On Tue, 6 Nov 2007, Mario. wrote:

> My method - and some of the idustry too - is to make the sampling
> frequency way over the real input frequency.. Audio cards do 192k
> samples per second for audio in range up to 18kHz (human hearing), so
> the same way, the processing has to be multiple cycles per one real
> input/output cycle and all will be well.

If you CAN afford supersampling that's good, Nyquist would be happy for you:D

But one of the topics in the thread was the limited interrupt rate :P

And, of course, it all depends on the frequencies you work with: even an analog 
PLL, a continue process in principle (so more-or-less infinite sample rate), 
takes more time to lock to lower frequencies...

In other words it could be tricky to react to a 1Hz stimulus in a 0.1s 
timeframe... (in general IIRC it's not possible)


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