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.

On 11/6/07, Anders Wallin <[EMAIL PROTECTED]> wrote:
>
> Baher: Analog&Digital signal processing is a pretty understandable book
> (depending on math background ofcourse)
>
> Matlab has a utility called 'fdatool' which lets you design and
> visualize filters very easily. There might be an open-sour equivalent
> somewhere.
>
> I would agree with the delay concern expressed earlier. If you get too
> much delay the negative feedback ends up being positive and you get ringing.
> If anyone finds a topology or filter class that is particularly suitable
> for putting inside PID loops I would be interested.
>
> AW
>
>
> > On Tue, 6 Nov 2007, Mario. wrote:
> >
> >> not want do only do 'some' filtering, ideally we want exact
> >> counterfilter for our aches. And we need someone with good math skills
> >> to show us precisely how.
> >
> > Or at least some basic DSP book :P I'm buying something for work (I have to 
> > filter out mechanical resonance from a load cell -- I know, it's a 
> > different thing), but I've seen somewhere a book on 'practical DSP' which 
> > could be useful :D
> >
> > At the moment I began to hate the Z-trasform (the discrete equivalent of 
> > the Laplace one...)
> >
> > At the end most of the DSP stuff is done evaluating polynomials... now I 
> > get WHY they always talk of multiply-and-accumulate!
>
>
>
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