[net.audio] f

jj@alice.UUCP (01/24/85)

Joel.

About your point number 1), regarding music not being sine
waves:
	ELEMENTARY communications theory shows you how to
calculate the bandwidth of a varying amplitude sine wave.
Given that, there is NO theoretical problem, small or large,
with the fact that music does not consist of sine waves.  Simply
put, once you have calculated the bandwidth of the signal (hell,
if you really want to get down to specifics, you can take the
TRANSFORM of the WHOLE SONG and use that.  It's a time-limited
signal that will have an EXACT spectrum) you know that you can
capture anything up to a given frequency in your digital representation.
That's all there is to it.

About your point 2).  It's rather riddled with incorrect
assumptions, your example is of a signal at exactly half the
sampling rate that has INFORMATION WELL ABOVE half the sampling rate,
and as such is just totally invalid.  Simply put, a signal that is
exactly at half the sampling rate cannot EVER change, or it has
too much bandwidth, since it then splashes past the halfway point.
A signal one Hz below half the sampling rate (I'm assuming the
"perfect" filter, but the argument holds for different frequencies with
imperfect filters) can be modulated by anything that has a bandwidth
of less than one Hz and go through just fine.  More than one
Hz and you loose information.  In other words, a rapidly varying tone
near Fs/2 has signal components ABOVE Fs/2, and will HAVE to be
filtered.

Wozencraft and Jacobs book is perhaps too advanced to recommend,
I suggest that you get a copy of Steiglitz's 
"An Introduction to Discrete Systems" and give it a good read.

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