This I believe is not correct. Implicit in the assumption of samples
is that they represent points along a wave form which is the Fourier
sum of sine waves. Thus by doing the D-A conversion (but not really),
one can predict with high precision the value a sample should have
that's arbitrarily somewhere inbetween any two samples (which is what
resampling is intended to do, at least that's how I understand it.)
Thus, I am of the understanding, which may be wholly wrong, that the
published algorithms for digital resampling are very well developed
and universally implemented by the better sound processing tools. And
that one can even predict the distortion caused by these algorithms, and
that it is typically very small. But then, I've been surprised by what
these tools can and cannot do. Anyone?