1

Class notes of EKT353 by Prof. Dr. Farid Ghani

Digital Filters

Introduction

A digital filter is a mathematical algorithm implemented in
hardware/ software that operates on a digital input to produce a
digital output. Digital filters play very important roles in DSP.
Compares with analog filters they are preferred in a number of
applications like data compression, speech processing, image
processing, etc., because of the following reasons

1. Digital filters can have characteristics that are not possible
with analog filters such as linear phase response.
2. The performance of digital filter does not vary with
environmental changes, for example thermal variations.
3. The frequency response of a digital filter can be adjusted if
it is implemented using a programmable processor.
4. Several input signals can be processed by one digital filter
without the need to replicate the hardware.
5. Digital filters can be used at very low frequencies.

The following are the main disadvantages of digital filters as
compared to the analog filters

1. Speed limitations
2. Finite word length effects
3. Long design and development times

Digital filters are classified as Finite Duration Impulse
Response (FIR) filters also called Non-Recursive Digital
Filters or Infinite Duration Impulse Response (IIR) filters also
called Recursive Digital Filters, depending upon the form of
the unit impulse response of the filter. In the FIR filter, the
discrete impulse response of the filter is of finite duration, i.e. it
has finite number of non-zero terms. The IIR filter has an infinite
2

Class notes of EKT353 by Prof. Dr. Farid Ghani

number of non-zero terms, i.e. its discrete impulse response is
of infinite duration. IIR filters are usually implemented using
structures having feedbacks (recursive structures- poles and
zeros) and FIR filters usually implemented using structures with
no feed- back (non-recursive structures all zeros).
FIR filters have the following advantages over IIR filters

1. They can have exact linear phase
2. They are always stable
3. The design methods are generally linear
4. They can be efficiently realized in hardware
5. The filter start-up transients have finite duration.

FIR filters are employed in filtering problems where linear phase
characteristic within the pass-band of the filter is required. If this
is not required, either an IIR or an FIR filter may be employed.
An IIR filter has lesser number of side lobes in the stop band
than an FIR filter with the same number of parameters. For this
reason if some phase distortion is tolerable, an IIR filter is
preferred. Also the implementation of an IIR filter involves fewer
parameters, less memory requirements and lower
computational complexity.

Representation of Digital Filters

Digital filters are normally represented in the following two
manners.

1. Block diagram representation.
2. Mathematical models.

Block Diagram Representation

In this form of representation each box (block) is an operation
on the input signal and the operation is shown on the box
itself. Some typical operations include integration,
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Class notes of EKT353 by Prof. Dr. Farid Ghani

differentiation, delay, summation, accumulation, multiplication
etc. An example of a digital filter is shown in figure below.

H1
H4
H2 H3 + +
DELAY
X(n)
Y1(n) =
H1(x(n))
Y4(n)
Y5(n)=H3(y4(n))
Y3(n)=H4(y(n))
Y2(n)=H2(x(n))
Y(n)
Y6(n)= y1(n-1)


Representation Using Mathematical Models

A filter can also be represented using mathematical models.
The mathematical model of a filter consists of mathematical
equations relating the signals of interests. Digital filters can be
modeled using

1. A linear difference equation.
2. The impulse response sequence or discrete impulse
respone.
3. The transfer function.

Linear difference equation

In a linear digital filter the input sequence x(n) is transformed
into an output sequence y(n) according to some difference
equation. For example

y(n) = x(n) + 3x(n-1) + 2x(n-2)

is a linear difference equation where the nth member of the
output sequence y(n) is obtained by adding or accumulating the
nth member of the input sequence with thrice the (n-1)th
member and twice the (n-2)nd member of the input sequence .
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Class notes of EKT353 by Prof. Dr. Farid Ghani

The block diagram representation of the difference equation is
as shown in the diagram.



















In general a linear time invariant digital filter with input x(n) and
output y(n) is characterized by a linear constant coefficient
difference equation (LCCDE) of the form


= =
=
q
0 k
p
1 k
k) a(k)y(n k) b(k)x(n y(n)
(1)

where the coefficients a(k) and b(k) are constants and define
the filter. If the difference equation has one or more terms a(k)
that are non-zero the difference equation and the
corresponding filter is called recursive. On the other hand if all
the terms a(k) are zero then the resulting difference equation
and the filter are termed as non-recursive. The order of
the filter is the highest degree of the denominator polynomial.

Unit
delay
Unit
delay
+
x(n)
x(n-1)
x(n-2)
y(n)
3
2

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Class notes of EKT353 by Prof. Dr. Farid Ghani

Just like in the case of linear differential equations the solution
of difference equations requires a set of initial conditions.

Impulse response/Discrete Impulse Response of a filter

The impulse response of a filter is one method of modeling
digital filter. The impulse response sequence or the discrete
impulse response also called the sampled impulse response
of a digital filter is the response of the digital filter to a unit
impulse . Initial conditions of the digital filter are assumed to be
zero. The discrete impulse response is usually represented by
the notation h(n). If y(n) is the digital filter response for an
input x(n) then the response of the digital filter when x(n) =
o(n) is y(n) = h(n).

Digital Filter
n
0
X(n) =
Y(n) = h(n)
n
y(n)=h(n)
o(n)
o(n)







x(n)=a o(n-k) y(n)=ah(n-k)


The impulse response sequence of a digital filter can be
obtained directly by solving the difference equation
describing the system.

Let F[.] denote the response of the digital filter . Thus
Digital
Filter
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Class notes of EKT353 by Prof. Dr. Farid Ghani


y(n) = F[x(n)]
= F
(

= k
k n k x ) ( ) ( o


Since x(k) is a number and not a variable
y(n) =
| | ) ( ) ( k n F k x
k

=
o

or y(n) =

=

k
k n h k x ) ( ) (


or y(n) = x(n)*h(n)

Thus the response y(n) of a digital filter to the input x(n) is
given by the convolution of its impulse sequence h(n) with
the input x(n).

Transfer Function

Consider the input output relationship of a digital filter

y(n) = x(n)*h(n)

Taking z-transform of the above relation, and recalling that
convolution in the time domain gets transformed into
multiplication in the frequency domain we get

Y(z) = X(z)H(z)

where Y(z) is the z-transform of the output y(n), X(z) is the z-
transform of the input x(n) and H(z) is the z-transform of the
discrete impulse response h(n) of the digital filter . Thus

H(z) = Y(z)/ X(z)
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Class notes of EKT353 by Prof. Dr. Farid Ghani


The z-transform of the impulse response sequence of the digital
filter which is also the ratio of the z-transform of the output to
the z-transform of the input is called the transfer function or
the system function of the digital filter.

We thus see that the transfer function is normally
expressed as a ratio of two polynomials in z that is it is a
rational function of z.

Consider again the difference equation of the system (Eqn.1)


= =
=
q
0 k
p
1 k
k) a(k)y(n k) b(k)x(n y(n)


Taking the z-transform we get
1 2 q
1 2 3 p
1 2 3 p
1 2 q
b(0) b(1)z +b(2)z .... b(q)z B(z)
H(z)
1+a(1)z +a(2)z a(3)z .... a(p)z A(z)
A(z) 1+a(1)z +a(2)z a(3)z .... a(p)z
H(z) B(z) b(0) b(1)z +b(2)z .... b(q)z




+ + +
= =
+ +
= + +
= = + + +

Or
p q
k k
k 1 k 0
Y(Z) 1+ a(k)z X(z) b(k)z

= =
(
=
(



Or
q
k
k 0
p
k
k 1
b(k)z
Y(Z)
H(z)
X(z)
1+ a(k)z

=
= =

Or
1 2 q
1 2 3 p
b(0) b(1)z +b(2)z .... b(q)z B(z)
H(z)
1+a(1)z +a(2)z a(3)z .... a(p)z A(z)


+ + +
= =
+ +

Where
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Class notes of EKT353 by Prof. Dr. Farid Ghani

1 2 q
B(z) b(0) b(1)z +b(2)z .... b(q)z

= + + +

is the numerator polynomial in z
-1
with degree q and
1 2 3 p
A(z) 1+a(1)z +a(2)z a(3)z .... a(p)z

= + +

is the denominator polynomial in z-1 with degree p. Both the
polynomial have real coefficients and in general p > q.

The roots of the numerator polynomial B(z) are the zeros and
the roots of the denominator polynomial A(z) are the poles of
the digital filter . As for causal systems the degree of the
numerator polynomial is less than that of the denominator
polynomial. The order of the filter is p.



NOTE: A LTI system is stable if all its pole lie within the
unit circle in the z-plane

It should be noted that for a non recursive filter, all the
coefficients a
k
are zero so that A(z) = 1. In this case the transfer
function is given by
1 2 q
H(z) B(z) b(0) b(1)z +b(2)z .... b(q)z

= = + + +


This indicates that a non- recursive filter has no poles but
only zeros.

On the other hand for a recursive filter some or all the a
k
are
non zero and the transfer function is
1 2 q
1 2 3 p
b(0) b(1)z +b(2)z .... b(q)z B(z)
H(z)
1+a(1)z +a(2)z a(3)z .... a(p)z A(z)


+ + +
= =
+ +


Digital
Filter
X(z) Y(z)
9

Class notes of EKT353 by Prof. Dr. Farid Ghani

This indicates that a recursive filter has both zeros and
poles.

Cascade and Parallel Connection

If L transfer functions H
1
(z), H
2
(z) . . . , H
L
(z) are connected in
cascade (series), the resultant transfer function is the product of
the individual transfer functions. That is
H(z) = H
1
(z). H
2
(z) . . . H
L
(z).

This is shown in the following block diagram.

H
1
(z)
h
1
(n)
y
1
(n)
Y
1
(z)
H
2
(z)
h
2
(n)
y
2
(n)
Y
2
(z)
H
L
(z)
h
L
(n)
y(n)
Y(z)
. . . .
x(n)
X(z)

Systems /filters connected in cascade

If the systems are connected in parallel than the overall transfer
function is the sum of the individual transfer functions. That is

H(z) = H
1
(z). H
2
(z) . . . H
L
(z).

This is shown in the following block diagram.

H
1
(z)
h
1
(n)
y
1
(n)
Y
1
(z)
H
2
(z)
h
2
(n)
y
2
(n)
Y
2
(z)
H
L
(z)
h
L
(n)
y
L
(n)
Y
L
(z)
.

.

.

.
x(n)
X(z)
.

.

.

.
+
y(n)
Y(z)

10

Class notes of EKT353 by Prof. Dr. Farid Ghani


Frequency response of digital filter

Consider the transfer function H(z) of a digital filter .
Since z is a complex frequency it can be expressed in polar
coordinates

z = r e
je

where r = magnitude of z and is the phase angle.
Thus
H(z) = H(re
je)

Letting r =1,

H(z) = H(e
je
)

H(e
je
) is called the frequency response of the digital filter and
is defined as the response of the filter to a sinusoidal of varying
frequency.

The equation states that the Transfer function of a digital filter
evaluated for | Z|= 1 gives its frequency response. In other
words transfer function of the system evaluated at the unit circle
gives the frequency response of the digital filter .

Following are the properties of the frequency response of a real
sequence h(n).

1. H(e
je
) takes on values for all on a continuous basis.
2. H(e
je
) is periodic with period 2t.
3. The magnitude response |H(e
je
)| is an even function of e
and is symmetrical about t.
4. The phase response ZH(e
je
) is an odd function of e and
is antisymmetrical about t.




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Class notes of EKT353 by Prof. Dr. Farid Ghani

Realization of Digital FiltersFilters

We have seen previously that a digital filter with rational
transfer function and having input x(n) and output y(n) is
described by a linear constant coefficient difference equation
q p
k 0 k 1
y(n) b(k)x(n k) a(k)y(n k)
= =
=

(1)
and the corresponding z-transform is given by

q p
k k
k 0 k 1
Y(z) X(z) b(k)z Y(z) a(k)z

= =
=

(2)
If the difference equation has one or more terms a(k) that are
non-zero the difference equation and the corresponding digital
filter is called recursive or Infinite Impulse Response (IIR) Filter.
On the other hand if all the terms a(k) are zero then the
resulting difference equation and the corresponding digital filter
is termed as Finite Impulse Response (FIR) Filter.

From Equations (1) and (2) it is seen that the basic
computational (hardware) elements required to find the output
of the filter are:
1. ADDER
2. MULTIPLIER
3. UNIT DELAY
The filter is realized by connecting these elements as per the
difference equation describing the filter. This operation then
provides the block diagram for the implementation or
realization of the filter having the given difference equation
/transfer function.
The notations that are used for the three hardware elements
and the block diagram are as shown below:
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Class notes of EKT353 by Prof. Dr. Farid Ghani

+
x(n)
y(n)
x(n)+y(n)

Adder

ax(n)
x(n)
a

Multiplier

z
-1
x(n) x(n-1)

Unit Delay

A Filter is represented pictorially using a signal flow graph that
is a network of directed branches that are connected at nodes.
Each branch has an input and an output with the direction
indicated by an arrow head. The nodes in a block diagram
correspond to either adders or branch points. Adders
corresponds to nodes with more than one incoming branches
and branch points with more than one out going branches as
shown.

Node j Node k
X
j
(n)
X
k
(n)

Signal flow graph consisting of nodes and branches and node
variables. Node j represents and adder and node k represents a
branch point

The output of each branch is a linear transformation of the
branch input and the linear operator is indicated next to the
arrow. For linear time invariant systems/filters these linear
13

Class notes of EKT353 by Prof. Dr. Farid Ghani

operator consists of multipliers and delays as shown in the
following diagrams.

x(n) y(n)=ax(n)
Input
output
a
x(n) y(n)=x(n-1)
Input
output
Z
-1
Multiplier
Unit Delay


Finally there are two special types of nodes

1. Source nodes: These are nodes that have no incoming
branches and are used for sequences that are input to the
system/filter.
2. Sink nodes: These are nodes that have only entering
branches and are used to represent output sequences.

Structures for FIR Filters

A causal FIR Filter has a transfer function H(z) that is a
polynomial in z
-1
. That is

N
-n
n=0
H(z) = h(n)z

(1)
h(n) for n= 0,1, 2 . .n is the discrete impulse response of the
filter.
Notice that the transfer function of the FIR filter has no
poles and has only zeros.

For an input x(n) to the filter the output y(n) is given by
y(n) = h(n)*x(n)
Or y(n) = h(k)x(n-k) n= 0, 1, 2, . . .
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Class notes of EKT353 by Prof. Dr. Farid Ghani


For each value of n evaluating this sum requires (N+1)
multiplications and N additions. There are several different
realization/implementation block diagrams for this system and
these are considered below.

1. Direct Form
The most common way to implement a FIR Filter is in direct
form using a tapped delay line as shown in figure below. Let
h(n) be the discrete impulse response of the filter of length
N+1 and H(z) its z-transform.

h(n) = h
0
, h
1
, h
2
, h
3
, h
4
, . . . . , h
N

and H(z) = h
0
+h
1
z
-1
+ h
2
z
-2
+ h
3
z
-3
+ h
4
z
-4
+ . . . . +h
N
z
-N



N
-n
n=0
H(z) = h(n)z

h(0)
h(1) h(2) h(3) h(N-1) h(N)
z
-1
z
-1
z
-1
z
-1 x(n)
y(n)
. .
. .


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Class notes of EKT353 by Prof. Dr. Farid Ghani

.

.

.
.

.

.
Z
-1
Z
-1
Z
-1
Z
-1
h(0)
h(1)
h(2)
h(N-2)
h(N-1)
h(N)
x(n) y(n)

This structure requires (N+1) multiplications, N addition and
N delays.

2. Cascade Form
For a causal filter the transfer function may be
factored into product of 2
nd
order factors of the form

1+ b
1
(1)z
-1
+ b
1
(2)z
-2

The realization of a single second order factor is as shown
in figure.

z
-1
z
-1
b
1
(1)
x(n)
y(n)
b
1
(2)

Realization of a second order factor
16

Class notes of EKT353 by Prof. Dr. Farid Ghani


Thus if the transfer function of the filter is of order N, there will
be N/2 number of such factors. Let
N1=N/2
Then H(z) can be written as

1 2 1 2
1 1 2 2
1 2 1 2
3 3 N1 N1
H(z) A 1 b (1)z b (2)z 1 b (1)z b (2)z
1 b (1)z b (2)z .... 1 b (1)z b (2)z


( (
= + + + +

( ( + + + +


Or

N1
1 2
k k
k 1
H(z) A 1 b (1)z b (2)z

=
(
= + +

[

Written in this form H(z) may be implemented as a cascade of
second order FIR filters as shown in figure below. The
advantage of this form of realization is that any number of
second order sections can be cascaded to realize a filter of any
length.
b
1
(1)
z
-1
z
-1
z
-1
z
-1
x(n)
y(n)
. .
b
2
(1) b
Ns
(1)
b
1
(2) b
2
(2) b
Ns
(2)
z
-1
z
-1










17

Class notes of EKT353 by Prof. Dr. Farid Ghani


Structures for IIR (Recursive) Digital Filters
Let the z-transfer function of the IIR filter be H(z) and its
sampled impulse response sequence be h(n). Let x(n) be the
input sequence with z-transform X(z) and y(n) with z-
transform Y(z) be the output sequence. This arrangement is
shown in the following figure.

IIR
FILTER
h(n)
H(z)
Input
x(n)
X(z)
Output
y(n)
Y(z)


The input x(n) and the output y(n) of a causal IIR filter are
related with a rational transfer function H(z) where H(z) is given
by


q
k
k 0
p
k
k 0
b(k)z
y(z) B(z)
H(z)
x(z) A(z)
a(k)z

=
= = =

(1)
Where it is assumed that
1 2 -(q-1) -q
B(z) b(0) b(1)z b(2)z . . . +b(q-1)z +b(q)z

= + + +

1 2 -(p-1) -p
A(z) 1 a(1)z a(2)z . . . +a(q-1)z +b(q)z

= + + +
(2)


So that the inverse z-transforms of B(z) and A(z) are

b(n) = b(0), b(1), b(2), . . . . . , b(q-1), b(q)

a(n) = a(0), a(1), a(2), . . . . . , a(p-1), a(p) (2a)


The corresponding difference equation is
q p
k 0 k 0
y(n) b(k)x(n k) a(k)y(n k)
= =
=

(3)
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Class notes of EKT353 by Prof. Dr. Farid Ghani


We assume that a(1) =1

There are several structures that can be used to implement
an IIR filter described by the above transfer function (Eqn.1)
or the difference equation (Eqn. 3) and these are considered
below.

Direct Form Structures
There are two direct form IIR filter structures Direct Form I
and Direct Form II.

Direct Form I Structure

The Direct Form I structure is an implementation that results
when Eqn. 3 above is written in terms of two equations as
considered below

Equation 1 can be represented in block diagram form as

H(z) X(z)
Y(z)
B(z)/
A(z)
X(z)
Y(z)
B(z) X(z) Y(z) 1/A(z)
W(z)
OR
OR

Where
W(z) = B(z) X(z)

or w(n) = b(n) * x(n)

q
k 0
w(n) b(k)x(n k)
=
=

(4)
Now
Y(z) = w(z)/A(z)
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Class notes of EKT353 by Prof. Dr. Farid Ghani

or Y(z) A(z) = W(z)

Putting the value of A(z) we get

1 2 -(p-1) -p
Y(z) 1 a(1)z a(2)z . . . +a(q-1)z +b(q)z W(z)

(
+ + + =


Or
1 2
-(p-1) -p
Y(z) a(1)Y(z)z a(2)Y(z)z . . .
W(z)
+a(q-1)Y(z)z +b(q)Y(z)z

(
+ + +
=
(
(


Or
1 2
-(p-1) -p
a(1)Y(z)z a(2)Y(z)z . . .
Y(z) W(z)
+a(q-1)Y(z)z +b(q)Y(z)z

(
+ +
=
(
(



Taking inverse z-transform
a(1)y(n 1) a(2)Y(n 2) . . .
y(n) w(n)
+a(q-1)y(n (p 1))+b(p)y(n p)
+ +
(
=
(




Or
p
k=1
y(n) =w(n) - a(k)y(n k)

(5)
Thus from Equations 3, 4 and 5 we see that equation 3 can be
split in Equations 4 and 5. That is in equations


q
k 0
w(n) b(k)x(n k)
=
=

p
k=1
y(n) =w(n) - a(k)y(n k)

The first of the above equations (Eqn. 4) corresponds to a FIR
filter with input x(n) and output w(n) and discrete weighting
sequence b(n). The second of the above equations
corresponds to a filter that has all poles and no zeros with input
20

Class notes of EKT353 by Prof. Dr. Farid Ghani

w(n) and output y(n). Thus the above pair of Equations (Eqns. 4
and 5) represent a cascade of two filters/systems

| |
1
Y(z) B(z)X(z)
A(z)
( =

(6)

As shown in figure below.

z
-1
z
-1
z
-1
z
-1
b(0)
b(1)
b(2)
b(q-1)
b(q)
-a(1)
-a(2)
-a(p-1)
-a(p)
x(n) y(n)
.

.

.
.

.

.
.

.

.
.

.

.
z
-1
z
-1

Direct Form I realization of an IIR Filter

The computational requirements for Direct Form I structure are
1. Number of multiplications = ( p+q+1) per output sample
2. Number of additions = (p+q) per output sample
3. Number of delay units = (p+q)




21

Class notes of EKT353 by Prof. Dr. Farid Ghani

Direct Form II Structure

The Direct Form II Structure is obtained by reversing the
order of the cascade of B(z) and A(z) as shown in the
following figure.
1/A(z) X(z) Y(z)
W(z)
B(z)


With this implementation x(n) is first filtered with the all pole filter
1/A(z) and then with B(z). In this case the two difference equations
describing the filter are similar to Equations 4, and 5 and are given
below

p
k 1
w(n) x(n) a(k)w(n k)
=
=

(7)
And

q
k 0
y(n) b(k)w(n k)
=
=

(8)
Just as in the case of Direct Form I structure, the structure
represented by the difference Equations 7 and 8 is as shown in
the following figure.

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Class notes of EKT353 by Prof. Dr. Farid Ghani

z
-1
z
-1
z
-1
z
-1
z
-1
z
-1
b(0)
b(1)
b(2)
b(q-1)
b(q)
-a(1)
-a(2)
-a(p-1)
-a(p)
x(n) y(n)
.

.

.
.

.

.
.

.

.
.

.

.

Direct Form II Structure of the IIR Filter obtained by reversing
the Direct Form I structure

This structure may then be further simplified by noting that the
two sets of delays are delaying the same sequence and,
therefore, the two sets can be combined into a single set. This
results in to give the Direct Form II structure that is shown in the
following figure.
Note that the following figure corresponds to the case
when p=q. If pq then the corresponding values of b(0) or
a(0) would be zero.
23

Class notes of EKT353 by Prof. Dr. Farid Ghani

z
-1
z
-1
z
-1
b(0)
b(1)
b(2)
b(q-1)
b(q)
-a(1)
-a(1)
-a(p-1)
-a(p)
x(n) y(n)
.

.

.
.

.

.
.

.

.
.

.

.

Direct Form II Structure for an IIR Filter when p=q



Cascade Structure

The cascade structure is derived by factoring the numerator and
the denominator polynomials of H(z) . Thus

q
k
k max(p.q)
k 0 k
p k
k k 1
k
k 1
b(k)z
1 z
H(z) A
1 z
1 a(k)z

=
=
|
= =
o
+

This factorization corresponds to cascade of first order filters
each having one pole and one zero. However, in general the
24

Class notes of EKT353 by Prof. Dr. Farid Ghani

will be complex for real a(k) and b(k) and will appear as
complex conjugate pairs. These may be combined to form
second order factors with real coefficients. Thus the k
th
second
order factor of H(z) is given by


1 2
1k 2k
k
1 2
1k 2k
1 z z
H (z)
1 z z


+| +|
=
+o +o

As an example a sixth order IIR filter implemented as a cascade
of three second order filters in Direct Form II is as shown in
figure.

This may be noted that there is a considerable flexibility in
how a filter may be implemented in cascade form by
selecting different pairs of poles and zeros for example.

z
-1
z
-1
-
11
-
21

11

21
z
-1
z
-1
-
12
-
22

12

22
z
-1
z
-1
-
13
-
23

13

23
x(n) y(n)
Second order Section I Second order Section II Second order Section III

A sixth order IIR Filter implemented as a cascade of three
Direct Form II second order filters

Parallel Structure
An alternative to factoring H(z) is to expand H(z) through partial
fractions and express it as sum of second order factors and
results in the parallel structure for the IIR filter. The procedure
is discussed below.
Consider the transfer function H(z) of the IIR filter given as

25

Class notes of EKT353 by Prof. Dr. Farid Ghani


q
k
k max(p.q)
k 0 k
p k
k k 1
k
k 1
b(k)z
1 z
H(z) A
1 z
1 a(k)z

=
=
|
= =
o
+

If p>q and
i

k
(the roots of the denominator polynomial are
distinct) H(z) may be expanded by partial fraction expansion as p first
order factors

( )
p
k
1
k 1
k
C
H(z)
1 z

=
=
o

Where the coefficients C
k
and
k
are in general complex. This
expression corresponds to a sum of p first order transfer
functions and may be realized by connecting these in parallel.
However, the coefficients of h(n) and H(z) are real and
therefore, the poles of H(z)
k
, will have complex values in
general and will appear as conjugate pairs. These complex
roots in partial fraction may be combined to form second order
systems with real coefficients. So that

1
Ns
0k 1k
1 2
k 1
1k 2k
z
H(z)
1 z z

=
+
=
+o +o

Figure shows the implementation of six order IIR filter as a
parallel combination of three second order Direct FORM II
filters.
26

Class notes of EKT353 by Prof. Dr. Farid Ghani

z
-1
z
-1
-
11
-
21

11
z
-1
z
-1
-
12
-
22
z
-1
z
-1
-
13
-
23
x(n) y(n)
Second order Section I
Second order Section II
Second order Section III

01

02

12

03

13

A six order IIR filter implemented as a parallel connection of
three second order direct form II structures

If p is less than or equal to q, the partial fraction expansion will
also contain a term of the form
1 2 -(q-p)
0 1 2 (q-p)
c c z c z . . . . +c z

+ + +

Which is an FIR filter that is placed in parallel with the other
terms in the expression of H(z).