I have stated that sampling higher than the nyquist rate on audio signals will produce a better reconstructed signal.
Well, there is a degree of truth to this. But it's irrelevant. Bear with me on this, rabeach.
First, "sampling higher than the nyquist rate" is meaningless. There is no such thing as a "nyquist rate". There is, however, a "Nyquist frequency", but it's not any specific frequency. The term is used to describe the minimum sample frequency for a given bandwidth. Raising the sample frequency increases the possible bandwidth that can be captured, so the Nyquist frequency is determined by the highest audio frequency you want to capture.
Second, the sampling theorem specifically applies to a
band-limited system. That means you decide up front what bandwidth you need and go from there. There is no such thing as sampling an unlimited bandwidth. You always start with the premise that there is a specific bandwidth you're interested in.
Why is this "band-limited system" idea so important? Because it determines what a properly reconstructed signal should look like. For example, if you sample a 20KHz square wave at 44.1KHz the reconstructed waveform will NOT be square, it will be a sine wave. (This, I think, is where you're going off track, because if you were to increase the sample rate you would indeed get something more closely resembling a square wave - although still far from square).
Why am I willing to accept a 20KHz
sine wave as a reasonable facsimile of the original
square wave?
Because I cannot hear any difference between the two! Not just me, nobody can. Not even dogs and bats.
That's because it's the odd harmonics that make a square wave square, and what make it sound different from other waveshapes. For a 20KHz fundamental, the frequency of the first odd harmonic is 60KHz, the next is 100KHz, and the next is 1400KHz - all of them far beyond audibility.
Losing them does not change the perceived sound at all.
So if you start with the presumption that only audible frequencies need to be recorded, then there is no need to capture anything beyond 20KHz. That is the definition of a band-limited system, the prerequisite for the application of sampling theory, and setting that maximum frequency is what allows us to capture data digitally in the first place.
This idea that frequencies can be just cast away may seem counter-intuitive, but you can easily demonstrate it for yourself. Don't use 20KHz, since most people can't hear that anyway; use 10KHz instead. Using a signal generator, record a 10Khz square wave at the highest sampling rate your interface supports. At 192KHz you can capture the first three harmonics, which won't yield a truly square-looking wave, but it will obviously not be a sine wave, either. Then record a 10Khz sine wave. Now do a blind A/B test to see if you can tell the difference. If your interface is working properly, you won't be able to. The reason is that those extra frequencies you captured simply don't matter because you cannot hear them.
So at the end of the day, you are right: the higher sampling rate will indeed capture a broader bandwidth and thus preserve the original waveform better. However, it doesn't matter in the slightest.