This article really misses the facts of the Nyquest-Shannon theory. In order to decimate a signal to 44.1 or 48khz, and preserve high-frequency content, high frequencies need to be phase-shifted. This phase-shift is similar to how lossy codecs work. For what it's worth: I'm a big fan of music in surround, and most of it comes in high sampling rates. When I investigated ripping my DVD-As and Blurays, I found that they…
> So the math is ideal, but what of real world complications? The most notorious is the band-limiting requirement. Signals with content over the Nyquist frequency must be lowpassed before sampling to avoid aliasing distortion; this analog lowpass is the infamous antialiasing filter. Antialiasing can't be ideal in practice, but modern techniques bring it very close. ...and with that we come to oversampling.
if you accept that the limit of hearing is around 20 kHz, then you must also accept that frequencies above that can freely be removed without loss of fidelity to the human ear.
the article notes that higher frequencies can be heard, but only in the form of ultrasonic intermodulation distortion. (i.e. not in fact the higher frequencies at all)