Chapter 7

IIR Filter Design

Content

Preliminaries
Characteristics of Prototype Analog Filters
Analog-to-Digital Filter Transformations
Frequency Transformations
 Preliminaries

How to design a digital filter

The design of a digital filter is carried out in three steps:
First: Specifications
Before we can design a filter, we must have some
specifications. These specifications are determined
by the applications.
Second: Approximations
Once the specifications are defined, we use various
concepts and mathematics to come up with a filter
description that approximates the given set of
specifications. This step is the topic of filter design.
 Preliminaries

Third: Implementation
The product of the above step is a filter description
in the form of either a difference equation, or a
system function, or an impulse response. From this
description we implement the filter in hardware or
software on a computer.

In this and the next chapter we will discuss in detail

only the second step, which is the conversion of
specification into a filter description.
 Preliminaries

The specifications
In many applications, digital filters are used to implement
frequency-selective operations;
Therefore, specifications are required in the frequency-
domain in terms of the desired magnitude and phase
response of the filter;
Generally a linear phase response in the passband is
desirable;
An FIR filter is possible to have an exact linear phase;
An IIR filter is impossible to have linear phase in
passband. Hence we will consider magnitude-only
specifications.
 Preliminaries

There are two ways to give the magnitude specifications

Absolute specifications
Provide a set of requirements on the magnitude response
function H (e j ) and generally used for FIR filters.
1 a1 H (e j ) 1 | | p [0, p ] Passband

j
[ s , ] Stopband
H (e ) a2 s | |
[ p , s ] Transition band
a1 The tolerance (or ripple) in passband
a 2 The tolerance (or ripple) in stopband
p The ending frequency of the passband. Bandwidth
s The beginning frequency of the stopband.
 Preliminaries

Relative specifications (dB)

Provide requirements in decibels (dB). This approach is
the most popular one in practice and used for both FIR
and IIR filters
H (e j 0 ) j p
1 20 lg j p
20 lg H (e ) 20 lg(1 a1 )
H (e )
H (e j 0 )
2 20 lg 20 lg H (e j s ) 20 lg a 2
H ( e j s )

1 The maximum tolerable passband ripple R p

2 The minimum tolerable stopband attenuation As
 Preliminaries

Examples
In a certain filters specifications the passband
ripple is 0.25dB, and the stopband attenuation is
50dB. Determine the a1 and a2.

0.25
( )
1 0.25 20 lg(1 a1 ), a1 1 10 20
0.0284
50
( )
2 50 20 lg a 2 , a 2 10 20
0.0032
 Preliminaries

Given the passband tolerance a1=0.02 and the

stopband tolerance a2=0.001, determine the
passband ripple 1 and the stopband attenuation 2

1 20 lg(1 a1 ) 20 lg(1 0.02) 0.1755dB

2 20 lg a 2 20 lg 0.001 60dB
 Preliminaries

The basic technique of IIR filter design

IIR filters have infinite-length impulse responses,
hence they can be matched to analog filters.
Analog filter design is a mature and well
developed field.
We can begin the design of a digital filter in the
analog domain and then convert the design into
the digital domain
 Preliminaries

There are two approaches to this basic technique

Approach 1
Designed
IIR filter
Apply freq. band Apply filter
Design analog
transformation transformation
lowpass filter
ss sz

Approach 2
Designed
IIR filter
Apply filter Apply freq. band
Design analog
transformation transformation
lowpass filter
sz zz

return
 Characteristics of Prototype Analog Filters

Magnitude-squared function
Let H a ( j ) be the frequency response of an analog
filter
1 2 2 1
H a ( j ) 1, | | p H a ( j ) at p
1 2
1 2

2 1 2 1
0 H a ( j ) 2 , | | s H a ( j ) 2 at s
A A
is a passband ripple parameter
A is a stopband attenuation parameter
p is the passband cutoff frequency in rad/sec
s is the stopband cutoff frequency in rad/sec
 Characteristics of Prototype Analog Filters

2
The properties of H a ( j )

H a ( j ) H a ( s ) s j ha (t ) is a real function
2
H a ( j ) H a ( j ) H a ( j ) H a ( j ) H a ( j ) H a ( s ) H a ( s ) s j

2
H a ( s ) H a ( s ) H a ( j )
2 s 2

The poles and zeros of H a ( s ) H a ( s ) are distributed in

a mirror-image symmetry with respect to the j axis.
For real filters, poles and zeros occur in complex
conjugate pairs.
 Characteristics of Prototype Analog Filters

H a ( s)H a ( s) j
s-plane

2
 Characteristics of Prototype Analog Filters

How to construct H a (s )
H a (s ) is the system function of the analog filter. It must be
causal and stable. Then all poles of H a (s ) must lie within
the left half-plane.
All left-half poles of H a ( s ) H a ( s ) should be assigned to H a (s )

Zeros are not uniquely determined. They can be halved

between H a (s ) and H a ( s ) . (Zeros in each half must occur
in complex conjugate pairs)
If a minimum-phase filter is required, the left-half zeros
should be assigned to H a (s )
 Examples 16( 25 2 ) 2
2
H a ( j )
(49 2 )( 36 2 )
2 16( 25 s 2 ) 2
H a ( s ) H a ( s ) H a ( j )
2 s2 (49 s 2 )( 36 s 2 )

poles s 7, s 6 2th order zeros s j 5

We can assign left-half poles s 7, s 6 and a pair

of conjugate zeros s j 5 to H a (s )

K 0 ( s 2 25) H a ( s ) s 0 H a ( j ) 0
H a ( s)
( s 7 )( s 6) K0 4

4( s 2 25) 4 s 2 100
H a ( s) 2
( s 7 )( s 6) s 13 s 42
 Characteristics of Prototype Analog Filters

Butterworth lowpass filters

This filter is characterized by the property that its
magnitude response is flat in both passband and
stopband. The magnitude-squared function of an
Nth-order lowpass filter is given by

2 1
H a ( j ) 2N

1
c
 Characteristics of Prototype Analog Filters

The properties of Butterworth lowpass filters

At 0 , H a ( j ) 1 for all N
At , H ( j ) 1 0.707 for all N, which
c a
2
implies a 3dB attenuation at
c

H a ( j ) is a monotonically decreasing function of

H a ( j ) approaches an ideal lowpass filter as N

H a ( j ) is maximally flat at 0 since derivatives of

all orders exist and are equal to zero
 Characteristics of Prototype Analog Filters

The poles and zeros of H a ( s ) H a ( s )

2 1 ( j c )2 N
H a ( s) H a ( s) H a ( j ) 2N
s / j
s
2N
s ( j c )2 N
1
j c

1 j ( 1 2 k 1 )
sk ( 1) 2N
( j c ) c e 2 2N
, k 1,2, ,2 N
 Characteristics of Prototype Analog Filters

There are 2N poles ofH a ( s ) H a ( s ) , which are

equally distributed on a circle ofradius
c
with angular

spacing
N of radians.
If the N is odd, there are poles on real axis.
If the N is even, there are not poles on real axis.
The poles are symmetrically located with respect to
the imaginary axis.
A pole never falls on the imaginary axis, and falls on
the real axis only if N is odd.
 Characteristics of Prototype Analog Filters

cN
H a ( s) N j ( 1 2 k 1 )
(s s k ) sk c e 2 2N
, k 1,2, , N
k 1

In general, we consider c 1 rad/s and this results in a

normalized Butterworth analog prototype filter H an (s )

When designing an actual filter H a (s ) with c 1 rad/s , we

can simply do a replacement for s, that is

s
H a ( s ) H an ( )
c
 Designing equations
Given p , s , 1 , 2 , two parameters are required to
determine a Butterworth lowpass filters : N , c

2N
p
at p 20 lg 1 1
c

2N
s
at s 20 lg 1 2
c

Solving these two equations for N , c
 1 2

lg (10 1) /(10
10 10
1)

N , N N
p
2 lg( )
s
Since the actual N chosen is larger than required,
specifications can be either met or exceeded at p or s
To satisfy the specifications exactly at p

p s
c c
2N 1 2N 2
10 10
1 10 10
1
To satisfy the specifications exactly at s
 Example
Determine the system function of 3th-order Butterworth
analog lowpass filter. Suppose c 2 rad/s

Solution:

2
H a ( j ) 1 1
H a ( s) H a ( s) 1
2N 6
1
6
s
1 1
c 2 64

j ( 1 2 k 1 )
s k 2e 2 6
, k 1,2, ,6

c3 8
H a ( s) 3
( s s1 )( s s2 )( s s3 ) s 4 s 2 8 s 8
Design the above filter with normalized Butterworth
analog prototype filter. See table 6-4 on page 261

d0
H an ( s ) , a0 a N 1
a0 a1 s a 2 s a N s
2 N

d 0 a0 in case of H a ( j 0) 1

For N 3 We can find a1 2, a2 2 s s s

c 2
H a ( s ) H an ( s ) s s
2

1 8
1 2( s ) 2( s ) 2 ( s ) 3 8 8s 4s 2 s 3
2 2 2
 Design a lowpass Butterworth filter to satisfy:
Passband 1 1 dB for 0 2 104 rad/s
Stopband 2 15 dB for 2 1.5 104 rad/s

Solution: 1 1 dB, p 2 104 rad/s

2 15 dB, s 2 1.5 10 4 rad/s

2N
2 10 4

20 lg 1 1
c

2N
2 1.5 10 4

20 lg 1 15
c

 1 2
1 15

lg (10 1) /(10
10 10
1) lg (10 1) /(10 1)
10 10

N
p 2 104
2 lg( ) 2 lg
s 4
2 1.5 10
5.8858
N 6
s 2 1.5 104
c 2 1.1279 104
2N 2 12 15
10 10
1 10 10
1

c 2 1.13 104 rad/s

 Look for table 6-4 on page 261
a1 3.864, a 2 7.464, a 3 9.142, a4 7.464, a5 3.864

1
H an ( s )
1 a1 s a 2 s a 3 s a4 s a5 s s
2 3 4 5 6

1

1 3.864 s 7.464 s 9.142 s 7.464 s 3.864 s s
2 3 4 5 6

H a ( s ) H an ( s ) s s H an ( s ) s s
c 2 1.13104
1.28 10 29
1.28 10 29 6.97 10 24 s 1.90 10 20 s 2 3.27 1015 s 3 3.76 1010 s 4 2.74 10 5 s 5 s 6
 To construct a cascade structure s1,6 0.259 j 0.966
Look for table 6-6 on page 263 s2 ,5 0.707 j 0.707
s3 ,4 0.966 j 0.259

1
H an ( s )
( s s1 )( s s2 )( s s3 )( s s4 )( s s5 )( s s6 )
1
2
( s 0.52 s 1.00)( s 2 1.41s 1.00)( s 2 1.93 s 1.00)
s s
s
c 2 1.13 10 4
1.28 10 29
H a ( s) 2
( s 3.69 10 4 s 5.04 10 9 )( s 2 10 5 s 5.04 10 9 )( s 2 1.37 10 5 s 5.04 10 9 )
 Characteristics of Prototype Analog Filters

Chebyshev lowpass filters

There are two types of Chebyshev filters
Chebyshev-I: equiripple in the passband and monotonic in
the stopband.
Chebyshev-II: monotonic in the passband and equiripple in
the stopband.
Chebyshev filters can provide lower order than Butterworth
filters for the same specifications.
Chebyshev-I 2 1
H a ( j )

1 2C N2
c
 H a ( j )
2 1

1 2C N2
c
N is the order of the filter
is the passband ripple factor. 0 1
C N2 is the Nth-order Chebyshev polynomial given by

cos( N cos 1 x ), | x | 1
C N ( x) where x
-1
ch( Nch x ), | x | 1 c

C N 1 ( x ) 2 xC N ( x ) C N 1 ( x )
C 0 ( x ) 1, C1 ( x ) x
 The properties of Chebyshev lowpass filters
At 0 H a ( j ) 1
:
H a ( j 0) 1 for N is odd 1 2C N2
c
H a ( j 0) 1 for N is even
1 2

At c H a ( j c ) 1 for all N
: 1 2
For0 c H a ( j) oscillates between 1 ~ 1
: 1 2
For c H a ( j ) decreases monotonica lly to 0
:
For s H a ( j s ) 1
: A
 Designing equations
Given c , s , As, A p , two parameters are required to
determine a Chebyshev-I filter: , N

0.1 A p
10
2
1
1 10 0.1 As
1
ch

N
s
1
ch

c 1 10 1
0.1 As
1
s c ch ch
N

3 dB c ch 1 ch1 1
N Note: this is only for c 3 dB
 Determine system function
To determine a causal and stable H a (s ) , we must find
the poles of H ( s ) H ( s ) and select the left half-plane
a a
poles for H a (s ) . The poles are obtained by finding the
roots of
s
1 C
2 2
0
j c
N

It can be shown that if sk k j k k 1,2, , N

are the (left half-plane) roots of the above polynomial,
then
( 2k 1)
k (a c ) sin
2 N k 1,2, , N
( 2k 1)
k (b c ) cos
2 N
 1 1 1 1
a 1 ( N
N
), b 1 ( N
N
)
2 2

1 12 1

H a ( s) K
N

(s s
k 1
k )

Where K is a normalizing factor chosen to make

1, N is odd
1
H a ( j 0) , N is even
1 2
 Determine poles by geometric method
The poles of H a ( s ) H a ( s ) fall on an ellipse with major axis b
c
and minor axis
a .
c
j
N

a c b c
 Examples
Determine the system function of 2th-order Chebyshev-I
lowpass filter. Suppose c 1 rad/s and Ap 1 dB
Solution: 10
2 0.1 Ap
1 10 1 0.2589
0.1

a0 1.1025
a1 1.0977
d0 d0
H a ( s)
a0 a1 s s 2
1.1025 1.0977 s s 2

H a ( j 0) 1 1 0.8913
1 2
1.2589
d0
H a ( s ) s 0 0.8913, d 0 0.9827
1.1025
 Design a lowpass Chebyshev-I filter to satisfy:
Passband cutoff: c 2 104 rad/s
Passband ripple: Ap 1 dB
Stopsband cutoff: s 2 1.5 10 rad/s
4

Stopband attenuation: As 15 dB

Solution: 0. 1 A p
10 1 10 1 0.5088
0.1

1
100.1 As 1
ch
ch 1 (10.8761)
N 1
3.1978
1 s ch (1.5)
ch

c
Ap 1 dB N 4 0.5088
a0 0.2756a1 0.7426, a 2 1.4539, a 3 0.9528

d0
H an ( s )
a0 a1 s a 2 s 2 a 3 s 3 s 4
d0

0.2756 0.7426 s 1.4539 s 2 0.9528 s 3 s 4

H a ( j 0) 1 1 0.8913
1 2 1.2589

d0
H an ( s ) s 0 0.8913, d 0 0.2456
0.2756
 H an ( s ) 0.2456
0.2756 0.7426 s 1.4539 s 0.9528 s s
2 3 4

s s s
c 2 104

H a ( s)
3.8278 1018
4.2954 1018 1.8420 1014 s 5.7398 109 s 2 5.9866 104 s 3 s 4
 Analog-to-Digital Filter Transformations

Impulse invariance transformation

Definition
To design an IIR filter having a unit sample response h(n)
that is the sampled version of the impulse response of the
analog filter. That is
h( n) ha ( nT )
T : Sampling interval
Since this is a sampling operation, the analog and digital
frequencies are related by
j
T , or e e j T
The system function H (z ) and H a (s ) are related by

1
2
H ( z ) z e sT
T

k
Ha (s j
T
k)

This implies a mapping from the s-plane to the z-plane

j
j Im[z ]
3
T

T
0
Re[z ]

T

3
T
 Analog-to-Digital Filter Transformations

Properties
Using Re[s ]
0 maps into | z | 1 (inside of the UC)
0 maps into | z | 1 (on the UC)
0 maps into | z | 1 (outside of the UC)
All semi-infinite left strips of width 2 / T map into | z | 1 .
Thus this mapping is not unique but a many-to-one mapping
Since the entire left half of the s-plane maps into the unit
circle, a causal and stable analog filter maps into a causal
and stable digital filter.
 Analog-to-Digital Filter Transformations

Aliasing occurs if the filter is not exactly band-limited

1 2k
Frequency response H (e ) H a ( j
j
)
T k T

If H a ( j ) H a ( j ) 0 for | |
T T
1
then H (e j ) H a ( j ), | |
T T
There will be no aliasing.
To minimize the effects of aliasing, the T should be
selected sufficiently small.
If the filter specifications are given in digital frequency
domain, we cannot reduce aliasing by selecting T.
 Analog-to-Digital Filter Transformations

Digitalizing of analog filters

Using partial fraction expansion, expand H a (s ) into
N
Ak
H a ( s)
k 1 s s k

The corresponding impulse response is

N
ha ( t ) L [ H a ( s )] Ak e sk t u( t )
1

k 1

To sample the ha (t )
N N
h( n) ha ( nT ) Ak e sk nT u( n) Ak (e sk T ) n u( n)
k 1 k 1
The z-transform of h(n) is
N N
Ak
H (z) h(n)z
n
n
Ak (e
n 0 k 1
sk T
z )
1 n

k 1 1 e s k T 1
z
N
Ak
Compared with H a ( s )
k 1 s s k

Conclusions:
s T
The pole sk in s-plane is mapped to the pole e k in z-plane
The partial fraction expansion coefficient of H (z ) is the
same as that of H a (s )
The zeros in the two domains do not satisfy the same
relationship
 Analog-to-Digital Filter Transformations

Advantages and disadvantages

The digital filter impulse response is similar to that of a
analog filter. This means we can get a good approximations
in time domain.
It is a stable design and that the frequencies and
are linearly related. So a linear phase analog filter can be
mapped to a linear phase digital filter.
Due to the presence of aliasing, this method is useful only
when the analog filter is essentially band-limited to a
lowpass or bandpass filter in which there are no oscillations
in the stopband.
 Design procedure
Given the digital lowpass filter specifications p , s , 1 , 2
Choose T and determine the analog frequencies
p s
p , s
T T
Design an analog filter H a (s ) using the specifications
p , s , 1, 2
Using partial fraction expansion, expand H a (s ) into
N
Ak
H a ( s)
k 1 s s k

Transform analog poles into digital poles to obtain the

digital filter N
A
H (z) k

k 1 1 e
sk T
z 1
 Analog-to-Digital Filter Transformations

Examples
2 1 1
Transform H ( s )
a
s 4s 3 s 1 s 3
2

into a digital filter H (z ) using the impulse invariance method

in which T=1

T T Tz 1 (e T e 3T )
H (z) 1 T
1 3T

1 z e 1 z e 1 z 1 ( e T e 3T ) z 2 e 4 T
0.3181z 1

1 0.4177 z 1 0.0183 z 2
 Analog-to-Digital Filter Transformations

Bilinear transformation
Definition
This is a conformal mapping that transforms the j -axis
into the unit circle in the z-plane only once, thus avoiding
aliasing of frequency components. This mapping is the
best transformation method.
1T 1T
j j
1T e 2
e 2
1 e j1T
c tan j c 1T 1T
c
2 j j 1 e j1T
e 2
e 2

1 e s1T 1 z 1 cs
s c s1T
c z
1 e 1 z 1 cs
 1 e s1T
s c
1 e s1T z e s1T
j j Im[z ]
s-plane s1-plane j 1 z-plane

0 0 1 0
Re[z ]

1
1 z
s c
1 z 1
 Analog-to-Digital Filter Transformations
Parameter c
Keeping a good corresponding relationship between
the analog filter and the digital filter in low
1 i.e.
frequencies. in low frequencies

1T 1T 2
c tan c , 1 then c
2 2 T
Keeping a good corresponding relationship between the
analog filter and the digital filter in a specific frequency
(for example, in the cutoff frequency, c 1cT )

1c T c c
c c tan c tan then c c cot
2 2 2
 Properties
Using s j , we obtain

c s ( c ) j (c ) 2 2
z , | z |
c s ( c ) j (c ) 2 2
So 0 | z | 1, 0 | z | 1, 0 | z | 1
Using z e j , we obtain

1 z 1 1 e j
s c 1
c j
jc tan( ) j
1 z 1 e 2
The imaginary axis maps onto the unit circle in a one-to-one
fashion. Hence there is no aliasing in the frequency domain.
The entire left half-plane maps into the inside of the unit
circle. Hence this is a stable transformation.
 Analog-to-Digital Filter Transformations

Advantages and disadvantages

It is a stable design;
There is no aliasing;
There is no restriction on the type of filter that can
be transformed;.
The frequencies and are not linearly related.
So a linear phase analog filter cannot be mapped
to a linear phase digital filter.
 Design procedure
Given the digital lowpass filter specifications p , s , 1 , 2
Choose a value for T. We may set T=1
Prewarp the cutoff frequencies p and s ; that is

2 p 2 s
p tan( ), s tan( )
T 2 T 2

Design an analog filter H a (s ) to meet the specifications

p , s , 1, 2
2 1 z 1
Finally, set H ( z ) H a ( 1
)
T 1 z
1
and simplify to obtain H (z ) as a rational function in z
 Analog-to-Digital Filter Transformations

Examples
2
Transform H a ( s ) 2
s 4s 3
into a digital filter using the bilinear transformation.
Choose T=1
2 1 z 1 1 z 1
H (z) Ha ( ) H a (2 )
T 1 z 1 T 1
1 z 1

2 0.13 0.27 z 1 0.13 z 2

1 2
1 z 1

2
1 z
1 1 0.13 z 0.07 z
2 1
4 2 1
3
1 z 1 z
 Design the digital Chebyshev-I filter using bilinear
transformation. The specifications are:
p 0.2 , 1 1dB
s 0.3 , 2 15dB
Solution
Let T=1
Prewarp the cutoff frequencies

2 p
p tan( ) 2 tan(0.1 ) 0.6498
T 2
2 s
s tan( ) 2 tan(0.15 ) 1.0191
T 2
 Design an analog Chebyshev-I filter H a (s ) to meet the
specifications p , s , 1 , 2

0.0438
H a ( s) 4
s 0.6192 s 3 0.6140 s 2 0.2038 s 0.0492

0.0018 0.0073 z 1 0.0110 z 2 0.0073 z 3 0.0018 z 4

H (z)
1 3.0543 z 1 3.8290 z 2 2.2925 z 3 0.5507 z 4
0.0018(1 z 1 )4

(1 1.4996 z 1 0.8482 z 2 )(1 1.5548 z 1 0.6493 z 2 )

 Analog-to-Digital Filter Transformations

Comparison of three filters

p 0.2 , 1 1dB
Given the digital filter specifications:
s 0.3 , 2 15dB
Using different prototype analog filters will give out different
N and the minimum stopband attenuations.

prototype Order N Stopband Att.

Butterworth 6 15 dB
Chebyshev-I 4 25 dB
Elliptic 3 27 dB

return
 Frequency Transformations

Introduction
The treatment in the preceding section is focused
primarily on the design of digital lowpass IIR filters. If we
wish to design a highpass or a bandpass or a bandstop
filter, it is a simple matter to take a lowpass prototype
filter and perform a frequency transformation.
There are two approaches to perform the frequency
transformation
Frequency transformations in the analog domain
Frequency transformations in the digital domain
 Frequency Transformations

Approach 1
Designed
IIR filter
Frequency Filter
Analog
transformation transformation
lowpass filter
ss sz

Approach 2
Designed
IIR filter
Filter Frequency
Analog
transformation transformation
lowpass filter
sz zz
 Frequency Transformations

Specifications of frequency-selective filters

Lowpass filter p , s , 1, 2

highpass filter s , p , 1, 2

bandpass filter s 1 , p1 , p 2 , s 2 , 1 , 2

bandstop filter p1 , s 1 , s 2 , p 2 , 1 , 2
 Frequency Transformations

Frequency transformations in the digital domain

H L (z ) the given prototype lowpass digital filter

H d (Z ) the desired frequency-selective digital filter

1 1
Define a mapping of the form z G ( Z )

Such that H d ( Z ) H L ( z ) z 1 G ( Z 1 )
1
To do this, we simply replace z everywhere in H L (z )
by the function G ( Z 1 )
 Frequency Transformations

Given that H L (z ) is a stable and causal filter, we also

want H d (Z ) to be stable and causal. This imposes the
following requirements:

The unit circle of the z-plane must map onto the unit
circle of the Z-plane
The inside of the unit circle of the z-plane must also
map onto the inside of the unit circle of the Z-plane.
G ( Z 1 ) must be a rational function in Z 1 so that H d (Z )
is implementable.
 Frequency Transformations

Let and be the frequency variables of z and Z ,

respectively. That is z e j , Z e j . Then
j j j j arg[G ( e j )]
e G (e ) G (e )e
G (e j ) 1, arg[G (e j )]
1
Hence the G ( Z ) is an all-pass function
1
N
Z a
z 1 G ( Z 1 ) k
1
, | a k | 1
k 1 1 a k Z

By choosing an appropriate order N and the

coefficients a k , we can obtain a variety of mappings
 Frequency Transformations

Frequency transformation formulae

c The cutoff frequency of prototype lowpass digital filter
Lowpass - Lowpass

Z 1
z 1
1 Z 1
c : Cutoff frequency of new digital filter
sin[( c c ) / 2]

sin[( c c ) / 2]
 Frequency Transformations

Lowpass - Highpass

Z 1
z 1
1 Z 1

c : Cutoff frequency of new digital filter

cos[( c c ) / 2]

cos[( c c ) / 2]
 Frequency Transformations

Lowpass - Bandpass

Z 2 1 Z 1 2
z 1
2 Z 2 1 Z 1 1
1 : lower cutoff frequency of bandpass digital filter
2 : upper cutoff frequency of bandpass digital filter
0 : center frequency of the passband
cos[( 2 1 ) / 2] 2 1 c
cos 0 , k cot( ) tan
cos[( 2 1 ) / 2] 2 2
2 k k 1
1 , 2
k 1 k 1
 Frequency Transformations

Lowpass - Bandstop

Z 2 1 Z 1 2
z 1
2 Z 2 1 Z 1 1
1 : lower cutoff frequency of bandstop digital filter
2 : upper cutoff frequency of bandstop digital filter
0 : center frequency of the stopband
cos[( 2 1 ) / 2] 2 1 c
cos 0 , k tan( ) tan
cos[( 2 1 ) / 2] 2 2
2 1 k
1 , 2
1 k 1 k
 Frequency Transformations

Design procedure
Determine the specifications of the digital prototype
lowpass filter;
Determine the specifications of the analog prototype
lowpass filter;
Design the analog prototype lowpass filter;
Transform the analog prototype lowpass filter into a digital
prototype lowpass filter using bilinear transformation;
Perform the frequency transformation in digital domain to
obtain the desired frequency-selective filters.
 Frequency Transformations

Examples
Given the specifications of Chebyshev-I lowpass filter
p 0.2 , 1 1dB
s 0.3 , 2 15dB
and its system function
0.01836(1 z 1 )4
H L (z)
(1 1.4996 z 1 0.8482 z 2 )(1 1.5548 z 1 0.6493 z 2 )

Design a highpass filter with the above tolerances but

with passband beginning at p 0.6
 Frequency Transformations

Solution

cos[(0.2 0.6 ) / 2]
0.3820
cos[(0.2 0.6 ) / 2]

H d ( Z ) H L ( z ) z 1 Z 1 0.3820
1 0.3820 Z 1

0.0243(1 Z 1 )4

(1 0.5561 Z 1 0.7647 Z 2 )(1 1.0416 Z 1 0.4019 Z 2 )

 Frequency Transformations

Using the Chebyshev-I prototype to design a

p 0.6 ,
highpass digital filter to satisfy R p 1dB
Solution s 0.46 , As 15dB
Determine the specifications of the digital prototype
lowpass filter
Choose the passband frequency with a reasonable value:
p 0.2
Z 1
Determine the stopband frequency by z 1
1 Z 1
e j e j
e j j
arg( j
)
1 e 1 e
 cos[( p p ) / 2] cos[(0.2 0.6 ) / 2]
0.3820
cos[( p p ) / 2] cos[(0.2 0.6 ) / 2]

e j s e j 0.46 0.3820
s arg( j s
) arg( j 0.46
) 0.3
1 e 1 0.3820e

Determine the specifications of the analog prototype

lowpass filter
Set T = 1 and prewarp the cutoff frequencies

2 p
p tan( ) 2 tan(0.1 ) 0.6498
T 2
2 s
s tan( ) 2 tan(0.15 ) 1.0191
T 2
 Design an analog Chebyshev-I prototype lowpass filter
to satisfy the specification: p , s , R p , As

0.0438
H a ( s) 4
s 0.6192 s 3 0.6140 s 2 0.2038 s 0.0492

Transform the analog prototype lowpass filter into a

digital prototype lowpass filter using bilinear transformation

0.0018 0.0073 z 1 0.0110 z 2 0.0073 z 3 0.0018 z 4

H L (z)
1 3.0543 z 1 3.8290 z 2 2.2925 z 3 0.5507 z 4
0.0018(1 z 1 )4

(1 1.4996 z 1 0.8482 z 2 )(1 1.5548 z 1 0.6493 z 2 )

 Perform the frequency transformation in digital domain
to obtain the desired digital highpass filter

H h ( Z ) H L ( z ) z 1 Z 1
1Z 1

0.0243(1 Z 1 )4

(1 0.5561 Z 1 0.7647 Z 2 )(1 1.0416 Z 1 0.4019 Z 2 )

 Frequency Transformations

Using the Chebyshev-I prototype to design a

bandpass digital filter to satisfy
p1 0.4 , p 2 0.5 , R p 1dB
s1 0.2 , s 2 0.7 , As 15dB
Solution

Determine the specifications of the digital prototype

lowpass filter

Choose the passband frequency with a reasonable value:

p 0.2
Determine the stopband frequency by Z 2 1 Z 1 2
z 1
2 Z 2 1 Z 1 1

cos[( p 2 p1 ) / 2]cos[(0.5 0.4 ) / 2]

0.1584
cos[( p 2 p1 ) / 2] cos[(0.5 0.4 ) / 2]

0.5 0.4 0.2

k cot( ) tan 2.0515
2 2
2 k k 1
1 0.2130, 2 0.3446
k 1 k 1
j 2 s 2 j s 2
e 1e 2
s arg( j 2 s 2 j s 2
) 0.69
2e 1e 1
 Determine the specifications of the analog prototype
lowpass filter
Set T = 1 and prewarp the cutoff frequencies

2 p
p tan( ) 2 tan(0.1 ) 0.6498
T 2
2 s
s tan( ) 2 tan( 0.3450 ) 3.7842
T 2
Design an analog Chebyshev-I prototype lowpass filter
to satisfy the specification: p , s , R p , As

0.4149
H a ( s) 2
s 0.7134 s 0.4656
 Transform the analog prototype lowpass filter into a
digital prototype lowpass filter using bilinear transformation

0.0704 (1 z 1 ) 2
H L (z)
1 1.1997 z 1 0.5157 z 2

Perform the frequency transformation in digital domain

to obtain the desired digital bandpass filter

H bp ( Z ) H L ( z ) z 1 Z 2 1 Z 1 2
2 Z 2 1 Z 1 1

0.0205 0.0410 Z 2 0.0205 Z 4

1 2 3 4
1 0.5731 Z 1.7020 Z 0.4814 Z 0.7106 Z
 Frequency Transformations

Using the Chebyshev-I prototype to design a bandstop

digital filter to satisfy
p1 0.25 , p 2 0.75 , R p 1 dB
s1 0.35 , s 2 0.65 , As 20 dB
Solution

Determine the specifications of the digital prototype

lowpass filter

Choose the passband frequency with a reasonable value:

p 0.2
Determine the stopband frequency by Z 2 1 Z 1 2
z 1
2 Z 2 1 Z 1 1

cos[( p 2 p1 ) / 2]cos[(0.75 0.25 ) / 2]

0
cos[( p 2 p1 ) / 2] cos[(0.75 0.25 ) / 2]

0.75 0.25 0.2

k tan( ) tan 0.1584
2 2
2 1 k
1 0, 2 0.7265
1 k 1 k
j 2 s 1 j s 1
e 1e 2
s arg( j 2 s 1 j s 1
) 0.1919
2e 1e 1
 Determine the specifications of the analog prototype
lowpass filter
Set T = 1 and prewarp the cutoff frequencies

2 p
p tan( ) 2 tan(0.1 ) 0.3168
T 2
2 s
s tan( ) 2 tan( 0.0959 ) 0.6217
T 2
Design an analog Chebyshev-I prototype lowpass filter
to satisfy the specification: p , s , R p , As

0.0156
H a ( s) 3
s 0.3131s 2 0.1243 s 0.0156
 Transform the analog prototype lowpass filter into a
digital prototype lowpass filter using bilinear transformation

0.0016 0.0049 z 1 0.0049 z 2 0.0016 z 3

H L (z)
1 2.6225 z 1 2.3692 z 2 0.7335 z 3

Perform the frequency transformation in digital domain

to obtain the desired digital bandstop filter

H bs ( Z ) H L ( z ) z 1 Z 2 1 Z 1 2
2 Z 2 1 Z 1 1

0.132 (1 Z 2 ) 3

(1 1.248 Z 1 0.776 Z 2 )(1 1.248 Z 1 0.776 Z 2 )(1 0.339 Z 2 )

return
H a ( j ) Magnitude Response

N=2
1 N=4
N=8
N=16

0.707

0 30 40 50 60
Analog frequency in rad/s
return
H a ( j ) Magnitude Response

1
Amplitude

0.707

0 2 4 6
Analog frequency in rad/s
Phase Response
3
Phase in rad

1
0
-1

-3
0 2 4 6
Analog frequency in rad/s
return
 Magnitude Response Magnitude in dB

1
0
-1
0.8913

decibels
H

-15

0.1778

0 -30
0 2 3 5 0 2 3 5
Analog frequency in pi units 4 Analog frequency in pi units 4
x 10 x 10
Phase Response Impulse Response
1
20000
0.5
ha(t)
0 10000
P

-0.5
0

-1
0 2 3 5 0 0.5 1 1.5
Analog frequency in pi units 4 time in seconds -4
x 10 x 10
return
 Magnitude Response Magnitude in dB

1
0
-1
0.8913

decibels
H

-15

0.1778

0 -30
0 2 3 5 0 2 3 5
Analog frequency in pi units 4 Analog frequency in pi units 4
x 10 x 10
Phase Response Impulse Response
1
20000
0.5
15000
ha(t)
0 10000
P

5000
-0.5
0

-1 -5000
0 2 3 5 0 0.5 1 1.5
Analog frequency in pi units 4 time in seconds -4
x 10 x 10
return
C0 ( x )

-1

-2

-1 -0.8 -0.6 -0.4 -0.2 0 0.2 0.4 0.6 0.8 1

C2 ( x)

-1

-2

-1 -0.8 -0.6 -0.4 -0.2 0 0.2 0.4 0.6 0.8 1

C3 ( x)

-1

-2

-1 -0.8 -0.6 -0.4 -0.2 0 0.2 0.4 0.6 0.8 1

C4 ( x )

-1

-2

-1 -0.8 -0.6 -0.4 -0.2 0 0.2 0.4 0.6 0.8 1

C5 ( x )

-1

-2

-1 -0.8 -0.6 -0.4 -0.2 0 0.2 0.4 0.6 0.8 1

C6 ( x)

-1

-2

-1 -0.8 -0.6 -0.4 -0.2 0 0.2 0.4 0.6 0.8 1

3

-1

-2

-1 -0.8 -0.6 -0.4 -0.2 0 0.2 0.4 0.6 0.8 1

return
 Magnitude Response N 4

0.8913
Amplitude

0 2 3 5
Analog frequency in rad/s
 Magnitude Response
N 5

0.8913
Amplitude

0 2 3 5
Analog frequency in rad/s
return
H a ( j ) Magnitude Response

1
0.8913
Amplitude

0 1 2 3 5
Analog frequency in rad/s
Phase Response
1

0
Phase in rad

-1

-2

-3
0 1 2 3 5
Analog frequency in rad/s
return
 Magnitude Response Magnitude in dB

1
0
-1
0.8913

decibels
H

-15

0.1778

0 -30
0 2 3 5 0 2 3 5
Analog frequency in pi units 4 Analog frequency in pi units 4
x 10 x 10
Phase Response Impulse Response
1 20000

15000
Phase in pi units

0.5
10000
ha(t)
0
5000
-0.5 0

-1 -5000
0 2 3 5 0 1 2 3 4
Analog frequency in pi units 4 time in seconds -4
x 10 x 10
return
H a ( j ) T 1
0.6

0.4

0.2

0 pi/T 2*pi/T

H ( e j )

0.6

0.4

0.2

0 pi 2*pi

H a ( j ) T 0.1
0.6

0.4

0.2

0 pi/T 2*pi/T

H ( e j )

0.6

0.4

0.2

0 pi 2*pi

return
 Magnitude Response Phase Response
1
1
0.8913 0.5

-0.5
0.1778
-1
0 0.2 0.3 1 0 0.2 0.3 1
Frequency in pi Frequency in pi
Magnitude Response in dB Group Delay
10
0
-1 8

-15 4

0 0.2 0.3 1 0 0.2 0.3 1

Frequency in pi Frequency in pi
return
H a ( j ) Magnitude Response T 1
0.6

0.4

0.2

0 pi/T 2*pi/T

frequency in rad/s
H ( e j )

0.6

0.4

0.2

0 pi 2*pi

frequency in rad/sample
H a ( j ) Magnitude Response T 0.1
0.6

0.4

0.2

0 pi/T 2*pi/T

frequency in rad/s
H ( e j )

0.6

0.4

0.2

0 pi 2*pi

frequency in rad/sample
return
 Magnitude Response Phase Response
1
1
0.8913 0.5

pi units
0

-0.5
0.1778
-1
0 0.2 0.3 1 0 0.2 0.3 1
Frequency in pi units Frequency in pi
Magnitude Response in dB Group Delay
15
0
-1 12

samples
9

-15 6

0 0.2 0.3 1 0 0.2 0.3 1

Frequency in pi units Frequency in pi units
return
 Magnitude Response Magnitude Response

1 1
0.8913 0.8913

0.1778 0.1778

0 0.2 0.3 1 0 0.6 1

Frequency in pi units Frequency in pi units
Magnitude Response in dB Magnitude Response in dB

0
-1 0
-1

-15 -15

0 0.2 0.3 1 0 0.6 1

Frequency in pi units Frequency in pi units
return
 Magnitude Response Phase Response
1
1
0.8913 0.5

pi units
0

-0.5
0.1778
-1
0 0.46 0.6 1 0 0.46 0.6 1
Frequency in pi units Frequency in pi
Magnitude Response in dB Group Delay
10
0
-1 8

samples
6

-15 4

0 0.46 0.6 1 0 0.46 0.6 1

Frequency in pi units Frequency in pi units
return
 Magnitude Response Phase Response
1
1
0.8913
0.5

pi units
0

-0.5
0.1778
-1
0 0.2 0.4 0.5 0.7 1 0 0.2 0.4 0.5 0.7 1
Frequency in pi units Frequency in pi
Magnitude Response in dB Group Delay
15
0
12
-20

samples
-40 9

-60 6

-80 3

-100
0 0.2 0.4 0.5 0.7 1 0 0.2 0.4 0.5 0.7 1
Frequency in pi units Frequency in pi units
return
 Magnitude Response Phase Response
1
1
0.8913
0.5

pi units
0

-0.5

0.1
-1
0 0.250.35 0.650.75 1 0 0.250.35 0.650.75 1
Frequency in pi units Frequency in pi
Magnitude Response in dB Group Delay
10
0
8
-10

samples
6
-20 4

-30 2

0 0.250.35 0.650.75 1 0 0.250.35 0.650.75 1

Frequency in pi units Frequency in pi units
return