Page 41 - Analog and Digital Filter Design
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38       Analog and Digital Filter Design





                 Digital Filter Types

                       Digital filters are becoming more widespread  in use and are replacing analog
                       filters in many systems. Digital filters process signals in the time domain. Analog
                       signals have first to be sampled and digitized at discrete (clock) intervals using
                       an analog-to-digital converter.

                       Because the analog signal is sampled, care has to be taken to prevent errors such
                       as aliasing. Aliasing, which was described earlier, occurs when the analog signal
                       has spectral energy at frequencies above half the sampling frequency. The analog
                       and sampling signals mix in such a way that it is impossible to recover the origi-
                       nal signal when it is converted back to analog. To prevent aliasing, the highest
                       frequency of the input signal must be filtered. In telecommunications, the upper
                       voice frequency is limited to 3.4kHz with a very steep skirt (Le., a sharp roll-
                       off), so that there is  no discernable energy at 4kHz or higher. The voice fre-
                       quency is then sampled at 8 kHz.

                       What would happen if  an analog signal at, say, 5 kHz is passed then sampled
                       at 8 kHz? Mixing between the 5 kHz signal and  the 8 kHz signal would cause
                       signals to be generated  at the sum and difference frequencies. Thus signals at
                       3kHz and  13kHz would  be  produced.  When  converted  back  to  analog,  the
                       13 kHz signal would be outside the passband of  the output filter, but the 3 kHz
                       signal would be inside the passband  and thus appear at the output as an alias.

                       Once  digitized,  the  signals  are  digitally  filtered  by  either  a  dedicated  IC  or
                       a digital signal processor (DSP) using a filtering software. Within every digital
                       filter there are delay elements, multiplying functions and adders, which process
                       the digitized signal. There are two types of digital filter: Finite Impulse Response
                       (FIR) filters,  which  are  also  described  as  nonrecursive  filters;  and  Infinite
                       Impulse Response (IIR) filters, which are recursive because part of  the output
                       signal is fed back to the input.

                       The recursive approach in digital filtering processes uses negative feedback  in
                       order  to  obtain  a  sharp roll-off  using the  minimum  of  delay, summing, and
                       multiplying elements. The feedback  comprises a small fraction  of  the  output
                       signal. Because of  the delays, any sudden change in the input signal affects the
                       output for some time (possibly forever, if  there is any instability in the design).
                       Recursive filters are  said  to  have  an Infinite  Impulse  Response  (IIR). Some
                       designs are sensitive to the filter coefficients used  as multiplying  factors,  and
                       truncating the coefficient (limiting the number of decimal places) can result in
                       positive feedback and hence oscillation.

                       Nonrecursive, or moving average, filters take several successive samples then sum
                       them  (perhaps with  a  scaling  factor  at  each  tapping  point)  to  produce  the
                       average of several samples. As time goes on the samples ripple through the filter,
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