Originally Posted by commodorejohn
However, my point is that the aliasing noise changes frequency with the frequency of the individual channel. (And since the filter(s) on the output of real Amiga hardware have a fixed cutoff frequency, it may fall entirely within the passband on lower notes.)
Oh certainly, but I think we can agree that the fault lay with whomever performed the original sampling process. So long as the original signal is properly filtered with a low pass filter before being sampled and stored, aliasing isn't a problem.
And the Amiga's scheme is actually pretty robust against aliases that appear in the waveform since the alias frequencies are all pitch-shifted in the same way that any other frequency in the waveform might be shifted as different notes are played.
In other words, the timbre of the sound, including aliases introduced during the original sampling of a signal, stays the same no matter what pitch is chosen for the current note. This is in contrast to sampling a signal at various rates where the the aliases appear at frequencies that are only arithmetically, but not geometrically related, creating a horrible dissonance.
And that to me is why the Amiga audio hardware shouldn't be called sample based.
Sample based synthesis draws values from a fixed waveform and when too few points are drawn from the sample, aliasing occurs.
But this never occurs when the Amiga is playing a waveform. It outputs every stored value in the waveform and so new alias frequencies never appear in the output.