You can
calculate your tranformation for the input signal S(x_i) once
And the same transformation for S(x_i)+1 again.
Won't that just give you the gradient at point x_i, ie. d/dt(S)?
yes.
We are talking about frequency domain aliasing here,
oh yeah i see. sorry.
which is when you
generate partials that would be above the nyquist frequencyi, so they get
reflected down into low frequencies. It is not directly related to the
differential of the signal, though a high differential is often indicative
of an aliasing problem.
Typically you prevent audio aliasing by generating the waveform in a way
so that it contains no partials above nyquist, or by generating it at a
sufficiently high sample rate that there are none, then decimating down.
but you can do the pretty same trick for the frequencies:
take two different numbers with no common divisors
(or just a pair of prime numbers) as sampling rates
and see how the output signal changes.
The difference give you some linear combination of aliased frequencies.
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