🚀 go-pugleaf

	I know nothing about DSP other than what I've figured out
on my own, based on simple common sense, but it seems to me that
there's a better way to lower pitch than just doubling samples.
Seems to me that you should interpolate them; if you have three
samples, 1000, 500, 250, and you want to make five samples out of
them, it seems to me that instead of doing 1000, 1000, 500, 500, 250
you should do 1000, 750, 500, 375, 250.  That's a reasonably simple
add-and-shift-right-one-bit algorithm that shouldn't take too long
and would preserve more of the fidelity than sample doubling.
	Also, if you're going to remove samples, I think you
shouldn't use a simple kill-every-nth-sample procedure.  It seems
to me that certain samples (local maxima and minima) are more
important than others.  Use a five- or seven-sample queue, where
you consider the middle one for removal.  If it's a local minimum
or maximum, zap one of the ones next to it instead.  That's a bit
more complicated, but I think a 10MHz 8086 could probably keep up
with a 30Khz 16-bit sample stream.
	Of course, this just addresses the problem of compression
or expansion of the waveform, not constant-pitch/variable-speed or
vice versa.  If we could do that in real time, to certain components
of a sound and not others (for example, vary the pitch of spoken
vowels but not consonants, or the pitch of a violin string but not
the pitch of the bow "scrape"), we could probably ditch acoustic
instruments altogether and replace them with samples.
	Again, while I'm very interested in this field, I am by no
stretch of the imagination anything remotely resembling an
authority, so keep your flames gentle :-).