Re: Anti-aliasing ADC samples



There is a way to digitally filter a signal without suffering from
aliasing. It is called the bi-linear z transform method. You design
an analog filter, and then transform it to a digital filter by
substituting (2/T)[(1-z^-1)/(1+z^-1)] for s, where T is the sampling
interval. Actually, the (2/T) factor is not necessary, but most books
on DSP include it. The resultant filter will have better attentuation
performance than the corresponding analog filter prototype. The
transformed filter suffers from "frequency warping". You compensate for
this by pre-warping the critical frequencies before you design the
analog prototype filter. Neither the phase vs frequency curve nor the
step response of the digital filter will match the correspondimg phase
and step response characteristic of the analog prototype filter. Any
book that covers digital filtering will have a section on the bilinear
z-transform. See, for example, "Therory and Application of Digital
Signal Processing" by Rabiner and Gold.

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