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In article <1319@beguine.UUCP> Jeff.Miller@samba.acs.unc.edu (Jeff Miller) writes:
>Of course the first thing any treatment of digital audio and A/D converters
>geos into is the Nyquist frequency and the crucial need for effective input
>filtering to avoid aliasing. That I've got covered.
> 
>But I don't recall any treatment of the need and mathematics of filtering
>the output of a D/A converter. I would imagine at CD or DAT frequencies
>it wouldn't be too critical as any harmonics would be ultrasonic (or would
>
>If the requirements are as stringent as the input filtering requirements,
>I may have some trouble. Would mathematically interpolating a few output
>levels between each input sample help, and deos that technique have a name
>I should look out for?
>
>Thanks, 
 The answer to the above question is yes. Different types of sampling
systems do have different requirements for filtering.

One way to look at this is to regard the sampled signal as a Taylor
series. Commonly only the current sample is used. This has been
called a zero order sampler. It requires the sinx/x filter that we
all know and love. If you start deriving the derivatives of the signal
from various orders and summing these into the output sampe, you get 
higher order samplers. This is your idea of interpolation. it is really expanding the signal around the sampling time as a Taylor series. 
In the limit, if derivitives of all orders are supplied, you
have ideal sampling and no filter is required.

>