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Implementing IIR/FIR Filters

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pass Butterworth filter (three second-order sections)<br />

in both the analog domain and digital domain. Note<br />

that the gain of the first section (k = 1) is greater than<br />

unity near the cutoff frequency but that the overall<br />

composite response never exceeds unity. This fact<br />

allows for easy implementation of the Butterworth filter<br />

in cascaded direct form (i.e., scaling of sections<br />

is not needed as long as the sections are implemented<br />

in the order of decreasing k). Overflow at the<br />

output of any section is then guaranteed not to occur<br />

(the gain of the filter never exceeds unity). Note that<br />

the digital response (see Figure 5-28) is identical to<br />

the analog response but warped from the right along<br />

the frequency axis. Imagine the zero at plus infinity<br />

in the analog response mapping into the zero at f s /2<br />

in the digital case. Also note that, because of this<br />

mapping, the digital response falls off faster than the<br />

-12 dB/octave of the analog filter when the cutoff is<br />

near f s /2.<br />

The previous analysis is nearly identical to the case<br />

of the highpass filter except the coefficients (see<br />

Figure 3-24) have slightly different values. Since<br />

the bandstop case is just the sum of a lowpass and<br />

highpass case, it can be analyzed by these techniques.<br />

The bandpass case, however, is more<br />

difficult and requires considerably more work (see<br />

Reference 14) because the center frequency of<br />

each section is now different and the formula for<br />

calculating these frequencies is not as simple as<br />

the formulas for the previous filter types. In addition,<br />

to complicate matters further, scaling between<br />

sections becomes more of a problem since the off-<br />

5-8 MOTOROLA

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