[net.audio] Fourier Transforms in Digital Audio

stan (12/20/82)

In these newsgroups, people justifying the reproduction ability of
digital audio systems say things like, "if we sample at rate X and
properly Fourier Transform the data ...."

Please explain where the Fourier Transform is used in a digital audio
(phonographic) system.

As I understand it, digital audio is the manipulation of discrete samples
digitized as soon after the microphone as possible, then converted back
to analog levels as late as possible before the speaker.  Although
the input and output must be low-pass filtered to prevent high frequencies
from folding into lower ones, this is simply a requirement dictated by
the discrete sampling.  I was not aware that Fourier transform was used
at all.

The use of the terminology is therefore misleading.  Please explain it
to me.