Introduction to IIR filter design

Department of Information Technology

SSN College of Engineering

. p.1/15

The S-domain

The Laplace transform can be used to analyze a large class of continuous-time problem
involving signals that are not absolutely integrable, such as impulse response of an unstable
system.
Whereas, Fourier transform does not exist for signals that are not absolutely integrable
(summable, in discrete-domain).
Let est be a complex exponential with complex frequency s = + j. The real part of s is
the exponential damping factor , and the imaginary part of s is the frequency of the cosine
and sine factor, .
est = et cos(t) + jet sin(t)

(1)

Let H(s) be a transfer function of a LTI system.

A pole at s = dk of H(s) is the left-half of the s-plane contributes an exponentially
decaying term to the impulse response.
A pole in the right-half of the s-plane contributes an increasing exponential term to the
impulse response.

. p.2/15

Converting analog filter into a digital filter

Analog filter design > matured and well developed field
Design a filter in analog-domain > convert the design in digital-domain
An analog-filter can be described by its system function Ha (s) as given below:
PM
k
B(s)
k=0 k s
= PN
Ha (s) =
k
A(s)
k=0 k s

(2)

where k and k are the filter coefficients. Ha (s) can be described by its impulse response as given
below:
Z
h(t)est dt
(3)
Ha (s) =

It can be described by a constant-coefficient differential equation as,

N
X

M
X
dk y(t)
dk x(t)
k
=
k
k
dt
dtk
k=0
k=0

(4)

where x(t) is the input signal and y(t) is the output of the filter.

. p.3/15

Analog filer to digital filter

Each of these equivalent characterization of an anlog filter leads to alternative methods for
converting the filter into digital-domain.
Ha (s) is stable if all its poles lie in the left-half of the s-plane.
The conversion technique is effective if the following conditions are met:
j axis in s-plane should map into the unit circle in the z-plane. This ensures a direct
relationship between two frequency variables in the two domains.
The left-half of the s-plane should map into the inside of the unit circle. This ensures
that a stble analog filter will be converted to a stable digital filter.
IIR filter cannot have linear phase.
Ideal filter > h(n) is of infinite length
Linear phase > h(n) should be symmetrical (two-sided with point of symmetry at
n=0
Two-sided (symmetrical) signal > ROC in Z-domain is an annular ring that includes
unit circle (so that Fourier transform exists)
ROC to be annular (including unit circle) > for each pole, a complex conjugate pole
exactly at the reciprocal location (outside the unit circle).
It is not causal. Linear phase IIR filter cannot be causal.

. p.4/15

IIR filter design by Approximation of Derivatives

Approximate the differential equation (given in equation (4)) by an equivalent difference equation
For the derivative

dy(t)
dt

at time t = nT , backward difference y(nT ) y(nT 1)/T is substituted.

dy(t)

dt t=nT

=
=

y(nT ) y(nT T )
T
y(n) y(n 1)
T

(5)

where T represents the sampling interval. The analog differentiator with output dy(t)/dt has the system function H(s) = s, while
the digital system that produces the output y(n) y(n 1)/T has the system function H(z) = (1 z 1 )/T . Consequently,
1
s = 1z
. The second order differentiation can be written as,
T

d2 y(t)
dt2

=
=
=
=

d [y(n) y(n 1)]

dt
1 d
T dt

T
[y(n) y(n 1)]

(6)

1 [y(n) y(n 1)] [y(n 1) y(n 2)]

T

[y(n) 2y(n 1) + y(n 2)]

T2

(7)
(8)

. p.5/15

Approximation of derivatives(Contd...)
d2 y(t)
dt2

is has the system function H(s) = s2 , while the digital system that produces the

2
[y(n)2y(n1)+y(n2)]
1z 1
output
has the system function H(z) =
. In general, we can write it as,
2
T
The analog second-order differentiator
T

1 z 1 k

(9)

The above equation can be written as,

z=

(10)

1 sT

If we substitute s = j (setting = 0), and by simple manipulation, it can be written as,

z=

1
1 + 2 T 2

+j

T
1 + 2 T 2

(11)

Substituting z = rej , we get,

re

1
1 + 2 T 2

+j

T
1 + 2 T 2

(12)

The squared-magnitude of equation (12) gives,

1
(1 + 2 T 2 )2

2 T 2
(1 + 2 T 2 )2

. p.6/15
(13)

Approximation of derivatives (Contd...)

We can easily show that the RHS of equation (13) is equal to

1
1+2 T 2

, which is same as the real part of equation (11). Here, let us

consider the real part of equation (11) as x and the imaginary part as y. In that case, the x and y are related to each other as given below:
2

x +y

=x

(14)

This can be written as

1 2
1
2
x
+y =
2
2

(15)

1 and centred at x = 1 .
Equation (15) is an equation of a circle with radius 2
2

Observations:
centered at 1
on
From equation (15), we can infer that the left-half of the s-plane mapped into inside the circle (with radius 1
2
2
the real axis of z-plane, as shown in the figure (next slide).
The j axis is mapped onto the circle.
In the z-plane, even at =
, we cannot have any poles. This implies that with this kind of conversion, only a low-pass
2
filter (with very-low cut-off frequency) or band-pass filter (with very-low cut-off frequencies) can be designed.

Exercise:
Convert the analog bandpass filter with system function Ha (s) =

1
(s+0.1)2 +9

into a digital filter by the use of

backward difference for the derivatives. Assume T = 0.1 and simplify the resultant filter.

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Approximation of derivatives

j
Unit Circle

z-plane

s-plane

. p.8/15

Impulse Invariance Method:

Let the system function of the analog filter in partial-fraction form be

Ha (s) =

N
X

k=1

ck
s pk

(16)

where {pk } are the poles of the analog filter and {ck } are the coefficients in the partial-fraction
expansion. In inverse Laplace transform of this can be given as,

ha (t) =

N
X

ck epk t

t0

(17)

k=1

If we sample ha (t) periodically at t = nT , we have

h(n)

ha (nT )

N
X

ck epk nT

(18)

k=1

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The z-transform of the impulse response of the filter h(n) is

H(z)

h(n)z n

n=0

N
X
X

n=0

k=1

ck epk nT z n

N
X

(epk T z 1 )n

ck

k=1

(19)

n=0

By applying geometic series formula,

H(z) =

N
X

k=1

ck
1 epk T z 1

(20)

We observe that the digital filter has poles at

zk = epk T

k = 1, 2, . . . , N

(21)

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Impulse Invariance Method (Contd...)

Let us consider the mapping of points from the s-plane to z-plane implied by the relation z = esT . If
we substitute s = + j and z = rei , then
rei = eT ejT

(22)

clearly, we must have

r = eT

&

= T

(23)

Consequently,
If = 0, we have r = 1
If < 0, then 0 < r < 1
If > 0, them r > 1
The LHP in s is mapped inside the unit circle in z and the RHP in s is mapped outside the unit
circle in z. The j-axis is mapped into the unit circle.
However, /T /T maps into . Poles outside this range also mapped
into .
Suitable for lowpass and bandpass filter design only

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Impulse Invariance

3
T

Unit circle

z-plane

s-plane

3
T

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Exercises:
Convert the analog filter with system function Ha (s) =
by means of the impulse invariance method.

s+0.1
(s+0.1)2 +9

into a digital IIR filter

. p.13/15

Bilinear Transformation
Approximation of derivatives method or impulse invariance method > for designing a lowpass or bandpass filter with very
low cut-off frequencies
Bilinear transformation > conformal mapping technique
j axis in s-plane > mapped into the unit circle only once
LHP of s-plane > inside the unit circle
RHp of s-plane > outside the unit circle
Let us consider an analog linear filter with system function

H(s) =

b
s+a

(24)

This system is also characterized by the differential equation

dy(t)
dt

+ ay(t) = bx(t)

(25)

Instead of substituting a finite difference for the derivative, intergrate the derivate and approximate the integral using trapezoidal formula
as give below:
Z t

y ( )d + y(t0 )
(26)
y(t) =
t0

Here y (t) denotes the derivative of y(t).

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Bilinear transformation (Contd...)

The approximation of the integral (given in equation (26)) by trapezoidal formula at t = nT and t0 = nT T yields

y(nT ) =

[y (nT ) + y (nT T )] = y(nT T )

(27)

At t = nT , the equation (26) can be written as,

y (nT ) = ay(nT ) + bx(nT )

(28)

Representing nT by n in equations (27) and (28) and substituting (28) in (27), we get

aT
aT
bT
y(n) 1
y(n 1) =
[x(n) + x(n 1)]
1+
2
2
2

(29)

By taking the z-transform of equation (29) and rearranging, we get

b

H(z) =
2
T

1z 1
1+z 1

(30)
+a

By comparing equations (24) and (30), we can say that

s=

2
T

1 z 1
1 + z 1

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