Comment by untangle

This article is one of the most lucid and accurate that I have read on this topic.

However, one thing that's missing here (and in nearly all other similar pieces) is a full discussion of the prerequisites of the sampling theorem. For example, the signal must be bandwidth-limited (and no finite-time signal can be).

But this is a minor concern, as there are many elements in the analog domain of the recording and playback chains that serve as low-pass filters - starting with the mics. So bandwidth-limiting is effectively achieved.

For a similar reason, the discussion of the "harmful" effect of high-frequencies to playback electronics and loudspeakers to be a bit overdone IMO. Peruse the excellent lab results of modern audio gear on Stereophile's web site. You'll find that bandwidths exceeding 30kHz are rare.

One last thing. When doing subjective "testing," keep in mind that what some folks are hearing may be limitations of their gear. For example, most DACs derive their clocks for higher sampling rates (88/96/176/192) by clock-multiplier circuits. IOW, 44kHz and 48kHz are the only ones clocked directly by a crystal. These multiplier circuits are often noisy, contributing to jitter. The audible effect of this jitter is hard to predict.

Bob

PS As an avid audiophile, I find the clash of subjectivists and objectivists on this normally-buttoned-down forum to be a bit of a trip.