🚀 go-pugleaf

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
they? And  I've always wondered: can two ultrasonic sounds or an ultrasonic
and audible sound beat to create an audible tone?) but at lower frequencies,
I would think that aproximating a sine wave with stepped squares would sound
bad. 

If someone could tell me loosely what the requirements are I would appreciate
it. I really must get a good handle on it as I am shooting for CD quality.

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, 

-Jeff 

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