IR Filter Part 2

Figure: *
Digital Signal Processing: EC253

Dr. V. Prakash Singh
Department of Electronics Engineering

National Institute ofTechnology Andhra Pradesh
Tadepalligudem,Andhra Pradesh
India
Feb 04 2023
IIR Filter Design Convolution and Correlation
Outline of the Lecture
IIR Filter Design

Analog to Digital Transformation
Impulse Invariant Transformation
Bilinear Transformation
Examples
Problems
Convolution and Correlation

Examples
Problems
Department of ElectronicsEngineering National Institute of Technology Andhra Pradesh, India

Outline
1 IIR Filter Design
2 Convolution and Correlation
Department of Electronics Engineering National Institute of Technology Surat Surat, Gujrat India
IIR Filter design techniques
Two approaches for digital IIR filter design
Figure: Digital IIR filter design techniques

Analog to Digital Transformation
The design of a digital filter from a prototype analog filter requires

transformation from Ha (s) to H(z).
We requie a mapping function s = d(z) from the s-plane to the

z-plane such that

H(z) = Ha (s)
s=d(z)
For an acceptable digital filter, this mapping must satisfy the

following properties.
The mapping from jΩ axis in s-plane to the unit circle in z-plane
should be one to one.
Points in the left half s-plane should map to inside of the unit circle
in z-plane.
The mapping should be a rational function of z.
Department of ElectronicsEngineering National Institute of Technology Andhra Pradesh India

The key ides of impulse invariance transform (IIT) is that impulse

response h(n) of the discrete time filter is similar in some sense to
the impulse response ha (t) of the continuous-time filter.
It means h(n) = ha (nTs ) where Ts is the sampling interval.
The frequency response of discrete time filter is given as

1 X j2πk
H(e jw ) = H jΩ +
Ts Ts
k
Since it is a sampling operation, analog and digital frequencies are

linearly related as w = ΩTs .

Mapping from s-plane to z-plane is given by z = e sTs
Figure: s-plane to z-plane mapping

If the continuous-time filter is band limited i.e.

Ha (jΩ) = 0 for |Ω| > Ω0 > Tπs
Discrete-time filter frequency response will closely aproximate the

continuous-time filter frequency response.
H(e jw ) = T1s Ha (jw /Ts ) for |w | < π.
However in practice, Ha (jΩ) is hardly ever band limited, so some

amount of aliasing is inevitable in this design procedure.
The advantage of impulse invariant transformation is the simplicity

of the design procedure.

Design Procedure of Digital Filter using IIT

Given a digital filter specifications wp , ws , Rp , As, we wish to
determine H(z) by first designing an equivalent analog filter Ha (s)
and then mapping it into the desired digital filter.
Chose Ts , determine the analog frequencies.
wp ws
Ωp = and Ωs =
Ts Ts
Design an analog filter Ha (s) using the specifications Ωp , Ωs , Rp , As .
Using partial fraction expansion, expand Ha (s) into
N
X Ak
Ha (s) =
s − pk
k=1
Transform analog poles {pk } into digital poles {e pk Ts } to obtain

H(z).
The following transformations are useful.

1 Ts
−→
s − pi 1 − e pi Ts z −1
1 (−1)m−1 d m−1 Ts
−→
(s − pi )m (m − 1)! dpim−1 1 − e pi Ts z −1
(s + a) 1 − e −aTs cos(bTs )z −1
−→ Ts
(s + a)2 + b 2 1 − 2e −aTs cos(bTs )z −1 + e −2aTs z −2
b e −aTs sin(bTs )z −1
−→ Ts
(s + a)2 + b 2 1 − 2e −aTs cos(bTs )z −1 + e −2aTs z −2


Example 1: Convert the analog filter with the system function
1
Ha (s) =
(s + 1)(s 2 + s + 1)
into a digital IIR filter by means of the impulse invariance
transformation.
Solution: Using the partial fraction expansion Ha (s) is written as
1
Ha (s) =
(s + 1)(s 2 + s + 1)
1 s
= − 2
s +1 s +s +1
√
1 (s + 1/2) 1 3/2
= − √ +√ √
s + 1 (s + 1/2)2 + ( 3/2)2 3 (s + 1/2) + ( 3/2)2
2

Using the transform expressions, the digital filter system function

H(z) is given as
√
H(z) Ts −1 (1 − e −Ts /2 cos( 3/2Ts )z −1 )
= (1 − e z ) − √
Ts 1 − 2e −Ts /2 cos( 3/2Ts )z −1 + e −Ts z −2
√
1 e −Ts /2 sin( 3/2Ts )z −1
+ √ √
3 1 − 2e −Ts /2 cos( 3/2Ts )z −1 + e −Ts z −2
Substituting z = e −jw , we can obtain the frequency response

H(e jw ). However, appropriate Ts is chosen to avoid the effect of
aliasing.

Problem 1
Convert the analog filter with the system function
(s + 1)
Ha (s) =
(s 2 + 5s + 1)
into a digital IIR filter by means of the impulse invariance

transformation with Ts = 0.1.

Problem 2
In impulse invariance method, a first order pole at s = pk is mapped

into a pole at z = e pk Ts i.e
1 Ts
−→
s − pk 1 − e pk Ts z −1
Determine how a second order pole is mapped i.e.
1
−→?
(s − pk )2

Problem 3
We have seen that in impulse invariance transform, h(n) = ha (nTs ).

Think of a step invariance transform in which the step response of
the discrete-time filter is obtained by sampling the step response of
the continuous-time filter.
Convert the analog filter with the system function
(s + 1)
Ha (s) =
(s 2 + 5s + 1)
into a digital IIR filter by means of the step invariance

transformation.
Hint The
Pnrelation between step Rand impulse response is given as
t
s[n] = k=−∞ h[n] and s(t) = −∞ h(τ )dτ

Impulse Invariance technique suffers from the aliasing and is suitable

only for low pass filters and a small class of band pass filters. The
bilinear transformation from s-plane to z-plane is given by
2 1 − z −1

s=
Ts 1 + z −1
Left half plane in s-domain is mapped into the inside of the unit
circle in the z-domain and Right half plane in s-domain is mapped
into the outside of the unit circle in the z-domain. Stable
transformation.

Mapping from s-plane to z-plane is shown below
Figure: s-plane to z-plane mapping

There is a nonlinear relationship (one to one) between the analog

and dgital frequency given by
2
Ω= tan(w /2)
Ts
Due to this non linear relationship, infinite axis in analog frequency
is mapped to a finite interval 0 to π in the digital frequency. This
frequency scaling causes a frequency distortion know as warping.
We need to take into account this frequency warping while designing

a digital flter. This is done by prewarping the frequencies so that
both distortions cancel out each other.

Design Procedure of Digital Filter using BLT

Given a digital filter specifications wp , ws , Rp , As we wish to
determine H(z) by first designing an equivalent analog filter Ha (s)
and then mapping it into the desired digital filter.
Using Ts , Prewarp the frequencies wp and ws to compute analog
frequencies
2 2
Ωp = tan(wp /2) and Ωs = tan(ws /2)
Ts Ts
Design an analog filter Ha (s) using the specifications Ωp , Ωs , Rp , As .
Transform analog filter transfer function Ha (s) into digital filter
transfer function H(z) as
H(z) = Ha (s)
−1
s= T2 1−z−1
s 1+z

Problem 4
Show that the bilinear transform converts a stable continuous-time

filter into a stable discrete-time filter i.e.
1 It maps the jΩ axis in the s-plane on to the unit circle |z| = 1 in
z-plane.
2 It maps the left-half s-plane into inside the unit circle in the z-plane.

Problem 5
Design a Butterworth digital lowpass filter using bilinear transform

to satisfy the following requirements
π
Passband edge frequency wp = 8
rad/sample
2π
Stopband edge frequency ws = 5
rad/sample
Maximum passband attenuation √ 1 = 0.90
1+2
Minimum Stopband attenuation 1/A = 0.20
Find the transfer function H(z) of the Butterworth low pass filter.

Problem 6
Design a Chebyshev-I digital lowpass filter using bilinear transform

to satisfy the following requirements
π
Passband edge frequency wp = 8
rad/sample
2π
Stopband edge frequency ws = 5
rad/sample
Maximum passband attenuation √ 1 = 0.90
1+2
Minimum Stopband attenuation 1/A = 0.20
Find the transfer function H(z) of the Butterworth low pass filter.

Outline
1 IIR Filter Design
2 Convolution and Correlation

Problem 7
Find the convolution of the two discrete-time signals

x1 (n) = (0.6)n u(n) and x2 (n) = nu(n), where u(n) is the unit step
sequence.
Show using an example that convolving a finite length discrete-time

signal with an infinite length discrete-time signal may result in an
output signal of finite length.

Problem 8
Find the cross-correlation between the discrete-time signals

x1 (n) = u(n) − u(n − 6) and x2 (n) = u(n − 2) − u(n − 5).
Find the auto-correlation of discrete-time signal x(n) = (0.8)n u(n).


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Figure: *

Digital Signal Processing: EC253

Department of Electronics Engineering

Outline of the Lecture

IIR Filter Design

Convolution and Correlation

Department of ElectronicsEngineering National Institute of Technology Andhra Pradesh, India

1 IIR Filter Design

2 Convolution and Correlation

IIR Filter design techniques

Two approaches for digital IIR filter design

Figure: Digital IIR filter design techniques

Department of ElectronicsEngineering National Institute of Technology Andhra Pradesh, India

Analog to Digital Transformation

The design of a digital filter from a prototype analog filter requires

We requie a mapping function s = d(z) from the s-plane to the

For an acceptable digital filter, this mapping must satisfy the

Department of ElectronicsEngineering National Institute of Technology Andhra Pradesh India

Impulse Invariant Transformation

The key ides of impulse invariance transform (IIT) is that impulse

It means h(n) = ha (nTs ) where Ts is the sampling interval.

The frequency response of discrete time filter is given as

Since it is a sampling operation, analog and digital frequencies are

Department of ElectronicsEngineering National Institute of Technology Andhra Pradesh, India

Impulse Invariant Transformation

Mapping from s-plane to z-plane is given by z = e sTs

Figure: s-plane to z-plane mapping

Department of ElectronicsEngineering National Institute of Technology Andhra Pradesh India

Impulse Invariant Transformation

If the continuous-time filter is band limited i.e.

Discrete-time filter frequency response will closely aproximate the

However in practice, Ha (jΩ) is hardly ever band limited, so some

The advantage of impulse invariant transformation is the simplicity

Department of ElectronicsEngineering National Institute of Technology Andhra Pradesh India

Design Procedure of Digital Filter using IIT

Transform analog poles {pk } into digital poles {e pk Ts } to obtain

Impulse Invariant Transformation

The following transformations are useful.

Department of ElectronicsEngineering National Institute of Technology Andhra Pradesh, India

Impulse Invariant Transformation

Department of ElectronicsEngineering National Institute of Technology Andhra Pradesh India

Impulse Invariant Transformation

Using the transform expressions, the digital filter system function

Substituting z = e −jw , we can obtain the frequency response

Department of ElectronicsEngineering National Institute of Technology Andhra Pradesh, India

Convert the analog filter with the system function

into a digital IIR filter by means of the impulse invariance

Department of ElectronicsEngineering National Institute of Technology Andhra Pradesh, India

In impulse invariance method, a first order pole at s = pk is mapped

Department of ElectronicsEngineering National Institute of Technology Andhra Pradesh, India

We have seen that in impulse invariance transform, h(n) = ha (nTs ).

into a digital IIR filter by means of the step invariance

Department of ElectronicsEngineering National Institute of Technology Andhra Pradesh, India

Impulse Invariance technique suffers from the aliasing and is suitable

bilinear transformation from s-plane to z-plane is given by

Department of ElectronicsEngineering National Institute of Technology Andhra Pradesh, India

Mapping from s-plane to z-plane is shown below

Figure: s-plane to z-plane mapping

Department of ElectronicsEngineering National Institute of Technology Andhra Pradesh, India

There is a nonlinear relationship (one to one) between the analog

We need to take into account this frequency warping while designing

Department of ElectronicsEngineering National Institute of Technology Andhra Pradesh, India

Design Procedure of Digital Filter using BLT

Department of ElectronicsEngineering National Institute of Technology Andhra Pradesh, India