The author confuses sampling theory with reality. Sampling theory says you can reconstruct a signal by sampling it at least twice per cycle. So 44.1KHz is an adequate sampling rate for an 22.05Khz signal. Unfortunately to make Digital-to-Analog conversion work according to theory, you must first construct an ideal analog filter which filters out everything above 22.05Khz while leaving everything less than 22.05Khz un…
Suppose you have a 96khz DAC coming from your computer. Surely you see that a computer can solve for the nyquist reconstruction of some lower sample-rate recording (e.g. CD audio) to that sample rate, and then (still in digital 96khz) remove ALL the noise between say 21khz and 96khz, at which point a final analog filter will have an easy time leaving 0-21khz unmolested while killing all higher frequencies, right? It's practically implied by what you wrote. Per what OP says, that's more or less the effect of any DAC you'll use today.