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The design of a digital filter is usually specified in terms of the characteristics of the signals to be passed through the filter. In many cases, the signals are described in terms of their frequency content. For example, even though it cannot be predicted just what a person may say, it can be predicted that the speech will have frequency content between 300 and 4000 Hz. Therefore, a filter can be designed to pass speech without knowing what the speech is. This is true of many signals and of many types of noise or interference. For these reasons among others, specifications for filters are generally given in terms of the frequency response of the filter.
The basic IIR filter design process is similar to that for the FIR problem:
This section develops several practical methods for IIR filter design. A very important set of methods is based on converting Butterworth, Chebyshev I and II, and elliptic-function analog filter designs to digital filter designs by both the impulse- invariant method and the bilinear transformation. The characteristics of these four approximations are based on combinations of a Taylor's series and a Chebyshev approximation in the pass and stopbands. Many results from this chapter can be used for analog filter design as well as for digital design.
Extensions of the frequency-sampling and least-squared-error design for the FIR filter are developed for the IIR filter. Several direct iterative numerical methods for optimal approximation are described in this chapter. Prony's method and direct numerical methods are presented for designing IIR filters according to time-domain specifications.
The discussion of the four classical lowpass filter design methods is arranged so that each method has a section on properties and a section on design procedures. There are also design programs in the appendix. An experienced person can simply use the design programs. A less experienced designer should read the design procedure material, and a person who wants to understand the theory in order to modify the programs, develop new programs, or better understand the given ones, should study the properties section and consult the references.
The mathematical problem inherent in the frequency-domain
filter design problem is the approximation of a desired complex
frequency-response function
For the digital filter design problem, the mathematics are
complicated by the approximation being defined on the unit circle.
In terms of
The details of the bilinear and alternative transformations are
covered elsewhere. For the purposes of this section, it is
sufficient to observe[4], [3] that the frequency response of a
filter in terms of the new variable is found by evaluating
There are two reasons that the approximation process is often formulated in terms of the square of the magnitude of the transfer function, rather than the real and/or imaginary parts of the complex transfer function or the magnitude of the transfer function. The first reason is that the squared-magnitude frequency- response function is an analytic, real-valued function of a real variable, and this considerably simplifies the problem of finding a “best" solution. The second reason is that effects of the signal or interference are often stated in terms of the energy or power that is proportional to the square of the magnitude of the signal or noise.
In order to move back and forth between the transfer function
which is related to the squared magnitude by
If
then
In this context, the approximation is arrived at in terms of
which gives the magnitude-squared frequency response when evaluated
around the unit circle, i.e.,
The next section develops four useful approximations using the continuous-time Laplace transform formulation in s. These will be transformed into digital transfer functions by techniques covered in another module. They can also be used directly for analog filter design.
Four basic filter approximations are considered to be standard. They are often developed and presented in terms of a normalized lowpass filter that can be modified to give other versions such as highpass or bandpass filters. These four forms use Taylor's series approximations and Chebyshev approximations in various combinations[5], [6], [1], [7]. It is interesting to note that none of these are defined in terms of a mean-squared error measure. Although it would be an interesting error criterion, the reason is that there is no closed-form solution to the LS-error approximation problem which is nonlinear for the IIR filter.
This section develops the four classical approximations in terms of the Laplace transform variable s. They can be used as prototype filters to be converted into digital filters or used directly for analog filter design.
The desired lowpass filter frequency response is similar to
the case for the FIR filter. Here it is expressed in terms of the
magnitude squared of the transfer function, which is a function of
The Butterworth filter uses a Taylor's series approximation to
the ideal at both
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