/configuring complete digital audio setups. I find myself baffled by exactly the sort of hands-on question that was posed in the referenced thread. Yet most of the relatively simple (IMO) questions I've thrown at the search engines have landed on generic "Nyquist says" posts, and almost all graphs have been frequency domain.
Like the author of the referenced post, if I draw a simple sine wave and "sample" it with something just over twice its frequency, then vary the phasing of the sample points (or the wave) a few times, I end up with very different squarewave patterns. Especially for an illustratively short burst, it's impossible for me to deduce how even the most perfect linear-phase brickwall filter could regenerate the exact analog input (amplitude plus phase), which would be critical in preserving harmonic structure for any instrument. Certainly Nyquist had little to draw upon except filters as a physical 'decoder'.
It is very easy to picture that, for a tiny increment over 2*Fm, something very similar to the zero-output example given in the referenced post would be generated for a fairly long burst. It's equally hard to see how any procedure would regenerate a perfect image of the input burst, intermediate oversampling etc. notwithstanding, whereas for a much lower frequency (same Fs) it looks feasible.
Since available hw/sw has improved so radically since CDs and DAT first showed up, has anyone seen a genuine graphed experiment? Something like a 20khz pure tone riding on a (reasonably short) 2khz tone burst (or any such two-tone combination to give a gauge for zero phase shift), sampled at a rate just higher (say 5% e.g.) than twice the higher frequency. I would love to see the results, in time domain, input vs. output. I'd also love for them to be exact over multiple sample/decode tests. Then I could get over the feeling that not just oversampling but huge oversampling might be a good idea after all.
Or does insistence on a readable burst of short length throw such a monkey wrench into the spectrum being sampled that Fmax would in reality be way, way higher?