CFWhitman

There is merit, especially during recording, to the view that 24 bit audio allows more "headroom" to compensate for not getting the volume just right. This should no longer be necessary once music has been mastered, but you can always argue that there's no guarantee that the master is perfect.

However, the idea that the digital filters used in recording audio (finite impulse response filters) can cause phase deviation is incorrect. Analog low pass filters can do this, but FIR filters do not do this, no matter how close to a "brick wall" you manage to make them. Tests with oscilloscopes prove this. There is no relation between sample rate and what kind of analog filter you might be using. Some early CD players had "brick wall" analog filters which caused distortion due to phase deviation. This was resolved with the use of digital filters and gentler analog filters.

I noticed also in "The Well Tempered Computer" article that they claim that 44.1 kHz audio is upsampled to 48 kHz through "interpolation." However, that is a mischaracterization. Interpolation implies that the calculation you make of where the sample would be is only an estimate. Sound waves are predictable, so the calculation of where it will be between any two known samples is not an estimate; it's actual. I suppose that theoretically, you couldn't be sure of which sample a sound wave ended with, but as long as it's within the Nyquist frequency, it's not going to make any difference anyway.

Just for information's sake, research indicates that higher than CD quality sampling rates (tells us what frequency sounds occur at) have no benefit (and under some conditions may actually be counterproductive). There is a better chance that going from 16 bit to 24 bit recordings (affects relative loudness of different sounds) would create perceptible changes, although it doesn't seem that it does with a properly created recording.

One way to explain why the sampling rate does not make a difference after a certain point is to realize that your playback equipment recreates the sounds you hear from digital information about them. This is sort of like drawing a geometric shape from instructions about its size. The frequency that a sound occurs at is rebuilt from the samples taken. Once you have enough samples to describe the frequency, adding more samples does not make any difference. It's like drawing a straight line. Once you have the end points, getting other points along it will not change how you draw it. The only difference more samples make at that point is their ability to describe sound at a higher pitch. Since CD quality accurately describes sound at as high a pitch as any human can hear, no more is necessary.