Design of Digital Filters

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 finite 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 specifications 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 filter to frequency

specifications 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 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.

1
 Figure 4.1: 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.

1 − δ1 ≤| H(ejω ) | ≤ 1, | ω |≤ ωp

| H(ejω ) | ≤ δ2 , ωs ≤| ω |≤ π

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. 4.1, the next step is to find 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

2
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.

4.1 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 for-

mulas. 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

N M
X dk ya (t) X dk xa (t)
ck = d k (4.1)
k=0 dtk k=0 dtk

And the corresponding rational function is

PM
dk sk ya (s)
Ha (s) = Pk=0
N k
= (4.2)
k=0 ck s xa (s)

The corresponding description for digital filters has the form

3
 N
X M
X
ak y(n − k) = bk x(n − k) (4.3)
k=0 k=0

and the rational function

PM
k=0 bk z −k Y (z)
H(z) = PN = (4.4)
k=0 ak z
−k X(z)

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 nit

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 here.

4.1.1 IIR Filter Design By Impulse Invariance:

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,

h(n) = ha (nT ) (4.5)

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

jωT 1 X ∞
2π
H(e )= Ha (jΩ + j k) (4.6)
T k=−∞ T

4
 Figure 4.2: Mapping of s-plane into z-plane

or

1 X ∞
2π
H(z) |z=esT = Ha (s + j k) (4.7)
T k=−∞ T

where s = jΩ and Ω = ω/T , Ω 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. 4.2. 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

5
 Figure 4.3: Illustration of the effects of aliasing in the impulse invariance technique

analog filter as

1 X ∞
ω 2π
H(ejω ) = Ha (j + j k) (4.8)
T k=−∞ T T

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

π
Ha (jΩ) = 0, | Ω |≥
T

Then

1 ω
H(ejω ) = Ha (j ), | ω |≤ π
T T

Unfortunately, any practical analog filter will not be band limited, and consequently

there is interference between successive terms in Eq. (4.8) as illustrated in Fig. 4.3.

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

6
 N
X Ak
Ha (s) = (4.9)
k=1 s − sk

The corresponding impulse response is

N
Ak esk t U (t)
X
ha (t) = (4.10)
k=1

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

N N
n
Ak esk nT u(n) = Ak (esk T ) U (n)
X X
h(n) = ha (nT ) = (4.11)
k=1 k=1

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

N
X Ak
H(z) = sk T z −1 )
(4.12)
k=1 (1 − exp

In comparing Eqs. (4.9) and (4.12) we observe that a pole at s=sk in the S-plane

transforms to a pole at expsk T 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.

4.1.2 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
 dya (t) y(n) − y(n − 1)
|t=nT −→ ∇(1) [y(n)] = (4.13)
dt T

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

peated application of Eq. (4.13); i.e.,

dk ya (t) d dk−1 ya (t)

| t=nT = ( ) |t=nT −→ ∇(k) [y(n)] = ∇(1) [∇(k−1) [y(n)]] (4.14)
dtk dt dtk−1

For convenience we define

∇(0) [y(n)] = y(n) (4.15)

Applying Eqs. (4.13), (4.14) and (4.15) to (4.1), we obtain

N M
ck ∇(k) [y(n)] = dk ∇(k) [x(n)]
X X
(4.16)
k==0 k=0

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,

1 − z −1
Z[∇(1) [x(n)]] = [ ]X(z)
T

and

1 − z −1 k
Z[∇(1) [x(n)]] = [ ] X(z)
T

Thus taking the Z-transform of each side in Eq. (4.16), we obtain

8
 PM −1
dk [ 1−zT ]k
k=0
H(z) = P 1−z −1 k
(4.17)
k=0 N ck [ T ]

Comparing Eq. (4.17) to (4.2), we observe that the digital transfer function can be

obtained directly from the analog transfer function by means of a substitution of variables

1 − z −1
s= (4.18)
T

So that, this technique does indeed truly correspond to a mapping of the S-plane to

the Z-plane, according to Eq. (4.18). To investigate the properties of this mapping, we

must express z as a function of s, obtaining

1
z=
1 − sT

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

1
z =
1 − jΩT
1 1 1
= + −
1 − jΩT 2 2
1 1 1 + jΩT
= + [ ]
2 2 1 − jΩT
1 1 + jΩT
= [1 + ]
2 1 − jΩT
1 −1
= [1 + ej2 tan (ΩT ) ] (4.19)
2

Which corresponds to a circle whose center is at z=1/2 and radius is 1/2, as shown

an Fig. 4.4. 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

9
Figure 4.4: Mapping of s-plane to z-plane corresponding to first backward-difference ap-
proximation to the derivative

circle. Therefore, although the requirement of mapping the jΩ -axis to the unit circle is

not satisfied, this mapping does satisfy the stability condition.

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.

4.1.3 IIR Filter Design By The Bilinear Transformation:

In the previous section a digital filter was derived by approximating derivatives by differ-

ences. An alternative procedure is based on integrating the differential equation and then

using a numerical approximation to the integral. Consider the first - order equation

10
 0
c1 ya (t) + c0 ya (t) = d0 xa (t) (4.20)

0
Where ya (t) is the first derivative of ya (t). The corresponding analog system function

is

d0
Ha (s) =
c0 + c 1 s
0
We can write ya (t) as an integral of ya (t), as in

Z t 0
ya (t) = ya (t)dt + ya (t0 )
t0

In pparticular, if t = nT and t0 = (n − 1)T ,

Z nT 0
ya (nT ) = ya (τ )dτ + ya ((n − 1)T )
(n−1)T

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

T 0 0
ya (nT ) = ya ((n − 1)T ) + [ya (nT ) + ya ((n − 1)T )] (4.21)
2

However, from Eq. (4.20),

0 c0 d0
ya (nT ) = − ya (nT ) + xa (nT )
c1 c1

Substituting into Eq. (4.21) we obtain

T c0 d0
[y(n) − y(n − 1)] = [− (y(n) + y(n − 1)) + (x(n) + x(n − 1))]
2 c1 c1

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

gives

11
 Y (z) d0
H(z) = = (4.22)
c0 + c1 T2 1−z
−1
X(z) 1+z −1

From Eq. (4.22) it is clear that H(z) is obtained from Ha (s) by the substitution

2 1 − z −1
s= (4.23)
T 1 + z −1

That is,

H(z) = Ha (s) |s= 2 1−z−1 (4.24)

T 1+z −1

This can be shown to hold in general since an N th - order differential equation of the

form of Eq. (4.1) can be written as a set of N first-order equations of the form of Eq.

(4.20). Solving Eq. (4.23) for z gives

1 + T2 s
z= (4.25)
1 − T2 s

The invertible transformation of Eq. (4.23) 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. (4.23), s is given by

2 1 − e−jω
s =
T 1 + e−jω
2 j sin(ω/2)
=
T cos(ω/2)
2
= j tan(ω/2)
T
= σ + jΩ

12
Figure 4.5: 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

TΩ
= tan(ω/2)
2

or

ω = 2 tan−1 (ΩT /2)

This relationship is plotted in Fig. (4.5), and it is referred as frequency warping.

From the figure 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. (4.6).

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

13
 Figure 4.6: Mapping of the s-plane into the z-plane using the bilinear transformation

onto the unit circle in the z-plane. The price paid for this, however, is the introduction

of a distortion in the frequency axis.

4.1.4 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

QM
k=1 (s − sk )
Ha (s) = QN (4.26)
k=1 (s − pk )

the corresponding digital filter is

QM
k=1 (1 − ezk T z −1 )
H(z) = QN pk T z −1 )
(4.27)
k=1 (1 − e

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 ).

14
4.1.5 Characteristics Of Commonly Used Analog Filters:

From the previous discussion it is clear that, IIT digital filters can be obtained by begin-

ning with an analog filter. Thus the design of a digital filter is reduced to designing an

appropriate analog filter and then performing the conversion from Ha (s) to H(z).

Analog filter design is a well - developed field, many approximation techniques, viz.,

Butterworth, Chebyshev, Elliptic, etc., have been developed for the design of analog low

pass filters. Our discussion is limited to low pass filters, since, frequency transformation

can be applied to transform a designed low pass filter into a desired high pass, band pass

and band stop filters.

Butterworth Filters:

Low pass Butterworth filters are all - pole filters with monotonic frequency response

in both pass band and stop band, characterized by the magnitude - squared frequency

response

1 1
| Ha (Ω) |2 = 2N
= (4.28)
1 + (Ω/Ωc ) 1 +  (Ω/Ωp )2N
2

Where, N is the order of the filter, Ωc is the -3dB frequency, i.e., cutoff frequency,

Ωp is the pass band edge frequency and 1/(1 + 2 ) is the band edge value of | Ha (Ω) |2 .

Since the product Ha (s)Ha (−s) and evaluated at s = jΩ is simply equal to | Ha (Ω) |2 , it

follows that

1
Ha (s)Ha (−s) = 2 (4.29)
1 + ( −s
Ω2
)N
c

15
The poles of Ha (s)Ha (−s) occur on a circle of radius Ωc at equally spaced points. From

Eq. (4.29), we find the pole positions as the solution of

−s2
= (−1)1/N = ej(2k+1)π/N , k = 0, 1, , · · · , N − 1 (4.30)
Ω2c

and hence, the N poles in the left half of the s-plane are

sk = Ωc ejπ/2 e(2k+1)π/2N , k = 0, 1, · · · , N − 1

= σk + jΩk (4.31)

Note that, there are no poles on the imaginary axis of s-plane, and for N odd there

will be a pole on real axis of s-plane, for N even there are no poles even on real axis of

s-plane. Also note that all the poles are having conjugate symmetry.

Thus the design methodology to design a Butterworth low pass filter with δ2 attenu-

ation at a specified frequency Ωs is

Find N,
log[(1/δ22 ) − 1] log(δ/)
N= = (4.32)
2 log(Ωs /Ωc ) log(Ωs /Ωp )
√
where by definition, δ2 = 1/ 1 + δ 2 . Thus the Butterworth filter is completely charac-

terized by the parameters N, δ2 ,  and the ratio Ωs /Ωp or Ωc .

Then, from Eq. (4.31) find the pole positions sk , k = 0, 1, · · · , N − 1. Finally the

analog filter is given by

N
Y 1
Ha (s) = (4.33)
k==1 (s − sk )

16
Chebyshev Filters:

There are two types of Chebyshev filters. Type I Chebyshev filters are all-pole filters that

exhibit equiripple behavior in the pass band and a monotonic characteristic in the stop

band. On the other hand, type II Chebyshev filters contain both poles and zeros and

exhibit a monotonic behavior in the pass band and an equiripple behavior in the stop

band. The zeros of this class of filters lie on the imaginary axis in the s-plane.

The magnitude squared of the frequency response characteristic of type I Chebyshev

filter is given as

1
| Ha (Ω) |2 = (4.34)
1+ 2 TN2 (Ω/Ωp )

where  is a parameter of the filter related to the ripple in the pass band as shown in Fig.

(4.7), and TN is the N th order Chebyshev polynomial defined as

(
cos(N cos−1 x), | x |≤ 1
TN (x) = (4.35)
cosh(N cosh−1 x), |x|>1

The Chebyshev polynomials can be generated by the recursive equation

TN +1 (x) == 2xTN (x) − TN −1 (x), N = 1, 2, · · ·

where T0 (x) = 1 and T1 (x) = x.

At the band edge frequency Ω = Ωp , we have

1
√ = 1 − δ1
1 + 2

17
 Figure 4.7: Type I Chebysehev filter characteristic

or equivalently

1
2 = −1 (4.36)
(1 − δ1 )2

where δ1 is the value of the pass band ripple.

The poles of Type I Chebyshev filter lie on an ellipse in the s-plane with major axis

β2 + 1
r1 = Ω p (4.37)
2β

and minor axis

β2 − 1
r2 = Ω p (4.38)
2β

where β is related to  according to the equation

√
1 + 2 + 1 1/N
β=[ ] (4.39)


The angular positions of the left half s-plane poles are given by

18
 π (2k + 1)π
φk = + , k = 0, 1, · · · , N − 1 (4.40)
2 2N

Then the positions of the left half s-plane poles are given by

sk = σk + jΩk , k = 0, 1, · · · , N − 1 (4.41)

where σk = r2 cos φk and Ωk = r1 sin φk . The order of the filter is obtained from

q q
log[( 1 − δ22 + 1 − δ22 (1 + 2 ))/δ2 ]
N = Ωs
q
Ωs 2
log[ Ω p
+ (Ω p
) − 1]
cosh−1 ( δ )
= Ωs (4.42)
cosh−1 ( Ω p
)

where, by definition δ2 = √ 1 .
1+δ 2

Finally, the Type I Chebyshev filter is given by

N
Y 1
Ha (s) =
k=1 (s − sk )

A Type II Chebyshev filter contains zero as well as poles. The magnitude squared

response is given as
1
| Ha (−Ω) |2 = Ωs (4.43)
1+ 2 [TN2 ( Ω p
)/TN2 ( ΩΩs )]

where TN (x) is the N-order Chebyshev polynomial. The zeros are located on the imaginary

axis at the points

Ωs
zk = j , k = 0, 1, . . . , N − 1 (4.44)
sin φk

and the left-half s-plane poles are given

19
 sk = σk + jΩk , k = 0, 1, . . . , N − 1 (4.45)

where
Ωs r2 cos φk
σk = q (4.46)
r22 cos2 φk + r12 sin2 φk

and
Ωs r1 sin φk
Ωk = q (4.47)
r22 cos2 φk + r12 sin2 φk

Finally, the Type II Chebyshev filter is given by

N
Y s − zk
Ha (s) = (4.48)
k=1 s − sk

The other approximation techniques are elliptic (equiripple in both passband and

stopband) and Bessel (monotonic in both passband and stopband).

4.2 Examples:

I Design a digital filter to satisfy the following characteristics.

(i) -3dB cutoff frequency of 0.5π rad.

(ii) Magnitude down at least 15dB at 0.75π rad.

(iii) Monotonic stop band and pass band

Using

(a) Impulse invariant technique

(b) Approximation of derivatives

(c) Bilinear transformation technique

20
 Figure 4.8: Frequency response plot of the example

Solution

a) Impulse Invariant Technique

From the given digital domain frequency, find the corresponding analog domain fre-

quencies.

ωc ωs
Ωc = T
and Ωs = T

where T is the sampling period and 1/T is the sampling frequency and it always

corresponds to 2π radians in the digital domain. In this problem, let us assume T = 1

sec.

Then Ωc = 0.5π and Ωs = 0.75π

Let us find the order of the desired filter using

( δ12 − 1)
N= 2

2 log( Ω
Ωc
s
)

21
Where δ2 is the gain at the stop band edge frequency ωs .

−15 dB = 20 log δ2

δ2 = 20 log δ2

π
δ2 = 10− 20 = 0.1778

1
log( (0.1778) 2 − 1
N= = 4.219 ' 5
2 log( 0.75π
0.5π
)

Order of filter N =5.

Then the 5 poles on the Butterworth circle of radius Ωc = 0.5π are given by

π π
s0 = 0.5πej( 2 + 10 ) = −0.485 + j1.493
π 3π
s1 = 0.5πej( 2 + 10 ) = −1.27 + j0.923
π 5π
s2 = 0.5πej( 2 + 10 ) = −1.57 + j0.0
π 7π
s3 = 0.5πej( 2 + 10 ) = −1.27 − j0.923
π 9π
s5 = 0.5πej( 2 + 10 ) = −0.485 − j1.493

Then the filter transfer function in the analog domain is

1
Ha (s) =
(s + 0.485 − j1.493)(s + 1.27 − j0.923)(s + 1.57)(s + 1.27 + j0.923)(s + 0.485 + j0.923)
5
X Ak
=
k=1 (s − sk )

where Ak ’s are partial fractions coefficients of Ha (s).

22
 Finally, the transfer function of the digital filter is

5
X Ak
H(z) = sk −1
, where sk ’s are the poles on the Butterworth circle
k=1 (1 − e z )

b)

H(z) = Ha (s) |s= 1−z−1 =1−z −1

T

5
X 1
H(z) =
k=1 (1 − z −1 − sk )

c) For the bilinear transformation technique, we need to prewarp the digital frequencies

into corresponding analog frequencies.

2
i.e., Ω = T
tan( ω2 )

0.5π
Ωc = 2 tan( ) = 2 rad.
2

and
0.75π
Ωs = 2 tan( ) = 4.828 rad.
2

Then the order of the filter

1
log( (0.1778) 2 − 1)
N=
2 log( 4.828
2
)

The pole locations on the Butterworth circle with radius Ωc = 2 are

π π
s0 = 2ej( 2 + 4 ) = −1.414 + j1.414
π 3π
s1 = 2ej( 2 + 4 ) = −1.414 − j1.414

Then the filter transfer function in the analog domain is

1
Ha (s) =
(s + 1.414 − j1.414)(s + 1.414 + j1.414)

23
Finally, the transfer function of the digital filter is

H(z) = Ha (s) |s= 2 1−z−1 =2 1−z−1

T 1+z −1 1+z −1

1
H(z) = 1−z −1 1−z −1
(2 1+z −1 + 1.414 − j1.414)(2 1+z −1 + 1.414 + j1.414)

II Design a digital filter using impulse invariant technique to satisfy following charac-

teristics

(i) Equiripple in pass band and monotonic in stop band

(ii) -3dB ripple with pass band edge frequency at 0.5π radians.

(iii) Magnitude down at least 15dB at 0.75π radians.

Solution: Assuming T=1, Ω = 0.5π and Ωs = 0.75π

The order of desired filter is

q q
log[( 1 − δ22 ) + 1 − δ22 (1 − 2 ))/δ2 ]
N= Ωs
q
Ωs 2
log[ Ω p
+ (Ω p
) − 1]

when
1
20 log √ = −3|mboxdB
1 + 2

i.e.,

10 log(1 + 2 ) = 3dB

2 = 100.3 − 1 = 0.9952

 = 0.9976

and

20 log δ2 = −15 dB

24
 δ2 = 10−0.75 = 0.1778

Hence
q q
[( 1 − (0.1778)2 + 1 − (0.1778)2 (1 + 0.9952))/0.9976 × 0.1778]
N = q
log[ 0.75π
0.5π
+ ( 0.75π
0.5π
)2 − 1]
= 2.48

' 3

The order of filter, N = 3.

The 3 poles on the ellipse are determined by

√ √
1 + 2 + 1 1 1 + 0.99762 + 1 1
β=[ ]N = [ ] 3 = 1.342
 0.9976

β2 + 1
r1 = Ωp
2β
(1.341)2 + 1
= 0.5π ×
2 × 1.341
= 1.639

β2 − 1
r2 = Ω p
2β
(1.341)2 − 1
= 0.5π ×
2 × 1.341
= 0.469

The angles,
π (2k + 1)π
φk = + , k = 0,1 ,2
2 2N

The poles are at

sk = r2 cos φk + jr1 sin φk

25
 4π 4π
s0 = 0.469 cos( ) + j1.639 sin( )
6 6
= −0.2345 + j1.419

s1 = 0.469 cos(π) + j1.639 sin(π)

= −0.469 + j0.0

8π 8π
s2 = 0.469 cos( ) + j1.6939 sin( )
6 6
= −0.2345 − j1.419

The analog filter transfer function is given by

1
Ha (s) =
s + 0.2345 − j1.419)(s + 0.469) + (s + 0.2345 + j1.419)
3
X Ak
=
k=1 (s − sk )

where Ak ’s are the partial fraction coefficients.

Finally, the digital filter transfer function is given by

3
X Ak
H(z) = sk −1 )
k=1 (1 − e z

26