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UNIT 8

Design of IIR Filters from Analog Filters

CONTENTS:-

DESIGN OF IIR FILTERS FROM ANALOG FILTERS (BUTTERWORTH AND CHEBYSHEV) - IMPULSE

INVARIANCE METHOD. MAPPING OF TRANSFER FUNCTIONS: APPROXIMATION OF DERIVATIVE

(BACKWARD DIFFERENCE AND BILINEAR TRANSFORMATION) METHOD, MATCHED Z

TRANSFORMS, VERIFICATION FOR STABILITY AND LINEARITY DURING MAPPING 7 HRS

RECOMMENDED READINGS:-

1. DIGITAL SIGNAL PROCESSING – PRINCIPLES ALGORITHMS & APPLICATIONS, PROAKIS &

MONALAKIS, PEARSON EDUCATION, 4TH EDITION, NEW DELHI, 2007.

2. DISCRETE TIME SIGNAL PROCESSING, OPPENHEIM & SCHAFFER, PHI, 2003.

3. DIGITAL SIGNAL PROCESSING, S. K. MITRA, TATA MC-GRAW HILL, 2ND EDITION, 2004.
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UNIT - 8
DESIGN OF IIR FILTERS FROM ANALOG FILTERS
(BUTTERWORTH AND CHEBYSHEV)

8.1 Introduction
A digital filter is a linear shift-invariant discrete-time system that is realized using finite
precision arithmetic. The design of digital filters involves three basic steps:

 The specification of the desired properties of the system.

 The approximation of these specifications using a causal discrete-time system.
 The realization of these specifications using _nite precision arithmetic.

These three steps are independent; here we focus our attention on the second step.
The desired digital filter is to be used to filter a digital signal that is derived from an analog
signal by means of periodic sampling. The speci_cations for both analog and digital filters are
often given in the frequency domain, as for example in the design of low
pass, high pass, band pass and band elimination filters. Given the sampling rate, it is straight

forward to convert from frequency specifications on an analog _lter to frequency

speci_cations on the corresponding digital filter, the analog frequencies being in terms of Hertz
and digital frequencies being in terms of radian frequency or angle around the unit circle with
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the point Z=-1 corresponding to half the sampling frequency. The least confusing point of
view toward digital filter design is to consider the filter as being specified in terms of angle
around the unit circle rather than in terms of analog frequencies.

Figure 7.1: Tolerance limits for approximation of ideal low-pass filter

A separate problem is that of determining an appropriate set of specifications on the digital

filter. In the case of a low pass filter, for example, the specifications often take the
form of a tolerance scheme, as shown in Fig. 4.1

Many of the filters used in practice are specified by such a tolerance scheme, with no
constraints on the phase response other than those imposed by stability and causality
requirements; i.e., the poles of the system function must lie inside the unit circle. Given a set
of specifications in the form of Fig. 7.1, the next step is to and a discrete time linear system
whose frequency response falls within the prescribed tolerances. At this point the filter design
problem becomes a problem in approximation. In the case of infinite impulse response (IIR)
filters, we must approximate the desired frequency response by a rational function, while in the
finite impulse response (FIR) filters case we are concerned with polynomial approximation.

7.2 Design of IIR Filters from Analog Filters:

The traditional approach to the design of IIR digital filters involves the transformation of an
analog filter into a digital filter meeting prescribed specifications. This is a reasonable
approach because:

 The art of analog filter design is highly advanced and since useful results can be
achieved, it is advantageous to utilize the design procedures already developed for
analog filters.
 Many useful analog design methods have relatively simple closed-form design
formulas.
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Therefore, digital filter design methods based on analog design formulas are rather simple to
implement.
An analog system can be described by the differential equation

------------------------------------------------------------7.1

And the corresponding rational function is

---------------------------------------------------------7.2

The corresponding description for digital filters has the form

--------------------------------------------------7.3
and the rational function

--------------------------------------------------------7.4

In transforming an analog filter to a digital filter we must therefore obtain either H(z)or h(n)
(inverse Z-transform of H(z) i.e., impulse response) from the analog filter design. In such
transformations, we want the imaginary axis of the S-plane to map into the finite circle of the
Z-plane, a stable analog filter should be transformed to a stable digital filter. That is, if the
analog filter has poles only in the left-half of S-plane, then the digital filter must have poles
only inside the unit circle. These constraints are basic to all the techniques discussed

7.3 IIR Filter Design by Impulse Invariance:

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This technique of transforming an analog filter design to a digital filter design corresponds to
choosing the unit-sample response of the digital filter as equally spaced samples of the impulse
response of the analog filter. That is,

-------------------------------------------------------------------------7.5
Where T is the sampling period. Because of uniform sampling, we have

---------------------------------------------7.6

Or

---------------------------------------------7.7

Figure 7.2: Mapping of s-plane into z-plane

Where s = jω and Ω=ω/T, is the frequency in analog domain and ω is the frequency in digital
domain.
From the relationship Z = eST it is seen that strips of width 2π/T in the S-plane map into the
entire Z-plane as shown in Fig. 7.2. The left half of each S-plane strip maps into interior of the
unit circle, the right half of each S-plane strip maps into the exterior of the unit circle, and the
imaginary axis of length 2π/T of S-plane maps on to once round the unit circle of Z-plane.
Each horizontal strip of the S-plane is overlaid onto the Z-plane to form the digital filter
function from analog filter function. The frequency response of the digital filter is related to
the frequency response of the
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Figure 7.3: Illustration of the effects of aliasing in the impulse invariance technique

analog filter as

------------------------------------------------7.8

From the discussion of the sampling theorem it is clear that if and only if

Then

Unfortunately, any practical analog filter will not be band limited, and consequently there is
interference between successive terms in Eq. (7.8) as illustrated in Fig. 7.3. Because of the
aliasing that occurs in the sampling process, the frequency response of the resulting digital
filter will not be identical to the original analog frequency response. To get the filter design
procedure, let us consider the system function of the analog filter expressed in terms of a
partial-fraction expansion

-----------------------------------------------------------------------7.9

The corresponding impulse response is

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--------------------------------------------------------------- 7.10

And the unit-sample response of the digital filter is then

--------------7.11

The system function of the digital filter H(z) is given by

------------------------------------------------------------7.12

In comparing Eqs. (7.9) and (7.12) we observe that a pole at s=sk in the S-plane transforms to
a pole at expskT in the Z-plane. It is important to recognize that the impulse invariant design
procedure does not correspond to a mapping of the S-plane to the Z-plane.

8.4 IIR Filter Design By Approximation Of Derivatives:

A second approach to design of a digital filter is to approximate the derivatives in Eq. (4.1) by
finite differences. If the samples are closer together, the approximation to the derivative would
be increasingly accurate. For example, suppose that the first derivative is approximated by the
first backward difference

--------------------------7.13

Where y(n)=y(nT). Approximation to higher-order derivatives are obtained by repeated

application of Eq. (7.13); i.e.,

-------------------------- 7.14
For convenience we define

-------------------------------------------------------------------7.15
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Applying Eqs. (7.13), (7.14) and (7.15) to (7.1), we obtain

---------------------------------------------7.16

Where y(n) = ya(nT) and x(n) = xa(nT). We note that the operation ∆(1)[ ] is a linear shift-
invariant operator and that ∆(k)[ ] can be viewed as a cascade of (k) operators ∆(1)[ ]. In
particular

And

Thus taking the Z-transform of each side in Eq. (7.16), we obtain

------------------------------------------------------------7.17

Comparing Eq. (7.17) to (7.2), we observe that the digital transfer function can be obtained
directly from the analog transfer function by means of a substitution of variables

---------------------------------------------------------------------------------7.18

So that, this technique does indeed truly correspond to a mapping of the S-plane to the Z-
plane, according to Eq. (7.18). To investigate the properties of this mapping, we must express
z as a function of s, obtaining

Substituting s = jΩ, i.e., imaginary axis in S-plane

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------------------------------------------------------7.19

Which corresponds to a circle whose center is at z =1/2 and radius is 1/2, as shown in Fig. 7.4.
It is easily verified that the left half of the S-plane maps into the inside of the small circle and
the right half of the S-plane maps onto the outside of the small circle. Therefore, although the
requirement of mapping the jΩ-axis to the unit circle is not satisfied, this mapping does satisfy
the stability condition.

Figure 4.4: Mapping of s-plane to z-plane corresponding to first backward-difference

approximation to the derivative

In contrast to the impulse invariance technique, decreasing the sampling period T, theoretically
produces a better filter since the spectrum tends to be concentrated in a very small region of
the unit circle. These two procedures are highly unsatisfactory for anything but low pass
filters. An alternative approximation to the derivative is a forward difference and it provides a
mapping into the unstable digital filters.

8.5 IIR Filter Design By The Bilinear Transformation:

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In the previous section a digital filter was derived by approximating derivatives by differences.
An alternative procedure is based on integrating the differential equation and then using a
numerical approximation to the integral. Consider the first - order equation

-----------------------------------------------------------7.20

Where y’a(t) is the first derivative of ya(t). The corresponding analog system function is

We can write ya(t) as an integral of y’a(t), as in

In particular, if t = nT and t0 = (n - 1)T,

If this integral is approximated by a trapezoidal rule, we can write

----------------------7.21

However, from Eq. (7.20),

Substituting into Eq. (4.21) we obtain

Where y(n) = y(nT) and x(n) = x(nT). Taking the Z-transform and solving for H(z) gives

--------------------------------------------7.22
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From Eq. (7.22) it is clear that H(z) is obtained from Ha(s) by the substitution

-------------------------------------------------------------------7.23

That is,

--------------------------------------------------------------7.24

This can be shown to hold in general since an Nth - order differential equation of the form of
Eq. (7.1) can be written as a set of N first-order equations of the form of Eq. (7.20). Solving
Eq. (7.23) for z gives

----------------------------------------------------------------------------7.25
The invertible transformation of Eq. (7.23) is recognized as a bilinear transformation. To see
that this mapping has the property that the imaginary axis in the s-plane maps onto the unit
circle in the z-plane, consider z = ejω, then from Eq. (7.23), s is given by
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Figure 7.5: Mapping of analog frequency axis onto the unit circle using the bilinear
Transformation

Thus for z on the unit circle, σ = 0 and Ω and ω are related by

T Ω/2 = tan (ω/2)

or
ω = 2 tan -1(T Ω/2)

This relationship is plotted in Fig. (7.5), and it is referred as frequency warping. From the
_gure it is clear that the positive and negative imaginary axis of the s-plane are mapped,
respectively, into the upper and lower halves of the unit circle in the z-plane. In addition to the
fact that the imaginary axis in the s-plane maps into the unit circle in the z-plane, the left half
of the s-plane maps to the inside of the unit circle and the right half of the s-plane maps to the
outside of the unit circle, as shown in Fig. (7.6). Thus we see that the use of the bilinear
transformation yields stable digital filter from analog filter. Also this transformation avoids the
problem of aliasing encountered with the use of impulse invariance, because it maps the entire
imaginary axis in the s-plane onto the unit circle in the z-plane. The price paid for this,
however, is the introduction of a distortion in the frequency axis.
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Figure 4.6: Mapping of the s-plane into the z-plane using the bilinear transformation

8.6 The Matched-Z Transform:

Another method for converting an analog filter into an equivalent digital filter is to map
the poles and zeros of Ha(s) directly into poles and zeros in the z-plane. For analog filter

-----------------------------------------------------------------7.26

the corresponding digital filter is

---------------------------------------------------------7.27

Where T is the sampling interval. Thus each factor of the form (s-a) in Ha(s) is mapped
into the factor (1- eaT z-1).
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Recommended questions with solution

Question 1
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Question 2

Question 3
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Question 4

Question 5
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Question 6
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Question 7
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