Lesson17 2

Dr. M. Venu Gopala Rao, Professor, Dept.
of ECE
KL University, Vaddeswaram,
Dept. of ECE,
Signal Processing (B. Tech all branches) 13-ES205
Lesson-17: IIR-Digital Filters
A digital filter is a Linear Time Invariant System. The input
sequence is modified according to the characteristics of the
system and gives some output. Hence the system is acting
some kind of filtering operation. In time domain the input and
outputs are related by y[n] h[n] x[n] , where x[n] , y[n] and
h[n] are the input sequence, output
sequence and impulse response of the system respectively. In frequency domain, they are related as
Y [ z] H [ z] X [ z] , where H [ z ]
Y [ z]
is referred to as System Function or Transfer Function or
X [ z]
Frequency Response of the system. Hence any filter can be characterized by either impulse response
h[n] or system function H [ z ] .

1. Conditions on Digital Filter:
a) Realizable filter condition: The impulse response of the system must be causal sequence.
h[n] 0, for n 0 .
b) Stable Condition: Absolutely summable. | h[n] |

n0
2. Classification of Digital Filters: Digital filters are categorized into two types.
(a) IIR Filter: Infinite duration Impulse Response. Recursive type. The present output sample
depends on the present input sample, past input samples and output samples.
(b) FIR Filter: Finite duration Impulse Response. Non-recursive type. The present output sample
depends on the present input sample and past input samples.
Frequency Selective Filters: A filter is one which rejects un-wanted frequencies from the input signal
and allow the desired frequencies to obtain the required shape of the output signal. The filters are
categorized into four types as described below:
Dr. M. Venu Gopala Rao, Professor, Dept. of ECE

Ideal Filters: The ideal digital filter transfer functions are illustrated below
1, for 0 || c
0, for c ||
1. Low Pass filter H [e j ]
0, for 0 || c
1, for c ||
2. High Pass Filter H [e j ]
0, for 0 || c
1
3. Band Pass Filter H [e j ] 1, for c || c

1
2
0, for c2 ||
1, for 0 || c
1
4. Band Stop Filter H [e j ] 0, for c || c

1
2
1, for c2 ||
Practical Filters
3. IIR Digital Filters: The input output relation of a IIR digital filter is described by a difference
equation:
N
k 1
k 0
y[n] ak y[n k ] bk x[n k ]

where ak and bk are filter coefficients, N is the order of the filter, N > M. The transfer function of the
digital filter is represented by
M
H [ z]
k
bk z
k 0
N
1 ak z k
k 10
The design of an IIR filter for the given specifications is determining filter co-efficients ak and bk of the
filter.
2

3(a) Design of Digital Filters from Analog Filters:
The most common technique used for designing IIR digital filters involves first designing an analog
prototype filter and then transforming the prototype to digital filter. For a given specifications of a digital
filter, the derivation of digital filter transfer function requires the following steps.
1. Map the desired digital filter specifications into those for an equivalent analog filter.
2. Derive the analog filter transfer function for analog prototype.
3. Transform the transfer function of the analog prototype into an equivalent digital filter transfer
function.
3(b) Magnitude response of the Filter:
p : Pass band frequency (digital) in radians
p : Pass band frequency (Analog) in radians
c : Cut off frequency (digital) in radians
c : Cut off frequency (Analog) in radians
s : Stopband frequency (digital) in radians
s : Stopband frequency (Analog) in radians
: Parameter specifying allowable pass band
: Parameter specifying allowable stop band
p : Passband error tolerance.
s : Max allowable magnitude on the stop band.
4 Analog Lowpass Filter Design: The most general form of analog filter transfer function is described
M
bk s k
N (s)
by H [ s ]
, where H [ s] is L.T. of h(t ) , ak and bk are filter coefficients, N is the
k 0
D( s) 1 N a s k
k
k 1
order of the filter, N > M. For stable filter, the poles of H [ s] must lie in the left half of s-plane.
3
5. Analog Lowpass Butterworth Filter: All pole filter. The magnitude function of the Butterworth
LPF
| H ( j) |
2N
, N = 1, 2, . .
where N is the order of the filter and c is cutoff frequency.

Properties:
(a) Monotonically decreasing function
(b) Smoothing at both passband and stopband.
(c) Max response at 0 .
(d) For c , | H ( j) | 1
(e) For c , | H ( j) | 1 or 3-dB
frequency or Half power frequency

(f) For c , | H ( j) | decreases rapidly.
The normalized Butterworth Lowpass Filter (i.e., c 1 rad/sec) is given by
| H ( j) |2
1
, N = 1, 2, . . .
1 () 2 N
1
1
As we know that s j or sj , we simplified as | H ( j) |2 | H (s 2 ) |

.
s
2
N
1 ( j )
1 ( s 2 ) N
Then the roots of the denominator is 1 (s 2 ) N 0 .
Case1: For N-odd: s 2 N 1 e j 2 k , k 1, 2,... , 2 N . sk e j k / N , k 1, 2,. . . , 2N .
Case2: For N-even:
s 2 N 1 e j (2k 1) , k 1, 2,... , 2 N . sk e j (2k 1) /2 N , k 1, 2,. . . , 2N .
For stable filter the poles must lie on the left half of s-plane. Therefore the stable files are derived from
the following equation:
sk e jk ,
where k
2
(2k 1)
,
2N
k 1, 2,. . . , N .

List of Butterworth polynomial
Denominator of H ( s)
(s 1)
(s 2 2s 1)
(s 1)(s 2 s 1)
(s 2 0.765s 1)(s 2 1.848s 1)
Determining the order of the filter:

From the given specifications determine the order of the filter
using the following formulas:
p : Pass band frequency (Analog) in radians.
p : Pass band attenuation

s : Stopband frequency (Analog) in radians.
s : Stop band attenuation

Case1: If p and s are given
Case2: If 1 and 2 are given
100.1 s 1
log
100.1 p 1
N

log s
p
log 2
1
2 1
1
N

log s
p
Case3: If and are given
log

N

log s
p
Round to the next higher integer.
Ex1: Determine the order of LPF if it has pass band attenuation of - 3dB at 500 Hz and stop
band attenuation of - 40dB at 1000 Hz.
Ans: The pass band attenuation is -3 dB.
The stop band attenuation is 40 dB.
The pass band cut off frequency in radian is p = 500 2 1000 rad / sec
5
The stop band cut off frequency in radian is s 1000 2 2000 rad / sec
p 3 dB; s 40 dB; p 1000 rad / sec; s 2000 rad / sec;

100.1 s 1
99.995
log
100.1 p 1 log
2
0.9976
The order of low pass filter is, N
6.64
s
2000 0.3
log
log
p
1000
Therefore the order of filter, N 6.64 7 .

% IIR1.m
clear all; close all; clc;
Op = 2*pi*500; Os = 2*pi*1000; rp = 3; rs = 40; T = 1;
N = ceil((log10((10.^(0.1*abs(rs))-1)./(10.^(0.1*abs(rp))-1)))/(2*log10(Os/Op)));
% N = 7
Ex2: Determine the order of LPF if it has pass band attenuation of - 1dB at 4 KHz and stop
band attenuation of - 40dB at 6 KHz.

Ans: The pass band attenuation is -1 dB.
The stop band attenuation is 40 dB.
The pass band cut off frequency in radian is p 4000 2 8000 rad / sec
The stop band cut off frequency in radian is s 6000 2 12000 rad / sec
p 1dB; s 40 dB; p 8000 rad / sec; s 12000 rad / sec;
The order of low pass filter is,
100.1 s 1
log
100.1 p 1
13.0239
N
s
log
p
% IIR2.m
Op = 2*pi*4000; Os = 2*pi*6000; rp = 1; rs = 40;
% N = ((log10((10.^(0.1*abs(rs))-1)./(10.^(0.1*abs(rp))-1)))/(2*log10(Os/Op)))
N
7. IIR Digital Filter Design:

7.1 Impulse Invariance Technique:
Mapping from s-plane poles to z-plane poles:
Let us consider the mapping of points from the s-plane to z-plane implied by the relation
z e sT , where s j and T is sampling interval. By substituting in z , we get
z e sT e( j)T e T e jT
And also we know that z re
. Therefore re
jT
.
e T e
r e T which is magnitude and T which is angular position.

Therefore the analog pole is mapped to a place in the z-plane of magnitude e T and angle T .
The real part of analog pole determines the radius of the z-plane pole and the imaginary part of analog
pole dictates the angle of the digital pole. The mapping of various positions s-plane poles into z-plane
are illustrated in the following figures.
Fig(a): Poles on j axis mapping onto the unit circle. (Marginally stable system).
Fig(b): Poles on left half of s-plane are mapping into inside the unit circle. (Stable system).
Fig(c): Poles on right half of s-plane are mapping into outside the unit circle. (Unstable system).
Design Procedure:
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 Unit Impulse Response of analog filter ha (t ) .
The Analog Filter transfer function is described by
H a ( s)
Ai
i 1 s pi
A1
s p1
A2
s p2
..
where Ai are the coefficients of partial fraction expansion, and
pi p1, p2 ,..., pN are the poles.

Then the impulse response ha (t ) = Inverse L.T. of H a ( s) . That is ha (t )
The Impulse Response
Ai e
pit
i 1
h[n] of digital filter is obtained by sampling the Impulse Response of analog
filter ha (t ) . Thus h[n] ha (t ) |t nT
Ai e
i 1
pi nT
, where T is sampling time.
Then the transfer function of digital filter is given by
H [ z]
n0
n0 i 1
h[n] z n Ai e
pi nT n
z
By interchanging the order of summation, we get

n
N p nT
N
piT 1
n
i
H [ z ] Ai
e
z Ai
e
z
i 1 n0
i 1 n0
N
Ai
pT
i 1 1 e i z 1
Limitations:
(a) Analog filters are band limited, so there will be aliasing due to sampling process. Because of
this aliasing, the frequency response of resulting digital filter will not be identical to the original
frequency response of analog filter.
(b) The change in the value of sampling time has no effect on the amount of aliasing.
(c) The analog frequency is in the range to , which maps into digital frequency
range of
in the
to . For a general range (i 1) T to (i 1) T , (where i is an integer) also
maps in the range of
to . Hence mapping from analog frequency to digital frequency is

8

termed as many to one. This mapping is not one to one.
Design Steps:
Step1: For the given specifications, determine the analog filter transfer function H a ( s) .
Step2: Select the sampling rate of the digital filter T sec/sample.
Step3: Express the analog filter transfer function as the sum of single pole filters.
N
H a ( s)
Ai
i 1 s pi
A1
s p1
A2
s p2
..
Step4: To convert analog low pass filter to digital low pass filter using impulse-invariant
transformation, substitute,
1
1
pk T 1 .
s pk
1 e z
Compute the Z.T. of the digital filter by using the formula: H [ z ]

Ex3: Consider the analog filter transfer function H a ( s)
Ai
pT
i 1 1 e i z 1
2
s2
Design digital filter using Impulse Invariance Method. Assume suitable data.
Ans: The sampling period is assumed to be T = 1 sec.
For the given analog filter transfer function the pole is p = -2.
To convert analog low pass filter to digital low pass filter using impulse-invariant transformation,
substitute,
1
1
pk T 1 .
s pk
1 e z
Therefore
H [ z]
1
p T
1 e k z 1
2
1 0.1353z 1
% IIR3.m.
% Analog poles and zeros
b = [2]; a = [1 2];
T = 1; fs = 1/T;
% Assume sampling frequency = 1 Hz.
% Transformation from analog to digital using impulse invariance method.

[bz,az] = impinvar(b,a,fs);
bz
az
% Normalized TF coefficients H[e^jw] = H[e^jw]/(H[e^j0]);
H0 = 2/(1-0.1353) ;
bz = bz/H0;
freqz(bz,az,512,fs);
Magnitude (dB)
1
0
-1
-2
-3
0.05
0.1
0.15
0.2
0.25
0.3
Frequency (Hz)
0.35
0.4
0.45
0.5
0.05
0.1
0.15
0.2
0.25
0.3
Frequency (Hz)
0.35
0.4
0.45
0.5
Phase (degrees)
0
-2
-4
-6
-8
7.2 Bilinear Transformation:

The limitations observed in Impulse Invariance method are overcome by using Bilinear Transformation.
The Bilinear Transformation is one to one mapping.
Let us consider an analog linear filter with system function
H a ( s)
b
sa
Then the corresponding digital filter transfer function is defined as H [ z ]
where s
b
2 1 z 1
T 1 z 1 a
2 1 z 1
1 sT / 2
.
or z
1
1 sT / 2
T 1 z
This relationship is known as Bilinear Transformation. The corresponding frequency transformations

are given as
T
.
tan or 2 tan 1
T
2
2
Warping Effect:
(a) At low frequencies
2
or = T , i.e., the digital
.
T 2 T
frequency follows analog frequency linearly.

(b) At high frequencies, the transformation is non-linear and distortion is introduced in the frequency
scale of the digital filter to that of analog filter as shown in the figure. This distortion is known as
warping effect . The warping effect on magnitude response and phase responses are shown
below.
10
Fig (a) Warping effect on magnitude
(b) Warping effect on phase angle
Pre-warping: This warping effect can be eliminated by pre-warping the analog frequencies as below:
p
2
2
and s tan s .
tan . Therefore p tan
T
2
T
2
T
2
Design Steps:
Step1: From the given specifications, find the prewarping analog frequencies using the formulas:
p
2
2
and s tan s .
tan
T
2
T
2
Step2: Using the analog frequencies determine the analog filter transfer function H a ( s) .
Step3: Select the sampling rate of the digital filter T sec/sample.
Step4: Determine the digital filter transfer function by substituting s
2 1 z 1
into the analog filter

T 1 z 1
transfer function.
11
Ex4: Design a digital low pass filter using bi-linear transformation for the following analog filter transfer
function.
H a ( s)
0.4225
s 2 0.9192s 0.4225
Ans: The digital filter transfer function is obtained as below:

By assuming T = 1 sec and by substituting s
H [ z ] H a ( s) |
s 2 1 z 1
T 1 z
2 1 z 1
T 1 z 1
0.4225
2
2 1 z 1
2 1 z 1
0.9192
0.4225

1
1
T
T
1
z
1
On simplification, we get
H [ z ] 0.0675
1 2 z 1 z 2
1 1.1428z 1 0.4127z 2
% IIR3.m.
b = [0.4225]; a = [1 0.9192 0.4225]; fs =1;
[bz,az] = bilinear(b,a,fs);
%bz = 0.0675
%az = 1.0000
0.1350
-1.1428
0.0675
0.4127
Magnitude (dB)
50
0
-50
-100
-150
0.05
0.1
0.15
0.2
0.25
0.3
Frequency (Hz)
0.35
0.4
0.45
0.5
0.05
0.1
0.15
0.2
0.25
0.3
Frequency (Hz)
0.35
0.4
0.45
0.5
Phase (degrees)
0
-50
-100
-150
-200
12
---------------------------------------------------------------------------------------------------------------------------------------
Algorithm for Design of Digital Butterworth Filter:

-------------------------------------------------------------------------------------------------------------------------The most common technique used for designing IIR digital filters involves first designing an analog
prototype filter and then transforming the prototype to digital filter. For a given specifications of a digital
filter, the derivation of digital filter transfer function requires the following steps.
1. Map the desired digital filter specifications into those for an equivalent analog filter.
2. Derive the analog filter transfer function for analog prototype.
3. Transform the transfer function of the analog prototype into an equivalent digital filter transfer
function.
Step1: Conversion of digital frequencies into analog frequencies.

Obtain the following digital filter parameters for the given problem.
p : Pass band frequency (digital) in radians.
s : Stop band frequency (digital) in radians.
s : Stop band attenuation
Convert these digital filter parameters into corresponding analog filter parameters according to the filter
type.
(a) Impulse Invariance Method:
Substitute for analog frequency
. That is p
and s s , where T is sampling
T
T
T
period. If the value of T is not explicitly given assume to be 1 sec.

(b) Bilinear Transformation Method:
Pre-warping the analog frequencies:
Therefore p
tan .
T
2
p
2
2
and s tan s .
tan
T
2
T
2
If the value of T is not explicitly given assume to be 1 sec or an appropriate value.
Step2: Determining the order of the analog low pass filter:

From the given specifications determine the order of the filter using the following formulas:
13
p : Pass band frequency (Analog) in radians.
s : Stopband frequency (Analog) in radians.
s : Stop band attenuation s , p , rs , rp
Case1: If p and s are given
Case2: If 1 and 2 are given
Case3: If and are given
100.1 s 1
log
100.1 p 1
N

log s
p
22
log
1
2 1
1
N
s
log
log

N

log s
p
Step3: Formulate the normalized analog low pass Butterworth filter transfer function as
1
H ( s)
Polynomial
where polynomial for various values of N is given below
List of Butterworth polynomial
Denominator of H ( s)
(s 1)
(s 2 2s 1)
(s 1)(s 2 s 1)
(s 2 0.765s 1)(s 2 1.848s 1)
Transfer Function
H ( s)
H ( s)
H ( s)
H ( s)
(s 1)
1
(s 2s 1)
1
(s 1)(s 2 s 1)
1
(s 0.765s 1)(s 2 1.848s 1)

14
Step4: Determine the cut off frequency using the formula c
Note: If
10
p
0.1 p
1
2N
p 3dB , c p
Step5: Determine the analog transfer function H a ( s) using the following:
s s
c
Low pass to high pass filter, substitute s c
(a) Low pass to low pass filter, substitute

(b)
Step6: Apply transformation technique to convert analog filter into respective digital filter.
(a) Impulse Invariance Technique: Substitute
1
s Pk
(b) Bilinear Transformation Technique: Substitute s
1
PT
1 e k z 1
2 1 z 1
T 1 z 1
Step7: The final digital filter transfer function must be presented in the following format.
b b z 1 b2 z 2 ....
H [ z ] 0 1 1
1 a1z a2 z 2 ....
15
Ex5: Design a digital low pass filter Butterworth filter using bilinear transformation with pass
band and stop band cut-off frequencies 800 rad/sec and 1800 rad/sec respectively. The pass
band attenuation is -3 dB and stop band attenuation is -10dB.
Ans: The given filter specification are:
p 3dB; s 10dB; p 800 rad / sec; s 1800 rad / sec

Apply the bilinear transformation technique for prewarping , i.e.
2

tan .
T
2
16

The value of T is not explicitly given in the problem; hence we assume T=1sec.
800
3.23 rad / sec
2
Therefore, p 2 tan
1800
s 2 tan
30.12 rad / sec
2
% IIR4.m.
% Bilinear transformation
wp = 800; ws = 1800; rp = 3; rs = 10; T = 1;
% Prewarping analog frequencies
Op = (2/T)*tan(wp/2);
Os = (2/T)*tan(ws/2);
Op
Os
100.1 s 1
3
log
100.1 p 1 log
0.997
The order of low pass filter is N
0.4913 1
s
30.12
log
log
3.23
N
The transfer function for 1 order normalized low pass filter is given by, H ( s)
The cut-off frequency, c
10
p
0.1 p
1
2N
1
s 1
3.23
3.24 rad / sec
(0.997)
Oc = Op / ((10^(0.1*abs(rp))-1)^(1/(2*N)));
Oc
The desired analog filter transfer function Replace, s
H a ( s)
b = [1]; a = [1
1
s
1
3.24
s
,
3.24
3.24
s 3.24
1]; % 1/(s+1); Normalized analog filter transfer function
[B A]=lp2lp(b,a, Oc); % Desired Analog filter transfer function with cut off
% frequency Oc
17

B
A
2 1 Z 1
To convert low pass filter to digital low pass filter using bilinear transformation, s
T 1 Z 1
in the magnitude response H ( s) .
Then the required digital filter transfer function H ( Z )
3.24
0.62(1 Z 1)
1 0.24Z 1
1 Z 1
2
3.24
1
1 Z
% Transformation from analog filter to digital filter

fs = 1/T;
[b1 a1] = bilinear(B,A, fs);
b1
a1
freqz(b1,a1,512,fs);
axis([0 fs/2 -20 1])
% --------------------------------------------------------------------------------
Magnitude (dB)
0
-5
-10
-15
-20
0.05
0.1
0.15
0.2
0.25
0.3
Frequency (Hz)
0.35
0.4
0.45
0.5
0.05
0.1
0.15
0.2
0.25
0.3
Frequency (Hz)
0.35
0.4
0.45
0.5
Phase (degrees)
-50
-100
Ex6: Design a digital low pass Butterworth filter using impulse-invariant transformation with pass band
and stop band cut off frequencies 200 Hz and 500 Hz respectively. The pass band and stop band cut
off frequencies 200 Hz and 500 Hz respectively. The pass band and stop band attenuation are -5 dB
and -12 dB respectively. The sampling frequency is 5000 Hz.
Ans: The given specifications of digital filter are:
p 5dB; s 12dB; Fpass 200 Hz; Fstop 500 Hz; Fs 5000Hz .

18

The corresponding digital frequencies are:
2 Fstop 2 500
2 Fpass 2 200
0.2 rad / sec
0.08 rad / sec and s

Fs
5000
Fs
5000
% IIR6.m:
% wp = 2*pi*(200/5000); ws = 2*pi*(500/5000); rp = 5; rs = 12; T = 1;
wp = 0.08*pi; ws = 0.2*pi; rp = 5; rs = 12; T = 1; fs=1/T;
Apply impulse-invariant transformation, T
so
.
T
Then the analog frequencies are given by
0.08 rad / sec and s s 0.2 rad / sec

T
T
% Impulse Invariance: Conversion of digital to analog frequencies

Op = 0.08*pi ; Os = 0.2*pi;
100.1 s 1
log
100.1 p 1
1.05 2
The order of low pass filter is, N

log s
p
N
The order of low pass filter for the given magnitude response is 2. The transfer function for II order
normalized low pass filter is given by,
H ( s)
The cut off frequency is given by c
1
s 2s 1
2
p
10
0.1 p
1
2N
0.2073 rad / sec
Oc = Op / ((10^(.1*abs(rp))-1)^(1/(2*N)));
Oc
The desired analog filter transfer function of low pass filter can be obtained by replacing, s
That is s
s
.
c
s
s
.
c 0.2073
Therefore the analog filter transfer function is computed as

19
H a ( s)
1
2
s
s
2
1
0.2073
0.2073
0.0430
s 2 0.2932s 0.0430
% Lowpass to low pass filter transformation

b = [1]; a = [1 sqrt(2) 1];
[B A]=lp2lp(b,a, Oc);
B
A
Impulse-Invariant Transformation:
By converting single pole filters as
0.1466j
0.1466j
H a ( s)
s (0.1466 j 0.1466) s (0.1466 j 0.1466)

To convert analog low pass filter to digital low pass filter using impulse-invariant transformation,
1
1
pk T 1 in the above equation.

s pk
1 e z
substitute,
The poles are given by p1 0.1466 j 0.1466, and p2 0.1466 j 0.1466 .

Then the digital filter transfer function is
H [ z]
0.0370z 1
1.0000 -1.7088z 1 0.7459z 2
[bz,az] = impinvar(B,A,fs);
bz
az
Magnitude (dB)
0
-10
-20
-30
-40
0.05
0.1
0.15
0.2
0.25
0.3
Frequency (Hz)
0.35
0.4
0.45
0.5
0.05
0.1
0.15
0.2
0.25
0.3
Frequency (Hz)
0.35
0.4
0.45
0.5
Phase (degrees)
0
-50
-100
-150
-200
20

Ex7: Design a Butterworth digital high pass filter with the following specifications
H (e j ) 0.2
0 0.2
0.8 H (e j ) 1
0.6
Using (i) Impulse-invariant transformation (ii) Bilinear transformation

Ans:
To design high pass filter, it is desired first to design LPF and transform into HPF. This is
illustrated as below.
The given specifications for high pass filter are:
1
1
0.2; 1
1
1
0.8; ph 0.6 ; sh 0.2
and s 20 log10 (0.2) 14dB
Then p 20 log10 (0.8) 1.9382 dB

The corresponding LPF specifications are
p 1.9382 dB, s 14dB, pl 0.2 and sl 0.6
(i) Impulseinvariant
transformation:
Apply impulse-invariant transformation, T
Since sampling period T is not given, assume T=1 second.
Therefore, p pT 0.2 rad / sec, and s sT 0.6 rad / sec
100.1 s 1
log
log
100.1 p 1
1.706
The order of the low pass filter is, N

log s
log s

p
p
The transfer function for second order normalized low pass filter is given by, H ( s)
The cut off frequency is c
10
p
0.1 p
1
2N
1
s 2 2s 1
0.7504 rad / sec
21
s sc
The denormalized high pass filter transfer function can be obtained by replacing s by
Then the analog filter transfer function
H a ( s)
1
2
0.7504
0.7504
2
1
s
s
s2
s 2 1.602s 0.5631
By converting into single pole filters
0.5129
0.5129
H a ( s) 1 2
2
s 0.5131 j 0.5131 s 0.5131 j 0.5131

Then the poles are p1 0.5131 j 0.5131, and p2 0.5131 j 0.5131
To convert analog high pass filter to digital high pass filter using impulse invariant transformation
1
1
pk T 1
s pk
1 e Z
Therefore, H ( z ) 1
0.5129
1 e
0.5129

(0.5131 j 0.5131)T 1
z
1 e
( 0.5131 j 0.5131)T 1
On simplification we get
H ( z)
0.025 0.508 z 1 0.3583z 2

1 1.043z 1 0.3583z 2
(ii) Bilinear Transformation:

Pre-warping in bilinear transformation
2

tan
T
2
0.2
0.6
p 2 tan
0.6498 rad / sec , and s 2 tan
2.7528 rad / sec
2
2
The order of low pass filter is
4.89
log
log

0.75 1.2986 2
N

2.7529
log s log
00.6498
p
The transfer function for second order normalized low pass filter is given by, H ( s)
The cut off frequency c
10
p
0.1 p
1
2N
s 2 2s 1
0.6498
0.7503 rad / sec
(0.75)1/2
22

The denormalized high pass filter transfer function can be obtained by
transformation s
c .
H a ( s)
1
2
0.7503
0.7503
2
1
s
s
s2
s 2 1.061s 0.562
To convert analog high pass filter to digital high pass filter using bilinear transformation
2 1 z 1
, we get the digital HPF transfer function as
T 1 z 1
H [ z]
1 z 1
4
1 z 1
1 z 1
1 z 1
4
1.061z 1 24
0.562
1 z 1
1 z 1
0.598 1.196 z 1 0.598 z 2

1 1.028 z 1 0.365 z 2
-------------------------------------------------------------------------------------------------------------------------------------
Design of Band Pass Filter

------------------------------------------------------------------------------------------------------------------------------------Formulate the BPF specifications into LPF specifications as shown in figure.
Find r min A , B , where A
12 l u
1(u l )
and B
22 l u
2 (u l )
log
.
Calculate the order of the filter. N
log r
Find the normalized LPF transfer function
The denormalized band pass filter transfer function H a ( s) can be obtained by replacing
s 2 l u
s(u l )
------------------------------------------------------------------------------------------------------------------------------------23

Ex6: Design a Butterworth band pass filter with the following specifications:
1 0.1 rad / sec; 2 0.7 rad / sec; l 0.25 rad / sec , u 0.35 rad / sec; p 3 dB and
s 30 dB . Using (i) Bilinear Transformation, and (ii) Impulse-invariant transformation
(i) Bilinear transformation:

The specifications for band pass filter are p 3 dB and s 30 dB
2 0.7 rad / sec; l 0.25 rad / sec u 0.35 rad / sec; T 1 sample / sec
Apply bilinear transformation,
2

tan
T
2
0.1
0.7
1 2 tan
0.31676 rad / sec; 2 2 tan
3.925 rad / sec
2
2
0.25
0.35
l 2 tan
0.8284 rad / sec; u 2 tan
1.225 rad / sec
2
2
12 l u
(0.3167)2 0.8284 1.225

7.279
1 (u l )
0.3167(1.225 0.8284)
22 l u (3.925)2 1.225 0.8284
B
9.244
2 (u l )
3.925(1.225 0.8284)
r min A , B 7.28 rad / sec
A
The order of the low pass filter is given by,
5.5338
log log

0.9976 0.863 1
N
log r
log 7.28
The transfer function for first order normalized low pass filter is given by, H ( s ) 1
s 1
Denormalized band pass filter can be obtained by replacing s
s 2 l u
s(u l )
24
s 2 l u s 2 0.8284 1.225
s( u l ) s(1.225 0.8284)
2
s s 1.048
0.3966s
Therefore, H ( s)
1
0.3966s
2
s 0.3966s 1.0148
s 1.0148
1
0.3966s
2
To convert analog low pass filter to digital band pass filter using bilinear transformation, substitute
2 1 Z 1
,
T 1 Z 1
1 Z 1
0.3966 2
1
1 Z
H (s)
2
1 Z 1
1 Z 1
4
0.3966
2
1.0148
1
1
1 Z
1 Z
On simplification, H ( z )
0.316(1 Z 2 )
1 1.027 Z 1 0.726Z 2
(ii)Impulse-invariant transformation:
Apply impulse-invariant transformation, T . Since sampling period T=1 sec
1 1 0.3141rad / sec; 2 2 2.199 rad / sec

l l 0.7854 rad / sec; u u 1.099 rad / sec
1 2 l u (0.3141) 2 0.7854 1.099
10.8852
1 ( u l )
0.3141(1.009 0.7854)
2 2 l u (2.199) 2 0.7854 1.099
8.079
2 ( u l )
2.199(1.099 0.7854)
r min A , B 8.079 rad / sec

The order of low pass filter is given as,
5.5338
log log

0.9976 0.82 1
N
log r
log 8.079
The transfer function for I order normalized low pass filter is given by H ( s)
1
.
s 1
25

Denormalized band pass filter can be obtained by replacing, s
s 2 lu
s ( u l )
s 2 l u s 2 0.7854 1.099
s( u l ) s(1.099 0.7854)
2
s s 0.863
0.3137 s
H ( s)
1
0.3137 s
2
s 0.3137 s 0.863
s 0.863
1
0.3137 s
2
On factorization, we get first order poles as
H a ( s)
0.1568 j 0.02686
0.1568 j 0.02686
s (0.1568 j 0.9156) s (0.1568 j 0.9156)
To convert analog band pass filter to digital band pass filter in impulse-invariant transformation.
1
1
pk T 1
s pk
1 e Z
The poles of equation (1) are p1 0.1568 j 0.9156; p2 0.1568 j 0.9156
0.1568 j 0.02686
0.1568 j 0.02686
0.1568
j
0.9156)
1
1 e
z
1 e( 0.1568 j 0.9156) z 1
0.1568 j 0.02686
0.1568 j 0.02686
j
0.9156
1
1 0.8549e
z
1 0.8549e j 0.9156 z 1
0.3136 0.1996 z 1
1 1.0416 z 1 0.7292 z 2
H ( z)
-------------------------------------------------------------------------------------------------------------------------------------
Design of Band Stop Filter

------------------------------------------------------------------------------------------------------------------------------------Formulate the BPF specifications into LPF specifications.
Find r min A , B , where A
1 (u l )
12 u l
and B
2 (u l )
22 u l
log
. Find the normalized LPF transfer function
Calculate the order of the filter. N
log r
The denormalized band stop filter transfer function H a ( s) can be obtained by replacing
s(u l )
s 2 l u
------------------------------------------------------------------------------------------------------------------------------------26

Ex7: Design a Butterworth band elimination filter with the following specifications:
p 2dB, s 10dB, and T 1sec .The pass band frequencies are 0.07 and 0.8 and stop band
frequencies are 0.2 and 0.3 .
Use (i) Impulsive-invariant transformation, and (ii) Bilinear transformation
Ans: Given specification for band elimination filter are,
p 2dB, s 10dB,0.07 p 0.8 , 0.3 s 0.2

(i) Impulsive-invariant transformation
Apply Impulse-invariant transformation,
since T=1 sec
T
1 0.2 rad / sec; 2 0.3 rad / sec; l 0.07 rad / sec; u 0.8 rad / sec
A
1 (u l )
0.2 (0.8 0.07 )
9.125
12 u l (0.2 )2 0.8 0.07
2 (u l )
0.3 (0.8 0.07 )
6.441
2 2 u l
(0.3 )2 0.8 0.07
Therefore r min A , B 6.441

Then the order of the low pass filter is given by,
3
log log

0.7647 0.7338 1
N
log r
log 6.441
The transfer function for I order normalized low pass filter is given by , H ( s ) 1
s 1
Denormalized band pass filter can be obtained by replacing s
s 2 l u
s(u l )
s(0.8 0.07 )
s 2 0.8 0.07
Therefore, H a ( s)
s 2 l u
s(u l )
2.29s
s 0.552
2
1
s 2 0.552
s 2 2.29s 0.552
2.29s
2
1
s 0.552
By computing single pole filter H a ( s) 1
2.65
0.3598
s 2.0162 s 0.2738
27

To convert analog band elimination filter to digital band pass filter in impulse-invariant
transformation substitute
H ( z) 1
1
1
p
s pk
1 e k T z 1
2.65
1 e2.0162T z 1
0.36
1 e0.2735T z 1
2.65
0.36
1
1 0.1332 z
1 0.7605z 1
On simplification, we get the digital filter transfer function as
H ( z)
2 1.6011z 1 0.1z 2
1 0.8938 z 1 0.1z 2
(ii) Bilinear transformation: By applying Bilinear transformation, we get
0.3
0.6495 rad / sec; 2 2 tan
1.018 rad / sec
2
0.07
0.8
l 2 tan
0.2206 rad / sec; u 2 tan
6.142 rad / sec
2
2
(u l )
0.6495(6.142 0.2206)
A 1 2
4.1216
1 u l (0.6495)2 6.142 0.2206
2

0.2
tan 1 2 tan
T
2
2
2 (u l )
22
u l
1.018(6.142 0.2206)
18.926
(1.018)2 6.142 0.2206
r min A , B 4.1216
The order of the low pass filter is given by,
3
log log

0.7647 0.965 1
N
log r log 4.1214
The transfer function for I order normalized low pass filter is given by , H ( s ) 1
s 1
s ( l )
Denormalized band pass filter can be obtained by replacing s 2 u
s lu
s(u l )
s(6.142 0.2206)
5.9214s
s
2
2
s 6.142 0.2206
s 1.3549
s l u
1
s 2 1.3549
Therefore, H ( s)
s 2 5.921s 1.3549
5.9214s
2
1
s 1.3549
2
To convert analog low pass filter to digital band pass filter using bilinear transformation, substitute
2 1 z 1
,
T 1 z 1
28

2
H [ z]
On simplification,
1 Z 1
2 1 Z 1 1.3549
1 Z 1
1 Z 1
2
5.9214
1.3549
1 Z 1
1 Z 1

H ( z)
0.3113 0.3076Z 1 0.3113Z 2

1 0.3076Z 1 0.377 Z 2
6. Chebyshev Filters:
Type1: All pole filters. Equi-ripple in the passband
monotonic in the stopband.
Type2: Both poles and zeros. Monotonic in the
passband and equiripple in the stopband.
The magnitude response of N-th order filter is expressed as
| H ( j ) | 2
1
2
1 2C N
p
where is ripple parameter and C N ( x) is the N-th order Chebyshev polynomial defined as
C N ( x) cos( N cos 1 x), | x | 1: Passband , and

C N ( x) cosh( N cosh 1 x), | x | 1: Stopband.
Properties:
C N ( x) C N ( x) , N -odd;
C N ( x) C N ( x) , N -even;
C N (0) 0 , N -odd;
N
C N (0) (1) 2 , N -even;

C N (1) 1 , for all N.
C N (1) 1 , N -even;
C N (1) 1 , N -odd;
C N ( x) oscillates with equal ripple between 1 for | x | 1 .
C N ( x) is monotonically increasing for | x | 1 , for all N.
Comparison with Butterworth Filter:
Chebyshev filter exhibits ripples in the passband or stopband, where as in Butterworth filter
monotonically decreases.
29

The transition band is more in Butterworth filter compared to Chebyshev filter.
The poles in Butterworth filter lies on circle and Chebyshev filter lies on ellipse.
The order of the Chebyshev filter is less than that of Butterworth filter. i.e., the number of poles
is more in Butterworth filter when compared to Chebyshev filter.
Chebyshev filter Design:
Step1: From the given specifications, find the order of the filter N,
0.1
p 1
10
100.1 s 1
cosh 1 s
p
cosh 1
log
22
12
1
1
log s
p
cosh 1
cosh 1
s
p

Step2: Using the following formulas, find the values of a and b which are minor and major areas of
the ellipse respectively.
N1 N1
a p
2
where
N1 N1
b p
2
1 2 1; 100.1 p 1 .
Step3: Calculate the poles of Chebyshev filter which lies on ellipse by using the formula:
sk a cos k j b sin k , where k
(2k 1)
,
2N
k 1, 2,. . . , N
Step4: To find the transfer function

1. Using above poles, find the denominator polynomial.
2. The numerator of 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 value by
1 2 .
30

Ex8: Design a digital Chebyshev low pass filter with the following specifications:
(a) The Impulse invariance analog prototype low pass filter transfer function
H a ( s)
0.262
s 2 0.512s 0.3277
(b) The Bi-linear transformation analog prototype low pass filter transfer function
H a ( s)
0.2809
s 2 0.53s 0.351
Ans: (a) The single pole filter transfer functions are derived from the given Impulse invariance analog
prototype filter transfer function as below:
H a ( s)
0.256 j
0.256 j
s (0.256 j 0.512) s (0.256 j 0.512)
The impulse Invariance transformation for converting an analog low pass filter to a digital low pass filter
is derived by substituting
H ( z)
1
1
p
s pk
1 e k T z 1
0.256 j
1 e(0.256 j 0.512)T z 1
On simplification, we get H ( z )
0.256 j
1 e(0.256 j 0.512)T z 1
0.194 z 1
1 1.3495 z 1 0.6 z 2
(b) The Bi-linear transformation for analog prototype low pass filter transfer function is computed by
substituting s
H ( z)
2 1 z 1
0.2809
in H a ( s)
, we get
1
2
T 1 z
s 0.53s 0.3561
0.2809
2
2 1 z 1
2 1 z 1
0.53
0.3561

1
T 1 z 1
T 1 z
1 2 z 1 z 2
0.052
1 1.349 z 1 0.609 z 2
Ex9: Design a digital Chebyshev high pass filter with the following specifications:
The analog prototype low pass filter transfer function
H a ( s)
0.06175
and the cut-off frequency is c 0.01 .
s 0.06175
Ans: Given that analog prototype low pass filter transfer function H a ( s)
The high pass filter transfer function is computed by substituting s
c
.
s
0.06175
s 0.06175
Therefore the high pass filter transfer function
31
0.06175
s
.
0.01
s
0.5085
0.06175
s
1
1
For Impulse Invariance transformation,
. Here pole pk 1 .
p
s pk
1 e k T z 1
H HPF ( s)
2 1 z 1

T 1 z 1
0.7973(1 z 1 )
H [ z]
1 0.596 z 1
2 1 z 1
0.5085

T 1 z 1
32
Exercise-14
1. Define the ideal digital filters both in mathematical expressions and graphical representations.
Compare the practical filters and with ideal filters.
2. Explain the magnitude response of a digital filter. Illustrate all the filter parameters.
3. Illustrate the steps involved in designing IIR digital high pass filters using Impulse Invariance
transformation.
4. Illustrate the steps involved in designing IIR digital low pass filters using Bi-linear transformation.
5. Design a digital low pass filter Butterworth filter with pass band and stop band cut-off frequencies
200 Hz and 500 Hz respectively. The pass band attenuation is -5 dB and stop band attenuation is 12 dB. The sampling frequency is 5 KHz.
Use (i) Impulse Invariance Method, and (ii) Bilinear transformation.
Hint: First convert analog frequencies into digital frequencies using formula 2
The digital pass band frequency p 2
The digital stop band frequency p 2
Fp
Fsam
F
Fsam
200
0.08 rad/sec and
5000
Fs
500
2
0.2 rad/sec.
Fsam
5000
Then Design the digital filter using the procedure given.

6. Design a digital Butterworth low pass filter with the following specifications:
0.7 | H [e j ] | 1,
0 0.2
| H [e j ] | 0.3,
0.2 0.6

7. The pass band and stop band cut-off frequencies are 350 Hz and 1000 Hz respectively. The
attenuation at pass band and stop band are -3 dB is -10 dB respectively. The sampling frequency is
5 KHz. Design a digital low pass filter Butterworth filter using (i) Impulse Invariance Method, and (ii)
Bilinear transformation.
8. Design a digital Butterworth high pass filter with the following specifications:
| H [e j ] | 0.2,
0.8 | H [e j ] | 1,
0 0.2
0.6
33

9. The pass band and stop band cut-off frequencies are 350 Hz and 1250 Hz respectively. The
corresponding pass band and stop band attenuation are -3 dB is -10 dB respectively. The sampling
frequency is 5 KHz. Design a digital high pass filter Butterworth filter using (i) Impulse Invariance
Method, and (ii) Bilinear transformation.
10. The specifications for digital band pass filter are as follows:
1 0.2 rad / sec; 2 0.7 rad / sec; l 0.35 rad / sec , u 0.35 rad / sec; p 3 dB
and s 10 dB . Design a digital Butterworth band pass filter Using
(i) Bilinear Transformation,
and (ii) Impulse-invariant transformation

11. Design a digital Butterworth band pass filter with the following specifications. The pass band cut-off
frequencies 600 Hz and 1000 Hz, and stop band frequencies 300 Hz and 1600 Hz respectively. The
pass band attenuation is -3 dB and stop band attenuation is -10 dB. The sampling frequency is 5
KHz. Use (i) Impulse Invariance Method, and (ii) Bilinear transformation.
12. Design a digital Butterworth band stop filter for the pass band attenuation is -3 dB and stop band
attenuation is -10 dB at sampling rate 1 second. The pass band frequencies are 0 0.03 and
0.7 , and the stop band frequencies range is 0.1 0.25 . Use (i) Impulse Invariance
Method, and (ii) Bilinear transformation.
13. Design a digital Chebyshev low pass filter with the following specifications:
0.457
The analog prototype filter transfer function H a ( s)

.
s 2 0.676s 0.571
14. Design a digital Chebyshev high pass filter with the following specifications:
The Impulse invariance analog prototype low pass filter transfer function H a ( s)
0.31416
and
s 0.31416
the cut-off frequency is 0.1 .

15. Compare Chebyshev filter approximations with Butterworth filter approximations.
16. Illustrate briefly how a digital filter is designed?
17. What is warping? Illustrate.
18. Illustrate briefly the designing steps for (a) Impulse Invariance (b) Bi-linear transformation.
34

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Dr. M. Venu Gopala Rao, Professor, Dept.

h[n] are the input sequence, output

h[n] or system function H [ z ] .

b) Stable Condition: Absolutely summable. | h[n] |

Dr. M. Venu Gopala Rao, Professor, Dept. of ECE

1. Low Pass filter H [e j ]

2. High Pass Filter H [e j ]

3. Band Pass Filter H [e j ] 1, for c || c

4. Band Stop Filter H [e j ] 0, for c || c

y[n] ak y[n k ] bk x[n k ]

Dr. M. Venu Gopala Rao, Professor, Dept. of ECE

3(b) Magnitude response of the Filter:

p : Pass band frequency (digital) in radians

p : Pass band frequency (Analog) in radians

c : Cut off frequency (digital) in radians

c : Cut off frequency (Analog) in radians

s : Stopband frequency (digital) in radians

s : Stopband frequency (Analog) in radians

: Parameter specifying allowable pass band

: Parameter specifying allowable stop band

p : Passband error tolerance.

s : Max allowable magnitude on the stop band.

Dr. M. Venu Gopala Rao, Professor, Dept. of ECE

where N is the order of the filter and c is cutoff frequency.

frequency or Half power frequency

As we know that s j or sj , we simplified as | H ( j) |2 | H (s 2 ) |

s 2 N 1 e j (2k 1) , k 1, 2,... , 2 N . sk e j (2k 1) /2 N , k 1, 2,. . . , 2N .

Dr. M. Venu Gopala Rao, Professor, Dept. of ECE

(s 2 0.765s 1)(s 2 1.848s 1)

Determining the order of the filter:

p : Pass band attenuation

s : Stop band attenuation

Case2: If 1 and 2 are given

Case3: If and are given

Round to the next higher integer.

Dr. M. Venu Gopala Rao, Professor, Dept. of ECE

p 3 dB; s 40 dB; p 1000 rad / sec; s 2000 rad / sec;

The order of low pass filter is, N

Therefore the order of filter, N 6.64 7 .

band attenuation of - 40dB at 6 KHz.

p 1dB; s 40 dB; p 8000 rad / sec; s 12000 rad / sec;

The order of low pass filter is,

Op = 2*pi*4000; Os = 2*pi*6000; rp = 1; rs = 40;

Dr. M. Venu Gopala Rao, Professor, Dept. of ECE

7. IIR Digital Filter Design:

z e sT , where s j and T is sampling interval. By substituting in z , we get

r e T which is magnitude and T which is angular position.

Dr. M. Venu Gopala Rao, Professor, Dept. of ECE

where Ai are the coefficients of partial fraction expansion, and

pi p1, p2 ,..., pN are the poles.

h[n] of digital filter is obtained by sampling the Impulse Response of analog

filter ha (t ) . Thus h[n] ha (t ) |t nT

, where T is sampling time.

Then the transfer function of digital filter is given by

By interchanging the order of summation, we get

to . For a general range (i 1) T to (i 1) T , (where i is an integer) also

maps in the range of

to . Hence mapping from analog frequency to digital frequency is

Dr. M. Venu Gopala Rao, Professor, Dept. of ECE

Compute the Z.T. of the digital filter by using the formula: H [ z ]

% Assume sampling frequency = 1 Hz.

% Transformation from analog to digital using impulse invariance method.

Dr. M. Venu Gopala Rao, Professor, Dept. of ECE

Op = 2pi4000; Os = 2pi6000; rp = 1; rs = 40;