At 07:42 PM 8/27/2003 -0600, Jon Noring wrote:

>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?

You are correct. Except in the simplest cases (e.g., halving the sample
rate), and not even always there, resampling entails interpolation or
extrapolation. The quality of the interpolation is selectable; higher order
means more processing and a better fit.


Mike
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