DIGITAL SIGNAL PROCESSING

(DSP-7CC10)

Unit-V DIGITAL FILTERS

Faculty: V.RAJENDRA CHARY(VRC),

Assistant Professor,ECE
Class: III B.Tech,II Semester, ECE
 SYLLABUS
UNIT-
1. INTRODUCTION
2. DISCRETE FOURIER TRANSFORM
3. FAST FOURIER TRANSFORMS
…..mid 1
4. REALIZATION OF DIGITAL FILTERS
5. DIGITAL FILTERS
6. MULTIRATE DIGITAL SIGNAL PROCESSING
…..mid 2
V.RAJENDRA CHARY 2
 UNIT-IV REALIZATION OF DIGITAL
FILTERS
• Review of Z-transforms, Applications of Z–transforms, Block
diagram representation of linear constant-coefficient difference
equations, Basic structures of IIR systems, Transposed forms,
Basic structures of FIR systems, System function.
• Applications: Design of digital system function to meet the given
specifications.

V.RAJENDRA CHARY 3
 UNIT-V DIGITAL FILTERS
• Analog Filter Approximations – Butterworth and Chebyshev
Approximations.
• IIR digital filters: Design of IIR Digital filters from analog filters-
Impulse Invariance, Step invariance and Bilinear Transformation
methods, Design Examples, Analog-Digital transformations.

V.RAJENDRA CHARY 4
 UNIT-V DIGITAL FILTERS(contd.)
• FIR digital filters: Characteristics of FIR Digital Filters, frequency
response, Design of FIR Digital Filters using Fourier series
method, Windowing Techniques-Rectangular, Triangular,
Hamming, Hanning and Bartlett’s Windows, Steps in Kaiser
windowing method, Frequency Sampling technique, Comparison
of IIR and FIR filters.
• Applications: Design of IIR/FIR digital filter conforming to
given specifications.

V.RAJENDRA CHARY 5
 UNIT-VI MULTIRATE DIGITAL SIGNAL
PROCESSING
• Decimation, interpolation, sampling rate conversion. Introduction
to DSP Processors.

• Applications of Multirate Digital Signal processing: Design of

digital filter banks and quadrature mirror filters etc.

V.RAJENDRA CHARY 6
REFERENCE TEXTBOOKS(R1)

V.RAJENDRA CHARY 7
REFERENCE TEXTBOOKS(R2)

V.RAJENDRA CHARY 8
REFERENCE TEXTBOOKS(R3)

V.RAJENDRA CHARY 9
 Outline Of The Unit
• Analog Filter Approximations – Butterworth and Chebyshev
Approximations.
• IIR digital filters: Design of IIR Digital filters from analog filters-
Impulse Invariance, Step invariance and Bilinear Transformation
methods, Design Examples
• Analog-Digital transformations.

V.RAJENDRA CHARY 10
 Outline Of The Unit(contd.)
• FIR digital filters: Characteristics of FIR Digital Filters,frequency
response
• Design of FIR Digital Filters using Fourier series method,
Windowing Techniques-Rectangular, Triangular, Hamming,
Hanning and Bartlett’s Windows, Steps in Kaiser windowing
method
• Frequency Sampling technique
• Comparison of IIR and FIR filters.
• Applications
V.RAJENDRA CHARY 11
V.RAJENDRA CHARY 12
 Introduction
• A filter is one which rejects unwanted frequencies from the input
signal and allow the desired frequencies.
• The range of frequencies of signals that are passed through the
filter is called pass band and those frequencies that are blocked is
called stop band.
• The filters are of different types viz.. Lowpass filter,highpass
filter,bandpass filter and bandstop filter etc…

V.RAJENDRA CHARY 13
Lowpass filter

V.RAJENDRA CHARY 14
Highpass filter

V.RAJENDRA CHARY 15
 Bandpass filter
• Magnitude response of a)Ideal and b)Practical Bandpass filter:

V.RAJENDRA CHARY 16
 Bandstop(or Bandreject) filter
• Magnitude response of a)Ideal and b)Practical Bandstop filter:

V.RAJENDRA CHARY 17
 Design of Digital filter from Analog filter
• The most common techniques used for designing IIR digital filters
known as indirect method, involves first designing an analog
prototype filter and then transforming the prototype to a digital
filter.
• For the given specifications of a digital filter, the derivation of the
digital filter transfer function requires three steps
1.Map the desired digital filter transfer function into equivalent
analog filter.
2.Derive the analog transfer function for the analog prototype.
V.RAJENDRA CHARY 18
 Design of Digital filter from Analog
filter(contd.)
3.Transform the transfer function from the analog prototype into an
equivalent digital transfer function.

V.RAJENDRA CHARY 19
Digital filter v/s Analog filter

V.RAJENDRA CHARY 20
Digital filter v/s Analog filter(contd.)

V.RAJENDRA CHARY 21
 Advantages of Digital filter
• The values of resisitors, capacitors and inductors used in analog
filters change with temperature. Since the digital filters do not have
these components, they have high thermal stability.
• In digital filters,the precision of the filter depends on the size of
registors used to store the filter coefficients. Hence by increasing
the register bit length in hardware,the performance characteristics
of the filter like accuracy,dynamic range,stability,tolerance and
frequency response can be enhanced.

V.RAJENDRA CHARY 22
 Advantages of Digital filter(contd.)
• The digital filters are programmable. Hence the filter coefficients
can be changed any time to implement adaptive features.
• A single filter can be used to process multiple signals by using the
techniques of multiplexing.
• Digital filters can be operated over a wide range of frequencies.
• There are no problems of input or output impedance matching with
digital filters.

V.RAJENDRA CHARY 23
 Advantages of Digital filter(contd.)
• The digital filters performance is not influenced by component
ageing, temperature and power supply variations.
• A digital filter is highly immune to noise.

V.RAJENDRA CHARY 24
 Disadvantages of Digital filter
• The bandwidth of the discrete signal is limited by the sampling
frequency. The bandwidth of real discrete signal is half the
sampling frequency.
• The quantization error arises due to finite word length in the
representation of signals and parameters.

V.RAJENDRA CHARY 25
V.RAJENDRA CHARY 26
 Analog Lowpass filter design
• The most general form of analog filter transfer function is

Where H(s) is the Laplace transform of the impulse response h(t)

V.RAJENDRA CHARY 27
 Analog Lowpass filter design(contd.)
• Here N ≥ M must be satisfied.
• For a stable analog filter,the poles of H(s) lie in the left half of the
s-plane.
• The various types of analog filter design are:
-Butterworth filter
-Chebyshev filter

V.RAJENDRA CHARY 28
 Analog Lowpass Butterworth filter
• The magnitude function of the Butterworth lowpass filter is given
by:

Where N is the order of the filter and Ωc is the cutoff frequency.

V.RAJENDRA CHARY 29
 Analog Lowpass Butterworth filter(contd.)

• The magnitude response of Butterworth Low pass filter for various

values of N is as follows:

V.RAJENDRA CHARY 30
 Analog Lowpass Butterworth filter(contd.)
• As shown in figure,the function is monotonicaly decreasing,where
the maximum response is unity at Ω=0.
• The ideal response is shown by dashed line.It can be seen that the
magnitude response approaches the ideal lowpass characteristics as
the order N increases.
• For values Ω < Ωc, |H(jΩ)| ≈1.For values Ω > Ωc, |H(jΩ)|
decreases rapidly.
• At Ω = Ωc, the curves pass through 0.707 which corresponds to
-3dB point.
V.RAJENDRA CHARY 31
 Analog Lowpass Butterworth filter(contd.)

• The magnitude square function of a normalized Butterworth filter

is:

• Now let us derive the transfer function of a stable filter. For this
purpose, substitute Ω =s/j

V.RAJENDRA CHARY 32
 Analog Lowpass Butterworth filter(contd.)

• In the above equation, when s/Ωc is replaced by sn(i.e letting Ωc =1

rad/sec),the transfer function is called normalized transfer function

V.RAJENDRA CHARY 33
 Analog Lowpass Butterworth filter(contd.)

• The above transfer function will have 2N poles which are given by
the roots of the denominator polynomial.

V.RAJENDRA CHARY 34
 Poles of Butterworth lowpass filter
• Let us equate the denominator polynomial of the normalized
transfer function(previous slide) to zero and solve the 2N poles of
Butterworth lowpass filter

V.RAJENDRA CHARY 35
Poles of Butterworth lowpass filter(contd.)

V.RAJENDRA CHARY 36
Poles of Butterworth lowpass filter(contd.)

V.RAJENDRA CHARY 37
Poles of Butterworth lowpass filter(contd.)

V.RAJENDRA CHARY 38
Poles of Butterworth lowpass filter(contd.)

V.RAJENDRA CHARY 39
Poles of Butterworth lowpass filter(contd.)

V.RAJENDRA CHARY 40
Poles of Butterworth lowpass filter(contd.)

V.RAJENDRA CHARY 41
Magnitude response of Butterworth lowpass
filter approximation

V.RAJENDRA CHARY 42
 Order of Butterworth lowpass filter(contd.)

• The order of Butterworth lowpass filter is given by

V.RAJENDRA CHARY 43
 Order of Butterworth lowpass filter(contd.)

• For simplicity, we define the parameters A and k as follows:

Where k=transition ratio

V.RAJENDRA CHARY 44
 Order of Butterworth lowpass filter(contd.)

• Finally the order equation for the lowpass Butterworth analog filter
is given by:

V.RAJENDRA CHARY 45
 Cutoff frequency of Analog Lowpass
Butterworth filter
• The cutoff frequency of analog lowpass Butterworth filter is given
by:

V.RAJENDRA CHARY 46
Properties of Butterworth filter

V.RAJENDRA CHARY 47
 Steps to design Analog Butterworth LPF
1)From the given specifications find the order of the filter N.
2)Round off it to the next higher integer.
3)Find the transfer function H(s) for Ωc=1 rad/sec for the value of N.
4)Calculate the value of cutoff frequency Ωc.
5)Find the transfer function Ha(s) for the above value of Ωc by
substituting s/Ωc in H(s).

V.RAJENDRA CHARY 48
 Analog Lowpass Chebyshev filters
• There are two types of Chebyshev filters viz….Chebyshev type-I
filter and Chebyshev type-II filter.
• Chebyshev type-I filters are all-pole filters that exhibit equiripple
behaviour in the pass band and a monotonic characteristics in the
stop band.
• Chebyshev type-II filters contain both poles and zeros.They exhibit
a monotonic behaviour in the pass band and an equiripple
behaviour in the stop band.

V.RAJENDRA CHARY 49
Analog Lowpass Chebyshev filters(contd.)

V.RAJENDRA CHARY 50
Analog Lowpass Chebyshev filters(contd.)
• The magnitude response of Nth order type-I lowpass Chebyshev
filter is given by:

• Where ϵ is the parameter related to the ripple in the passband

V.RAJENDRA CHARY 51
Analog Lowpass Chebyshev filters(contd.)

• CN(x) is the Nth order Chebyshev polynomial defined as:

V.RAJENDRA CHARY 52
Properties of Chebyshev polynomial

V.RAJENDRA CHARY 53
 Order of Lowpass Chebyshev filter
• The order of Lowpass Chebyshev filter(type-I&II) is given by:

V.RAJENDRA CHARY 54
 Pole locations of Chebyshev filter
• The poles of Chebyshev filter is given by

V.RAJENDRA CHARY 55
Pole locations of Chebyshev filter(contd.)
• The poles of the Chebyshev transfer function are located on an
ellipse in the s-plane as shown in figure.

V.RAJENDRA CHARY 56
Pole locations of Chebyshev filter(contd.)
• The equation of the ellipse is given by:

Where a and b are minor and major axes of the ellipse respectively

V.RAJENDRA CHARY 57
 Steps to design Analog Lowpass Chebyshev
LPF
1)From the given specifications find the order of filter N.
2)Round off it to the next higher integer.
3)Using the following formulas find the values of a and b,which are
minor and major axis of the ellipse respectively.

V.RAJENDRA CHARY 58
 Steps to design Analog Lowpass Chebyshev
LPF(contd.)
4)Calculate the poles of Chebyshev filter which lie on an ellipse by
using the formula

5)Find the denominator polynomial of the transfer function using the

above poles.

V.RAJENDRA CHARY 59
 Steps to design Analog Lowpass Chebyshev
LPF(contd.)
6)The numerator of the transfer function depends on the value of N
a)For N=odd,substitute s=0 in the denominator polynomial and find
the value.This value is equal to the numerator of the transfer
function.
b)For N=even,substitute s=0 in the denominator polynomial and
divide the result by .This value is equal to the numerator of
the transfer function.

V.RAJENDRA CHARY 60
Properties of Chebyshev filter

V.RAJENDRA CHARY 61
 Chebyshev Type-II filter
• Chebyshev type-II filter has both poles and zeros.The magnitude
square response is given by:

• Where CN(x) is Nth order Chebyshev polynomial

V.RAJENDRA CHARY 62
 Chebyshev Type-II filter(contd.)
• The zeros are located on the imaginary axis at the points,

• The poles are located at the points (xk,yk),where

V.RAJENDRA CHARY 63
Chebyshev Type-II filter(contd.)

V.RAJENDRA CHARY 64
Chebyshev Type-II filter(contd.)

V.RAJENDRA CHARY 65
V.RAJENDRA CHARY 66
 Problem
• Given the specifications αp=1dB, αs=30dB, Ωp=200 rad/sec,
Ωs=600 rad/sec. Determine the order of Butterworth filter.
(R1 eg5.1)

V.RAJENDRA CHARY 67
 Problem
• Design an analog Butterworth filter that has a 2dB passband
attenuation at a frequency of 20 rad/sec and atleast 10dB stopband
attenuation at 30 rad/sec. (R1 eg5.4)

V.RAJENDRA CHARY 68
 Problem
• For the given specifications, design an analog Butterworth filter
0.9 ≤|H(jΩ)| ≤1 for 0 ≤ Ω ≤ 0.2 Π ; |H(jΩ)| ≤0.2 for 0.4 Π ≤ Ω ≤ Π
(R1 eg5.5)

V.RAJENDRA CHARY 69
 Problem
• Given the specifications αp=3dB, αs=16dB, fp=1KHz, fs=2KHz.
Determine the order of filter using chebyshev approximation. Find
H(s) (R1 eg5.6)

V.RAJENDRA CHARY 70
 Problem
• Obtain an analog Chebyshev filter transfer function that satisfies
the constraints (R1 eg5.7)

V.RAJENDRA CHARY 71
 Problem
• Determine the order and the poles of a type-I lowpass Chebyshev
filter that has a 1 dB ripple in the passband and passband frequency
Ωp=1000Π ,stopband frequency Ωs =2000Π and an attenuation of
40dB or more (R1 eg5.8)

V.RAJENDRA CHARY 72
 Comparision b/w Butterworth filter and
Chebyshev filter
• The magnitude response of Butterworth filter decreases
monotonically as the frequency Ω increases from 0 to ∞,whereas
the magnitude response of the Chebyshev filter exhibits ripples in
the pass band or stop band according to the type.
• The transition band is more in Butterworth filter when compared to
Chebyshev filter.
• The poles of the Butterworth filter lie on a circle,whereas the poles
of Chebyshev filter lie on an ellipse.

V.RAJENDRA CHARY 73
 Comparision b/w Butterworth filter and
Chebyshev filter(contd.)
• For the same specifications, the number of poles in Butterworth are
more when compared to the Chebyshev filter i.e the order if the
chebyshev filter is less than that of Butterworth. This is a great
advantage because less number of discrete components will be
necessary to construct the filter.

V.RAJENDRA CHARY 74
 Frequency transformation in Analog Domain

• Lowpass to lowpass
• Lowpass to highpass
• Lowpass to bandpass
• Lowpass to bandstop

V.RAJENDRA CHARY 75
 Lowpass to Lowpass filter
• Given a normalized lowpass filter, it is desirable to have a lowpass
filter with a different cutoff frequency Ωc (or passband frequency
Ωp).This can be accomplished by transformation given by:

V.RAJENDRA CHARY 76
Lowpass to Lowpass filter(contd.)

V.RAJENDRA CHARY 77
 Lowpass to Highpass filter
• Given a normalized lowpass filter,if it is desirable to have a
highpass filter with cutoff frequency Ωc.Then the transformation is

V.RAJENDRA CHARY 78
Lowpass to Highpass filter(contd.)

V.RAJENDRA CHARY 79
 Lowpass to Bandpass filter
• The transformation for converting a normalized lowpass filter to a
bandpass filter with cutoff frequencies Ωl, Ωu can be accomplished
by

V.RAJENDRA CHARY 80
Lowpass to Bandpass filter(contd.)

V.RAJENDRA CHARY 81
 Lowpass to Bandstop filter
• The transformation to convert a normalized lowpass filter to a
bandstop filter is

V.RAJENDRA CHARY 82
Lowpass to Bandstop filter(contd.)

V.RAJENDRA CHARY 83
 Frequency Transformation in digital domain

• Lowpass to Lowpass
• Lowpass to highpass
• Lowpass toBandpass
• Lowpass to Bandstop

V.RAJENDRA CHARY 84
Lowpass to Lowpass

V.RAJENDRA CHARY 85
Lowpass to Highpass

V.RAJENDRA CHARY 86
Lowpass to Bandpass

V.RAJENDRA CHARY 87
Lowpass to Bandstop

V.RAJENDRA CHARY 88
V.RAJENDRA CHARY 89
 Design of IIR digital filters from analog
filters
• There are several methods that can be used to design digital filters
from analog filter viz..
1)Impulse Invariance transformation
2)Bilinear transformation
3) Step Invariance transformation

V.RAJENDRA CHARY 90
 Impulse Invariance method
• In Impulse Invariance method the IIR filter is designed such that
the unit impulse response h(n) of digital filter is the sampled
version of the impulse response of analog filter.
• Let Ha(s) be the system function of an analog filter. This can be
expressed in partial fraction as:

V.RAJENDRA CHARY 91
 Impulse Invariance method(contd.)
• Where {pk} are the poles of analog filter and {ck}are coefficients in
the partial fraction expansion.
• The inverse Laplace transform of the above system
function(previous slide) is given by

V.RAJENDRA CHARY 92
 Impulse Invariance method(contd.)
• If we sample ha(t) periodically at t=nT,we have

V.RAJENDRA CHARY 93
 Impulse Invariance method(contd.)
• Substituting the expression of h(n) in H(z),we get

V.RAJENDRA CHARY 94
 Impulse Invariance method(contd.)
• For high sampling rates(for small T),the digital filter gain is high.
Therefore we use

• Due to the presence of aliasing,the impulse invariance method is

appropriate for the design of lowpass and bandpass filters only.
• This method is unsuccessful for implementing digital filters such
as high pass filter.
V.RAJENDRA CHARY 95
 ImImpulse Invariance method(contd.)n
• The Z-Transform of an IIR filter is
 
H (Z )  
n0
h[n ] Z n
 H (Z ) Z  e sT
  h [ n ] e  sTn
n0

Let us consider the mapping of points from S-plane to Z-Plane implied by

the relation sT
Z  e
by substituting s    j  and Z  re j

 re j
 e (   j ) T  r  e  T and   T
V.RAJENDRA CHARY 96
 Impulse Invariance method(contd.)

• Consider any pole

On jΩ-axis
  0  r  e 0 .T  1  UNIT CIRCLE
Left-half S-plane
  0  r  e T  1  INSIDE UNIT CIRCLE
Right-half S-plane
  0  r  e T  1  OUTSIDE UNIT CIRCLE

V.RAJENDRA CHARY 97
 Impulse Invariance method(contd.)

UNIT CIRCLE Im jΩT

Z-plane
S-plane

ΩT ΩT

Re
σT

Mapping of jΩ-axis, Stable and Unstable poles from S-plane to Z-plane

V.RAJENDRA CHARY 98
 Impulse Invariance Transformation(contd.)

• Therefore the impulse invariance method maps

1. Poles from the jΩ-axis of S-plane to Unit circle of Z-plane.

2. Poles from Left-half (Negative real part) of S-plane to inside unit

circle of Z-plane.
3. Poles from Right-half (Positive real part) S-plane to outside unit
circle of Z-plane.
• Disadvantage:
It is not one-to-one mapping it is many-to-one mapping.

V.RAJENDRA CHARY 99
 Impulse Invariance Transformation(contd.)

jΩ

UNIT CIRCLE Im
Z-plane 2π/T
S-plane
π/T

ΩT Ω

Re σ
2π/T

-π/T

Impulse Invariance pole many-to-one mapping

V.RAJENDRA CHARY 100
 Impulse Invariance Transformation(contd.)
• Impulse invariance pole mapping
s 1    j   z 1  e (   j ) T  e  T e j T

  j   T
s 2    j   
2
2
T z2  e T
 e T e j T
e j 2

s 2    j   2
T  z 2  e T e j T
( because e j2
 1)

•Here z1=z2, there are infinite number of S-plane poles that map to the
same location in the Z-plane.
– They have same real part, but imaginary parts differ by 2π/T.

The S-plane poles having imaginary parts between –π/T to π/T

causes aliasing, when sampling analog signals.
101
V.RAJENDRA CHARY
Steps to design a digital filter using Impulse
Invariance method
• For the given specifications, find Ha(s),the transfer function of an
analog filter.
• Select the sampling rate of the digital filter,T seconds per sample.
• Express the analog filter transfer function as the sum of single-pole
filters
N
Ck
Ha (s)  
k 1 s  pk

V.RAJENDRA CHARY 102

Steps to design a digital filter using Impulse
Invariance method(contd.)
• Compute the Z-transform of the digital filter using the formula

N
 Ck 
H ( z)    pk T 1 
k 1  1  e z 

• For high sampling rates,

N
 TCk 
H ( z)    pk T 1 
k 1  1  e z 

V.RAJENDRA CHARY 103

Relation b/w analog and digital frequencies in
Impulse Invariance method
• The relationship between analog frequency, Ω and digital
frequency, ω is :

V.RAJENDRA CHARY 104

Useful Impulse Invariance Transformation

V.RAJENDRA CHARY 105

V.RAJENDRA CHARY 106
 Bilinear Transformation
• The Bilinear transformation is a conformal mapping that
transforms the jΩ axis into the unit circle in the z-plane only once,
thus avoiding aliasing of frequency components.
• In this mapping all points in the left half of s-plane are mapped
inside the unit circle in the z-plane and all points in the right half of
s-plane are mapped outside the unit circle in the z-plane.

V.RAJENDRA CHARY 107

 Bilinear Transformation method-
Derivation
Let us consider an analog linear filter of system function
b Y (s) b
H (s)   (1 )    (2)
s  a X (s) s  a
So
sY ( s )  aY ( s )  bX ( s )  ( 3 )
It’s inverse Laplace transform (differential equation) is equal to

dy ( t )
 ay ( t )  bx ( t )  ( 4 )
dt
y(t) can be approximated by the trapezoidal formula
t
y (t )  t0
y ' ( ) d   y ( t 0 )  ( 5 )
Where y’(t) denotes the derivative of y(t).

V.RAJENDRA CHARY 108

 Bilinear Transformation method-
Derivation
Instead of substituting a finite difference for the derivative, suppose
that we integrate the derivative and approximate the integral by the
trapezoidal formula:
x 2
1
 f ( x )dx  x2  x 1  f ( x 2 )  f ( x 1 ) 
2
Thus: x1
nT
t
y (t )  
t0
y '(  ) d   y ( t 0 )  
nT T
y '( ) d   y ( n T  T )

The approximation of the previous integral by trapezoidal formula is

T
y (nT )  y '( n T )  y '( n T  T )   y ( n T  T )
2
V.RAJENDRA CHARY 109
 Bilinear Transformation method
Derivation
The approximation of above integral by the trapezoidal formula at t=nT and t0=nT-T yields

y ( nT )  T
2
y '

( nT )  y ' ( nT  T )  y ( nT  T )  ( 6 )
From the differential equ(4) we obtain

y ' ( nT )   ay ( nT )  bx ( nT )  ( 7 )
By Substituting equ.7 in equ.6, we get

y ( nT )  T
2  ay ( nT )  bx ( nT )  ay ( nT  T )  bx ( nT  T )   y ( nT  T )
1  aT
2
y ( nT )  1  aT
2
y ( nT T )  bT
2
x ( nT )  x ( nT  T )   ( 8 )
The Z-transform of this differential equation is

1  aT
2
Y ( z )  1  aT
2
z  1Y ( z )  bT
2
1  
z 1 X ( z )

V.RAJENDRA CHARY 110

 Bilinear Transformation method
Derivation
Transfer (System) function H(z) of the equivalent digital filter is

H (z) 
Y (z)

bT
2 1  z 1 
X (z) 1  aT
2   1  aT2 z  1


bT
1  z  1

bT
2
 
2
1  z 1
  1 
aT
2 z 1
 1  z 1
1  z 1
aT
2

b
H (z)   (9 )
2
T 1  z 1
1  z 1
 a
V.RAJENDRA CHARY 111
 Bilinear Transformation method
Derivation
Transfer function of the analog filter: Transfer function of the digital filter:

b b
H (z) 
H (s) 
s  a 2
T 
1 z
1 z
1
1  a
By Comparing H(s) and H(z), it can be seen that H(z) can be obtained from H(s) by using the
mapping relation

S  2
T
1 z 1
1  z 1
 (10 )
This relationship between s and Z is known as Bilinear transformation.

V.RAJENDRA CHARY 112

 Steps to design digital filter using Bilinear
Transformation
1)From the given specifications, find prewarping analog frequencies
using the formula

2)Using the analog frequencies find H(s) of the analog filter.

V.RAJENDRA CHARY 113

 Steps to design digital filter using Bilinear
Transformation(contd.)
3)Select the sampling rate of the digital filter, call it T seconds per
sample.
4)Substitute into the transfer function found in
step2.

V.RAJENDRA CHARY 114

 Warping effect
• Let Ω and ω represent the frequency variables in the analog filter
and the derived digital filter respectively.
• We have,

V.RAJENDRA CHARY 115

 Warping effect(contd.)
• For small values of ω ,

V.RAJENDRA CHARY 116

 Warping effect(contd.)
• For low frequencies the relationship between Ω and ω are linear.As
a result, the digital filter will have the same amplitude response as
the analog filter.
• For high frequencies, however the relationship between Ω and ω
becomes non-linear(shown in figure in next slide) and the
distortion is introduced in the frequency scale of the digital filter to
that of the analog filter. This is known as warping effect.

V.RAJENDRA CHARY 117

 Warping effect(contd.)
• The relation between Ω and ω in Bilinear transformation is shown
in figure below:

V.RAJENDRA CHARY 118

 Warping effect(contd.)
• The warping effect on magnitude response is as shown in figure
below:

V.RAJENDRA CHARY 119

 Warping effect(contd.)
• The warping effect on phase response is as shown in figure below:

V.RAJENDRA CHARY 120

 Prewarping
• The warping effect can be eliminated by prewarping the analog
filter. This can be done by finding prewarping analog frequencies
using the formula

V.RAJENDRA CHARY 121

 Prewarping (contd.)
• Therefore, we have

V.RAJENDRA CHARY 122

 Step Invariance Transformation

hs[t] hs[n]

3 4 5 6 7 n
t
-8 -7 -6 -5 -4 -3 -2 -1 1 2 8 9 10

Step response of the discrete-time system be the same as the corresponding step response of
the reference analog filter at sampling points.

 z 1
H (z)    Z .T I . L .T  H (s)
s

 z 
V.RAJENDRA CHARY 123
 Step Invariance Transformation
As shown in the figure it is required that hs [ n ]  hs (t ) t  nT
H (s)
We know that hs (t )  h (t ) * u (t )  H s ( s ) 
s
or h s ( t )  I . L .T  H (s)
s

H s ( z )  Z .T of h s ( t ) after sampling H s ( z )  Z .T  I . L .T  H (s)
s
  I
We also know that hs (n )  u[n ] * h (n )
 z 
then H s(z)    H ( z )  II
 z  1 
From I and II  z 
  H ( z )  Z .T  I . L .T  H (s)
s

 z 1

 z 1   H (s) 
So we can write H (z)    Z .T  I . L .T   
 z    s 
V.RAJENDRA CHARY 124
V.RAJENDRA CHARY 125
 Problem
• Determine H(z) for the following analog transfer function using
Impulse Invariance method and Bilinear transformation. Assume
T=1 sec (R1 eg5.11,5.16)

V.RAJENDRA CHARY 126

 Problem
• Design a third order Butterworth digital filter using impulse
invariance method. Assume sampling period T=1sec (R1 eg5.13)

V.RAJENDRA CHARY 127

 Problem
• For the analog transfer function

Determine H(z) using Impulse Invariance transformation and

Bilinear transformation if a)T=1sec b)T=0.1sec (R3 eg7.1, R3
eg7.4)

V.RAJENDRA CHARY 128

 Problem
• Apply impulse invariance method and find H(z) for

H(s)= (R1 eg5.14)

V.RAJENDRA CHARY 129

 Problem
• Convert the analog filter with system transfer function

into a digital IIR filter by means of Impulse Invariance method

(R3 eg7.2)

V.RAJENDRA CHARY 130

 Problem
• Design a Butterworth digital IIR filter using impulse invariance
method for the specifications
0.8 ≤|H(ejω )| ≤1 for 0 ≤ ω ≤ 0.2Π
|H(ejω)| ≤0.2 for 0.6Π ≤ ω ≤ Π
with T=1 sec (R1 eg.5.32)

V.RAJENDRA CHARY 131

 Problem
• Design a Chebyshev digital IIR filter using a)bilinear
transformation b)impulse invariance method , for the specifications
0.8 ≤|H(ejω )| ≤1 for 0 ≤ ω ≤ 0.2Π
|H(ejω)| ≤0.2 for 0.6Π ≤ ω ≤ Π
with T=1 sec. (R1 eg.5.34)

V.RAJENDRA CHARY 132

 Problem
• Design a Butterworth digital IIR filter satisfying the constraints
0.707 ≤|H(ejω )| ≤1 for 0 ≤ ω ≤ Π/2
|H(ejω)| ≤0.2 for 3Π/4 ≤ ω ≤ Π
with T=1 sec using a)bilinear transformation and realize the
structure using direct form-II,b)impulse invariance method and
realize the structure using parallel form (R1 eg.5.30)

V.RAJENDRA CHARY 133

 Problem
• Determine H(z) that results when the bilinear transformation is
applied to

Assume T=1sec (R1 eg5.18)

V.RAJENDRA CHARY 134

 Problem
• Convert the analog filter with system function H(s) into digital
filter using Bilinear transformation (R3 eg7.7)

V.RAJENDRA CHARY 135

 Solved Problem
2
For an analog system whose transfer function is given by H (s) 
( s  1 )( s  2 )

.Determine the digital transfer

function using step invariance method, if the sampling frequency is 10Hz.
Solution: H (s)  2
( s  1 )( s  2 )  H (s)
 2
s s ( s  1 )( s  2 )
Using partial fractions
 H s (s)  H (s)
s  A
s  B
s 1  C
s2
We can get
A  1, B   2 , C  1
 H s (s)  1
s  2
s 1  1
s2  hs (t )  u [t ]  2 e  t  e 2t
It’s sampled version
h s ( n )  u [ n ]  2 e  nT  e  2 nT
By using
Z .T u [ n ]  
z
z 1
 
Z .T e  t 
z
z  e T

Z .T e  2 t   z
z  e  2T
V.RAJENDRA CHARY 136
 Solved Problem
Z-transform of the sampled version with T=0.1 sec is

z 2z z z 2z z
H s(z)      
z 1 z  e T z  e 2T z 1 z  0 . 9048 z  0 . 8187

z ( z 2  1 . 7235 z  0 . 7408  2 z 2  3 . 637 z  1 . 637 )

H s(z) 
( z  1 )( z 2  1 . 7235 z  0 . 7408 )

10  3 z ( 9 . 055 z  8 . 194 )
H s(z) 
( z  1 )( z 2  1 . 7235 z  0 . 7408 )

 z 1 9 . 055 x 10  3 ( z  1  0 . 9048 z  2 )

H (z)   H s(z) 
 z  (1  1 . 7235 z  1  0 . 7408 z  2 )

V.RAJENDRA CHARY 137

V.RAJENDRA CHARY 138
 Advantages of FIR digital filters
FIR filters are preferred over their IIR counterparts. The following
are the main advantages of FIR filter over IIR filter:
• FIR filters are always stable
• FIR filters with exactly linear phase can easily be designed.
• FIR filters can be realized in both recursive and non-recursive
structures.
• FIR filters are free of limit cycle oscillations, when implemented
on a finite word length digital system
V.RAJENDRA CHARY 139
Advantages of FIR digital filters(contd.)
• Excellent design methods are available for various kinds of FIR
filters.

V.RAJENDRA CHARY 140

 Disadvantages of FIR filter
• The implementation of narrow transition band FIR filters are very
costly, as it requires considerably more arithmetic operations and
hardware components such as multipliers, adders and delay
elements.
• Memory requirement and execution time are very high.

V.RAJENDRA CHARY 141

 Frequency Response
• The frequency response of the filter is the Fourier transform of its
impulse response.
• If h(n) is the impulse response of the system, then the frequency
response of the system is denoted by H(ejω) or H(ω).

V.RAJENDRA CHARY 142

Frequency Response Characteristics of Linear
phase FIR filters

V.RAJENDRA CHARY 143

Summary of magnitude function for Linear
phase FIR filters

V.RAJENDRA CHARY 144

 Design of FIR Digital filters
• The various methods for designing FIR filters are:
1)Fourier series method
2)Window method
3)Frequency sampling method

V.RAJENDRA CHARY 145

 Design of FIR Digital filters using Fourier
Series method
The following two concepts leads to the design of FIR filters by
Fourier Series method:
• The frequency response of a digital filter is periodic with period
2π.
• Any periodic function can be expressed as a linear combination of
complex exponentials.

V.RAJENDRA CHARY 146

 Design of FIR Digital filters using Fourier
Series method(contd.)
• The desired frequency response Hd(ejω) of an FIR digital filter can
be represented by the Fourier series as

Where hd(n) is the desired impulse response of the filter

V.RAJENDRA CHARY 147

 Design of FIR Digital filters using Fourier
Series method(contd.)
• The samples of hd(n) can be determined taking the IFT of Hd(ejω)

• The impulse response obtained in above equation is an infinite

duration sequence. For FIR filters we truncate this infinite impulse
response to a finite duration sequence of length N,where N is odd.

V.RAJENDRA CHARY 148

 Design of FIR Digital filters using Fourier
Series method(contd.)

• The above transfer function represents a noncausal filter(due to the

presence of positive powers of z).Hence the transfer function is
multiplied by z-(N-1)/2.
V.RAJENDRA CHARY 149
Design of FIR Digital filters using Fourier
Series method(contd.)

V.RAJENDRA CHARY 150

 Design of FIR Digital filters using Fourier
Series method(contd.)
• We observe that the causality is brought about by multiplying the
transfer function by the delay factor (N-1)/2.
• This modification does not affect the amplitude response of the
filter, however the abrupt truncation of the Fourier series results in
oscillations in the passband and stopband.
• These oscillations are due to the slow convergence of the Fourier
series, particularly near the points of discontinuity. This effect is
known as Gibbs phenomenon.

V.RAJENDRA CHARY 151

 Design of FIR Digital filters using Fourier
Series method(contd.)
• The undesirable oscillations can be reduced by multiplying the
desired impulse response coefficients by an appropriate window
function.

V.RAJENDRA CHARY 152

Design of FIR Digital filters using Fourier
Series method

V.RAJENDRA CHARY 153

Specifications and desired impulse response
for FIR filter design by Fourier Series
method

V.RAJENDRA CHARY 154

V.RAJENDRA CHARY 155
 Design of FIR filter using Windows
• The windows are finite duration sequences used to modify the
impulse response of the FIR filters in order to reduce the ripples in
the passband and stopband and also to achieve the desired
transition from passband to stopband.

V.RAJENDRA CHARY 156

 Design of FIR filter using Windows(contd.)

• The procedure for designing FIR filter using windows is

1) Choose the desired frequency response of the filter Hd(ω).
2) Take inverse Fourier transform of Hd(ω) to obtain the desired
impulse response hd(n).
3) Choose a window sequence w(n) and multiply hd(n) by w(n) to
convert the infinite duration impulse response to a finite duration
impulse response h(n)

V.RAJENDRA CHARY 157

 Design of FIR filter using Windows(contd.)

4) The transfer function H(z) of the filter is obtained by taking Z-

transform of h(n)

V.RAJENDRA CHARY 158

 Design of FIR filter using Windows(contd.)

The various window techniques are:

• Rectangular window
• Triangular or Barlett’s window
• Hanning window
• Hamming window
• Kaiser window

V.RAJENDRA CHARY 159

 Rectangular Window
• The N-point rectangular window sequence is given by

• The frequency response is obtained by taking Fourier transform of

rectangular window sequence

V.RAJENDRA CHARY 160

Rectangular Window(contd.)

V.RAJENDRA CHARY 161

 Rectangular Window(contd.)
• The rectangular window sequence and its frequency response for
N=31 is as shown in figure:

V.RAJENDRA CHARY 162

Rectangular Window(contd.)

V.RAJENDRA CHARY 163

 Rectangular Window(contd.)
• The spectrum of WR(ejω) has two features that are very
important,they are width of main-lobe and side-lode amplitude.
• For rectangular window,the main-lobe width is equal to 4π/N.
• The maximum side-lobe magnitude for WR(ejω) occurs for the first
side-lobe and is approximately equal to -13dB.
• As the window is made longer the main lobe becomes narrower
and higher and the side lobes become more concentrated around ω
=0.

V.RAJENDRA CHARY 164

 Rectangular Window(contd.)
• From the plot of frequency response, we find that the frequency
response differs from the desired response in many ways.
• It doesnot follow quick transition in the desired response.
• The desired response changes abruptly from passband to
stopband,but the frequency response changes slowly.
• This region of gradual change is called filter’s transition
region,which is due to the convolution of desired response with
window response’s main lobe.

V.RAJENDRA CHARY 165

 Rectangular Window(contd.)
• As the filter length N increases, the main lobe becomes narrower
decreasing the width of the transition region.
• The convolution of the desired response and window response’s
sidelobes give rise to ripples in both the passband and stopband.
• The amplitude of the ripples is dictated by the amplitude of the
sidelobes.
• For rectangular window,the amplitude of the sidelobes is
unaffected by the length of the window.

V.RAJENDRA CHARY 166

 Rectangular Window(contd.)
• So increase in length N will not reduce the ripples, but increase its
frequency.
• An American mathematician Josiak Willard Gibbs showed that a
finite length lowpass filter will possess an 8.9% maximum ripple
no matter how long the filter is made.
• This effect where maximum ripple occurs just before and after the
transition band is known as Gibbs phenomenon

V.RAJENDRA CHARY 167

 Rectangular Window(contd.)
• The Gibbs phenomenon can be reduced by using a less abrupt
truncation of filter coefficients.
• This can be achieved using a window function that tapers smoothly
towards zero at both ends.

V.RAJENDRA CHARY 168

 Rectangular Window(contd.)
• The main lobe of the response is the portion that lies between the
first two zero crossings.
• The side lobes are defined as the portion of the response for ω <-2
π /N or ω >2 π /N.

V.RAJENDRA CHARY 169

 Triangular/Bartlett’s Window
• The N-point Triangular window sequence is:

V.RAJENDRA CHARY 170

 Triangular/Bartlett’s Window(contd.)
• The frequency response of Triangular window is:

V.RAJENDRA CHARY 171

 Triangular/Bartlett’s Window(contd.)
• The Triangular/Bartletts’s window sequence and its frequency
response for N=31 is as shown in figure:

V.RAJENDRA CHARY 172

Triangular/Bartlett’s Window(contd.)

V.RAJENDRA CHARY 173

 Triangular/Bartlett’s Window(contd.)
• In triangular window, the side-lobe level is smaller than that of
rectangular window being reduced from -13dB to -25dB.However
the main-lobe width is 8π/N(i.e twice that of rectangular window).
• The Triangular window produces a smooth magnitude response in
both passband and stopband.

V.RAJENDRA CHARY 174

 Triangular/Bartlett’s Window(contd.)
• The triangular window has the following disadvantages when
compared to magnitude response obtained by using rectangular
window
the transition region is more
the attenuation in stopband is less.
• Because of these disadvantages, the triangular window is not a
good choice.

V.RAJENDRA CHARY 175

 Raised cosine window
• The N-point Raised cosine window sequence is:

V.RAJENDRA CHARY 176

 Raised cosine window(contd.)
• The frequency response of raised cosine window is given by:

V.RAJENDRA CHARY 177

 Hanning Window
• The Hanning window sequence is one type of raised cosine
window, which is obtained by putting a=0.5 in raised cosine
sequence.

V.RAJENDRA CHARY 178

 Hanning Window(contd.)
• The frequency response of Hanning window is given by:

V.RAJENDRA CHARY 179

 Hanning Window(contd.)
• The Hanning window sequence and its frequency response for
N=31 is as shown in figure:

V.RAJENDRA CHARY 180

Hanning Window(contd.)

V.RAJENDRA CHARY 181

 Hanning Window(contd.)
• The main-lobe width of Hanning window is twice that of the
rectangular window, which results in a doubling of the transition
region of the filter.
• The magnitude of the side-lobe is -31dB which is 18dB lower than
that of rectangular window
• This results in smaller ripples in both stopband and passband of
lowpass filter designed using Hanning window.

V.RAJENDRA CHARY 182

 Hanning Window(contd.)
• The minimum stopband attenuation of the filter is 44dB which is
23dB lower than the filter designed using rectangular window. At
higher frequencies, the stopband attenuation is even greater.
• When compared to triangular window, the main-lobe width is same
but the magnitude of the side-lobe is reduced. Hence the Hanning
window is preferable to triangular window.

V.RAJENDRA CHARY 183

 Hamming Window
• The Hamming window sequence is given by:

• This can also be obtained by putting a=0.54 in raised cosine window

sequence
V.RAJENDRA CHARY 184
 Hamming Window(contd.)
• The frequency response is given by:

V.RAJENDRA CHARY 185

 Hamming Window(contd.)
• The Hamming window sequence and its frequency response for
N=31 is as shown in figure:

V.RAJENDRA CHARY 186

Hamming Window(contd.)

V.RAJENDRA CHARY 187

 Hamming Window(contd.)
• The peak sidelobe level is down about 41dB from the main lobe
peak, an improvement of 10dB relative to the Hanning window.
But this improvement is achieved at the expense of the side-lobe
magnitudes at higher frequencies, which are almost constant with
frequency(in Hanning window, the side-lobe amplitudes decrease
with frequency).
• At high frequencies the stopband attenuation is low when
compared to that of Hanning window.

V.RAJENDRA CHARY 188

 Hamming Window(contd.)
• Because the hamming window generates lesser oscillation in the
side lobes than that of Hanning window for the same main lobe
width, the Hamming window is generally preferred.

V.RAJENDRA CHARY 189

Summary of Window parameters

V.RAJENDRA CHARY 190

 Kaiser Window
• From the summary of window parametrs,we can find that a trade-
off exists between mainlobe width and the sidelobe amplitude.
• The main lobe width is inversely proportional to N.
• An increase in window length decreases the transition band of the
filter.
• However the stopband attenuation is independent of N and is a
function of the selected window.

V.RAJENDRA CHARY 191

 Kaiser Window(contd.)
• Thus inorder to achieve prescribed minimum stopband attenuation
and passband ripple,the designer must find a window with
appropriate sidelobe level and then choose N to achieve the
prescribed transition width.
• In this process,the designer may often have to settle for a window
with undesirable design specifications.
• To overcome this problem,Kaiser has chosen a class of windows
based on the prolate spheroidal functions.

V.RAJENDRA CHARY 192

 Kaiser Window(contd.)
• These functions have the property that are limited as much as
possible in both time and frequency domains.

V.RAJENDRA CHARY 193

 Kaiser Window(contd.)
• The kaiser window is given by:

V.RAJENDRA CHARY 194

 Kaiser Window(contd.)
Where α is the adjustable parameter and Io(x) is the modified zeroth-
order Bessel function of the first kind of order zero given by:

V.RAJENDRA CHARY 195

 Kaiser Window(contd.)
• The frequency response is given by:

V.RAJENDRA CHARY 196

 Kaiser Window(contd.)
• The kaiser window sequence and its frequency response for α=4.5
and N=31 is as shown in figure:

V.RAJENDRA CHARY 197

Kaiser Window(contd.)

V.RAJENDRA CHARY 198

 Design steps in Kaiser window
• Consider the lowpass filter specification shown in figure

V.RAJENDRA CHARY 199

 Design steps in Kaiser window(contd.)
• Let the passband ripple in decibels, minimum stopband attenuation
in decibels and transition width in radians/sec are given
respectively by:

B= ωs- ωp
ωp= passband frequency in rad/sec, ωs= stopband frequency in rad/sec
V.RAJENDRA CHARY 200
 Design steps in Kaiser window(contd.)
• A filter with a passband ripple equal to or lessthan αp’,a minimum
stopband attenuation equal to or greater than αs’ and a transition
bandwidth B can be designed using the following procedure:
1)Determine hd(n) using Fourier series method for an ideal frequency
response.

V.RAJENDRA CHARY 201

 Design steps in Kaiser window
2)Choose δ in the equations of αp and αs such that αp ≤ αp’ and
αs ≥ αs’

V.RAJENDRA CHARY 202

 Design steps in Kaiser window
3)Calculate αs using the formula

4)Determine the parameter α from the following equation

V.RAJENDRA CHARY 203

 Design steps in Kaiser window
5)Choose the parameter D as follows

6)Choose the filter order for the lowest odd value of N

V.RAJENDRA CHARY 204

 Design steps in Kaiser window
7)Compute the window sequence using the equation of kaiser
window.
8)Compute the modified impulse response using h(n)=wk(n)hd(n).
9)The transfer function is given by:

V.RAJENDRA CHARY 205

 Design steps in Kaiser window
10)The magnitude response can be obtained using

Where a(0)=h((N-1)/2)
a(n)=2h((N-1)/2 – n)

V.RAJENDRA CHARY 206

 Advantages of Window design for FIR filter

• The filter coefficients can be obtained with minimum computation

effort.
• The window functions are readily available in closed form
expression.
• The ripples in both passband and stopband are almost completely
removed.

V.RAJENDRA CHARY 207

 Disadvantages of Window design for FIR
filter
• It is not always possible to obtain a closed form expression for the
Fourier series coefficients h(n).
• Windows provide limited flexibility in the design.
• It is somewhat difficult to determine, in advance, the type of
window and the duration N required to meet a given prescribed
frequency specification.

V.RAJENDRA CHARY 208

 Frequency Sampling Technique
• Let h(n) be the filter coefficients of an FIR filter and H(k) be the
DFT of h(n).

V.RAJENDRA CHARY 209

 Frequency Sampling Technique(contd.)
• The DFT samples H(k) for an FIR sequence can be regarded as
samples of the filter z-transform evaluated at N-points equally
spaced around the unit circle.
i.e

The transfer function H(z) of an FIR filter with impulse response is

given by:

V.RAJENDRA CHARY 210

 Frequency Sampling Technique(contd.)
• Substitute the equation of h(n) in H(z),we get

V.RAJENDRA CHARY 211

 Frequency Sampling Technique :Type-I
design procedure
1)Choose the ideal(desired) frequency response Hd(ejω)
2)Sample Hd(ejω) at N-points by taking ω=ωk=2πk/N where
k=0,1,2,…(N-1) to generate the sequence H(k).

V.RAJENDRA CHARY 212

 Frequency Sampling Technique(contd.)
3)Compute the N samples of impulse response h(n) using the
following equations:

V.RAJENDRA CHARY 213

 Frequency Sampling Technique :Type-I
design procedure
4) Take Z-transform of the impulse response h(n) to get the filter
transfer function H(z)

V.RAJENDRA CHARY 214

 Frequency Sampling Technique :Type-II
design procedure
1)Choose the ideal(desired) frequency response Hd(ejω)
2)Sample Hd(ejω) at N-points by taking ω=ωk=2π(2k+1)/N,where
k=0,1,2,…(N-1) to generate the sequence H(k).

V.RAJENDRA CHARY 215

 Frequency Sampling Technique :Type-II
design procedure
3)Compute the N samples of impulse response h(n) using the
following equations:

V.RAJENDRA CHARY 216

 Frequency Sampling Technique :Type-II
design procedure
4) Take Z-transform of the impulse response h(n) to get the filter
transfer function H(z)

V.RAJENDRA CHARY 217

Comparision of IIR and FIR filters

V.RAJENDRA CHARY 218

V.RAJENDRA CHARY 219
 Problem
• Design an ideal lowpass filter with a frequency response

Find the values of h(n) for N=11.Find H(z) using Fourier Series
method and calculate frequency response
(R1 eg6.5)
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 Problem
• Design a filter with

H(ejω ) ≤ e-j3ω , -Π/4 ≤ ω ≤ Π/4

=0 , Π/4 ≤ |ω| ≤ Π

Using Hamming window with N=7 (R1 eg6.10)

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 Problem
• Determine the coefficients of a linear phase FIR filter of length
N=15 which has a symmetric unit sample response and frequency
response that satisfies the conditions using type-I design (R3 eg6.9)

V.RAJENDRA CHARY 222

V.RAJENDRA CHARY 223
 Problem
• Design a high pass filter with specifications

H(ejω ) =1 , -Π/4 ≤ |ω| ≤ Π

=0 , |ω| ≤ Π/4

using hanning and hamming window (R1 eg6.9)

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 Problem
• Design a FIR lowpass filter with cutoff frequency 1KHz and
sampling frequency of 4KHz with 11 samples using Fourier series
method (R3 eg.6.1)

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 Problem
• Design a linear phase FIR lowpass filter using rectangular window
by taking 7 samples of window sequence and cutoff frequency,
ωc=0.2π rad/sample (R3 eg6.5)

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 Problem
• Design a linear phase FIR highpass filter using hamming window
with a cutoff frequency, ωc=0.8π rad/sample and N=7 (R3 eg6.6)

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 Problem
• Design a linear phase FIR bandpass filter to pass frequencies in the
range 0.4π to 0.65π rad/sample by taking 7 samples of hanning
window sequence (R3 eg6.7)

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 Problem
• Design a FIR lowpass filter satisfying the following specifications
using Kaiser window(R1 e.6.11)

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 Problem
• Determine the frequency response of FIR filter defined by
y(n)=0.25x(n)+x(n-1)+0.25x(n-2) (R1 eg6.1)

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 Application
• Design of IIR/FIR digital filter conforming to given specifications.

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Appendix

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Appendix

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Appendix

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Appendix

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 Conclusion of the Unit
In this unit we studied about:
• Introduction
• Analog Filter Approximations – Butterworth and Chebyshev
Approximations.
• IIR digital filters: Design of IIR Digital filters from analog filters-
Impulse Invariance, Step invariance and Bilinear Transformation
methods, Design Examples
• Analog-Digital transformations.

V.RAJENDRA CHARY 236

 Conclusion of the Unit(contd.)
• FIR digital filters: Characteristics of FIR Digital Filters, frequency
response.
• Design of FIR Digital Filters using Fourier series method,
Windowing Techniques-Rectangular, Triangular, Hamming,
Hanning and Bartlett’s Windows, Steps in Kaiser windowing
method ,Frequency Sampling technique .
• Comparison of IIR and FIR filters.
• Applications
• Appendix
V.RAJENDRA CHARY 237
 Thank you
For Your Attention !
Any Questions
V.RAJENDRA CHARY 238