======================DIGITAL_FILTERS======================================

FIR                   Finite Impulse Response
                      input samples  put in shift register with limited length.
                      output is function of what is stored in shift register
                      stuff below has some errors/typos:
                      ex. y(k)=a*y(k)+b*y(k-1)+c*y(k-2)
                      The above difference equation is recursive, w
                      hich makes it IIR.
                      (Actually it's not even a filter, since there is no input)
                      I guess it was a typo :)
                      It should read:
                                          y(k)=a*x(k)+b*x(k-1)+c*x(k-2)
                      Transfer function:  H(z) = Y(z)/X(z) = a + b*z^-1 + c*z^-2
                      H(z) has only zeros, no poles.
                      coefficients a, b, c may be determined by studying 
                      response of a dirac pulse (spike).
                      a=0, b=0.5, c=0.25 would probably give a nice low pass
                      filter, to make an unscientific guess. To be 
                      scientific,
                      use the z-transform.
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IIR                   Infinite Impulse Response
                      recursive algorithm, where filter value is determined by
                      input value and current value.
                      I think this is more accurate:
                      A recursive algorithm where the filter output value is 
                      determined by current and past input values, 
                      as well as past  output values.
                      >
                      >  ex. first order low pass filter y(k)=(d-1)y(k)-d*y(k-1)
                      >      where the rise time is determined by d and the 
                      sample rate
                      >
                      Again, this equation does not have any input. (Where do 
                      you put the
                      'stuff' you want to filter?)
                      Example of IIR:    y(k) = b0*x(k) + b1*x(k-1) - a1*y(k-1) 
                      - a2*y(k-2)
                      Transfer function of same filter:
                                         H(z) = Y(z)/X(z) = (b0 + b1*z^-1) / (1 
                      + a1*z^-1 + a2*z^-2)
                                         H(z) has both zeros and poles.
                      >
                      >This is the general idea, off the top of my head.
                      >Study a book on digital filters to get the details.
                      >

                      Original Subject: Re: digital filtering
                      Two common filter structures are used:
                      FIR- Finite Impulse Response
                        The input samples are put in a shift register with 
                      limited length.
                        The output is a function of what is stored in the shift 
                      register
                        for the time being.
                        ex. y(k)=a*y(k)+b*y(k-1)+c*y(k-2)
                            coefficients a, b, c may be determined by studying 
                      the
                            response of a dirac pulse (spike).
                            a=0, b=0.5, c=0.25 would probably give a nice low 
                      pass
                            filter, to make an unscientific guess. To be 
                      scientific,
                            use the z-transform.
                      IIR- Infinite Impulse Response
                        A recursive algorithm, where the filter value is 
                      determined by
                        the input value and the current value.
                        ex. first order low pass filter y(k)=(d-1)y(k)-d*y(k-1)
                            where the rise time is determined by d and the 
                      sample rate