Discrete-Time IIR Filter Design from

Continuous-Time Filters

Inderdeep Kaur Aulakh

Asst. Prof. (IT),UIET
PU, CHD
Analog Filters- Parameters and Classification

Parameters
 Passband: Range of frequencies for which attenuation is zero
 Stopband: Range of frequencies for which attenuation is
infinity
 Cut-off Frrequency: Frequency which separates the stopband
and passband
Classification
 Low-pass Filter
 High pass Filter
 Band Pass Filter
 Band Reject Filter
 All Pass Filter
 Analog Vs Digital Filters

Implementation
 Analog Filters: Analog components –Resistors,
Inductors & Capacitors
 Digital Filters: Described by difference equations
and implemented with S/W like C or Assembly
language, MATLAB
Classification of Digital Filters
 Infinite Impulse Response Filters
 Finite Impulse Response Filters
Advantages of Digital Filters
 Many I/P signals can be filtered by one digital filter
without replacing the h/w.
 Linear Phase response in FIRs
 Environment( temperature, humidity) doesn’t vary
performance
 Filtering is using computers so filtered and unfiltered
data can be saved for future use.
 Portable
 Highly Flexible
 VLSI technology can reduce H/W and power
consumption
 Use at very low frequencies as in Biomedical applications
 Maintenance not required frequently as in analog filters
Digital Filters: Advantages and Disadvantages

Disadvantages
 Speed Limitation:
 Use of ADC and DAC reduces speed of filter, depending
on conversion time of ADC and settling time of DAC
 Speed of Digital Processor controls digital filtering
operation
 Finite Word length : Accuracy of digital filtering depends
on
 Word length for binary encoding
 ADC noise, due to quantization of continuous signals
 Round off noise during computation
 Design and development time for digital H/W is more
 Ideal Low Pass Filter
Ideal Discrete LPF are not realizable as:
 They have constant gain in passband and zero gain in stopband
 They have Linear Phase Response

H     1,   c
H     0, c    

1
 h k e and h n 

H     jk
  H  e
j n
d
k   2 

Sincn
h n  ,n  0
n Ideal filter characteristics are non-causal
c hence unrealizable.
h n  ,n  0

Filter Specifications- Realizable Digital LPF

 Specifications
 Passband

1  1  H     1   2 0     p
 Stopband

 2  H     1  1  p     s

H      2 s    

 Parameters
Peak deviation in passband
1
 2Peak deviation in stopband

 ppassband edge frequency

sstopband edge frequency
P
 Analog Filters: Butterworth Lowpass Filters
 Passband is designed to be maximally flat
 The magnitude-squared function is of the form
1
H c  j  
2

1   j / j c 
2N

1
Hc  s
2

1   s / j c 
2N

sn    1  j c 
1/ 2 N

 j c e  j  2 n 1  / 2 N 
for n  0,1,..., N - 1
N poles on a circle of radius
  c  Cutoff  frequency
Chebyshev Filters
 Equiripple in the passband and monotonic in the stopband
 Or equiripple in the stopband and monotonic in the passband

Hc  j 
2

1
1   V   / c 
2 2

VN  x   cos N cos 1 x 
N
 VN(x)=N order chebyshev polynomial.
IIR Digital Filter design

Analog IIR Filter designed first then converted to Digital Filter

 Procedure for Analog Filter design is readily available and highly
advanced
 Implementation is easier if digital filter is designed using analog
filter
Methods
 Impulse Invariance
 Bilinear Transformation
Impulse Invariance
h(t)=Impulse Response of Analog Filter to be designed
H(s)=Transfer Function of Analog Filter to be designed
h(nt)=Sampled version of h(t)= h(nTs)
H(z)=Transfer Function of h(nTs)
 The Digital Filter’s Impulse response must resemble h(nTs) for
Impulse Invariance, then the two filters will perform similarly.
 IIR Filter Design by Impulse Invariance

Sampled Impulse response is h n  h nTs 


Analog Domain Laplace Transform is H  s   h (t )e  st

n 
Laplace of sampled h(t) is
H  s   H  s    h nTs e  snTs
*

n 0
Z Transform of h(n) is n 
H  z    sTs
 h
n0
n z n
; z  e

s   j  z plane to s plane mapping

z  e sTs  eTs e jTs 1.=0,  r=1, so j axis of s plane maps to unit circle
2.0,  0 r 1,  LHS of s plane maps to interior of
z  re j unit circle
3.0,  r 1,  RHS of s plane maps to exterior of
r  eTs ,   Ts unit circle
Impulse Invariance of System Functions

 Develop impulse invariance relation between system functions

 Partial fraction expansion of transfer function of analog filters
N
Ak
H c  s  
 Corresponding impulse response k 1 s  ak
N
h t    Ak e ak t u  t 
k 1

 Impulse response of discrete-time filter

h n  h nT    Ak e   u n
N N
u n  Ak e ak T n
ak nT

k 1 k 1
 System function
N
Ak
H  z   ak T 1
k 1 1  e z
ak T
 Pole s=ak in s-domain transform into pole at
ze
Examples

1. Determine H(z) using Impulse Invariance method

at 5 Hz sampling frequency from Ha(s)

2
H a  s 
Solution
 s  1 s  2 0.29 z
H  z  2
z  1.488 z  0.54

2. Determine H(z) using Impulse Invariance method

with T=1s from Ha(s)

1 Solution
H a  s 
 s  1 s  2
0.2326 z 1
H  z 
1  0.5032 z 1  0.0498 z  2
Examples

3. Determine H(z) using Impulse Invariance method from Ha(s)

H a  s 
 s  2
 s  1 s  3

Solution

1 1
H  z   1 / 2[ T 1
 3T 1
]
1 e z 1 e z
Disadvantages Impulse Invariance Method

 Mapping is many to one,

 Analog filters not band-limited so aliasing occurs due to
sampling
 Frequency response of digital filter is not identical to that of
Analog Filter
 Change in Sampling time has no effect on amount of aliasing

z plane to s plane mapping

1.=0,  r=1, so j axis of s plane maps to unit circle
2.0,  0 r 1,  LHS of s plane maps to interior of
unit circle
3.0,  r 1,  RHS of s plane maps to exterior of
unit circle
 IIR Filter Design by Bilinear Transformation

 Bilinear transformation
2  1  z 1 
s  
1 
 Transformed system function
T  1 z 

2  1  z 1 
H  z  Hc   
1  
 Again T cancels out so we can ignore it  T  1  z 
 We can solve the transformation for z as

j 1   T / 2  s 1  T / 2  jT / 2
Which yields ,comparing real andimaginary parts
z  re  s    j
1   T / 2 s 1  T / 2  jT / 2


2  r 2 1  2    T 
    tan  or   2 tan 1  d 
 1  r  2r cos  
2
Td Td 2  2 
z plane to s plane Mapping

2  r 2 1  2    T 
    tan  or   2 tan 1  d 
 1  r  2r cos  
2
Td Td 2  2 

1. =0,  r=1, so j axis of s plane maps to unit circle

2. 0,  0 r 1,  LHS of s plane maps to interior of unit
circle
3. 0,  r 1,  RHS of s plane maps to exterior of unit
circle
4. One to one mapping
5. Nonlinear Frequency transformation
6. Frequency response subject to warping
Bilinear Transformation
Examples
1.Design a digital filter using BZT for resonant frequency of

r   / 4
Analog filter System Function is
s  0 .1
H a  s 
 s  0.1 2  9
Solution

1  0.027 z 1  0.973z 2
H  z 
8.572  11 .84 z 1  8.177 z  2
 Digital Frequency Transformation
 In original z plane  and in new Z plane  is frequency variable
 Unit circle in one plane has to map into unit circle in other plane
 Inside of unit circle in z plane must map into inside of unit circle in Z
plane.
LPF to LPF
Z a1 sin   2 
z 1  G ( Z 1 )  a
1  aZ 1 sin   2 
LPF to HPF
z 1 1
 G(Z ) 

 Z 1  a  a
 cos   2 
1  aZ 1 cos   2 

LPF to BPF 2 2ak 1 k  1

Z  Z 
 k 1 sin  2 21 
  
1 1 k 1
k  cot  2 21  tan  2 
z  G(Z )  
k  1  2 2ak 1 a  2 1  , 1   2
Z  Z 1 sin 2
k 1 k 1
LPF to BRF 2a 1 1  k
Z 2 
k 1
Z 
1 k cos  2 21 

k  tan  2 1 
2
tan 
 
2
z 1  G ( Z 1 )  
1  k 2
Z 
2a 1
Z 1
a
cos  2 1 
2
, 1   2
1 k 1 k
Thank you