K. Deergha Rao - Signals and Systems (2018, Birkhäuser) - 2

K.
Deergha Rao
Signals
and Systems
K. Deergha Rao
Signals and Systems

K. Deergha Rao
Department of Electronics and Communication Engineering
Vasavi College of Engineering (Affiliated to Osmania University)
Hyderabad, Telangana, India
ISBN 978-3-319-68674-5 ISBN 978-3-319-68675-2 (eBook)

https://doi.org/10.1007/978-3-319-68675-2
Library of Congress Control Number: 2017958547
Mathematics Subject Classification (2010): 94A12; 94A05; 93C55; 93C20; 35Q93
© Springer International Publishing AG, part of Springer Nature 2018

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To
My Parents Dalamma and Boddu,
My Beloved Wife Sarojini,
and
My Mentor Prof. M.N.S. Swamy
Preface
The signals and systems course is not only an important element for undergraduate
electrical engineering students but the fundamentals and techniques of the subject
are essential in all the disciplines of engineering. Signals and systems analysis has a
long history, with its techniques and fundamentals found in broad areas of applica-
tions. The signals and systems is continuously evolving and developing in response
to new problems, such as the development of integrated circuits technology and its
applications.
In this book, many illustrative examples are included in each chapter for easy
understanding of the fundamentals and methodologies of signals and systems. An
attractive feature of this book is the inclusion of MATLAB-based examples with
codes to encourage readers to implement exercises on their personal computers in
order to become confident with the fundamentals and to gain more insight into
signals and systems. In addition to the problems that require analytical solutions,
MATLAB exercises are introduced to the reader at the end of some chapters.
This book is divided into 8 chapters. Chapter 1 presents an introduction to signals
and systems with basic classification of signals, elementary operations on signals,
and some real-world examples of signals and systems. Chapter 2 gives time-domain
analysis of continuous time signals and systems, and state-space representation of
continuous-time LTI systems. Fourier analysis of continuous-time signals and sys-
tems is covered in Chapter 3. Chapter 4 deals with the Laplace transform and
analysis of continuous-time signals and systems, and solution of state-space equa-
tions of continuous-time LTI systems using Laplace transform. Ideal continuous-
time (analog) filters, practical analog filter approximations and design methodolo-
gies, and design of special class filters based on pole-zero placement are discussed in
Chapter 5. Chapter 6 discusses the time-domain representation of discrete-time
signals and systems, linear time-invariant (LTI) discrete-time systems and their
properties, characterization of discrete-time systems, and state-space representation
of discrete-time LTI systems. Representation of discrete-time signals and systems in
frequency domain, representation of sampling in frequency domain, reconstruction
of a band-limited signal from its samples, and sampling of discrete-time signals are
vii
viii Preface
detailed in Chapter 7. Chapter 8 describes the z-transform and analysis of LTI

discrete-time systems, the solution of state-space equations of discrete-time LTI
systems using z-transform, and transformations between the continuous-time sys-
tems and discrete-time systems.
The salient features of this book are as follows:
• Provides introductory and comprehensive exposure to all aspects of signal and
systems with clarity and in an easy way to understand.
• Provides an integrated treatment of continuous-time signals and systems and
discrete-time signals and systems.
• Several fully worked numerical examples are provided to help students under-
stand the fundamentals of signals and systems.
• PC-based MATLAB m-files for the illustrative examples are included in
this book.
This book is written at introductory level for undergraduate classes in electrical
engineering and applied sciences that are the prerequisite for upper level courses,
such as communication systems, digital signal processing, and control systems.
Hyderabad, India K. Deergha Rao

Contents
1 Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1
1.1 What is a Signal? . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1
1.2 What is a System? . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1
1.3 Elementary Operations on Signals . . . . . . . . . . . . . . . . . . . . . . . 1
1.3.1 Time Shifting . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2
1.3.2 Time Scaling . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2
1.3.3 Time Reversal . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3
1.4 Classification of Signals . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5
1.4.1 Continuous-Time and Discrete-Time Signals . . . . . . . . . 5
1.4.2 Analog and Digital Signals . . . . . . . . . . . . . . . . . . . . . . 5
1.4.3 Periodic and Aperiodic Signals . . . . . . . . . . . . . . . . . . . 6
1.4.4 Even and Odd Signals . . . . . . . . . . . . . . . . . . . . . . . . . 9
1.4.5 Causal, Noncausal, and Anticausal Signal . . . . . . . . . . . 12
1.4.6 Energy and Power Signals . . . . . . . . . . . . . . . . . . . . . . 13
1.4.7 Deterministic and Random Signals . . . . . . . . . . . . . . . . 20
1.5 Basic Continuous-Time Signals . . . . . . . . . . . . . . . . . . . . . . . . . 20
1.5.1 The Unit Step Function . . . . . . . . . . . . . . . . . . . . . . . . 20
1.5.2 The Unit Impulse Function . . . . . . . . . . . . . . . . . . . . . . 21
1.5.3 The Ramp Function . . . . . . . . . . . . . . . . . . . . . . . . . . . 22
1.5.4 The Rectangular Pulse Function . . . . . . . . . . . . . . . . . . 22
1.5.5 The Signum Function . . . . . . . . . . . . . . . . . . . . . . . . . . 23
1.5.6 The Real Exponential Function . . . . . . . . . . . . . . . . . . . 23
1.5.7 The Complex Exponential Function . . . . . . . . . . . . . . . 24
1.5.8 The Sinc Function . . . . . . . . . . . . . . . . . . . . . . . . . . . . 24
1.6 Generation of Continuous-Time Signals
Using MATLAB . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 28
1.7 Typical Signal Processing Operations . . . . . . . . . . . . . . . . . . . . . 30
1.7.1 Correlation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 30
1.7.2 Filtering . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 31
1.7.3 Modulation and Demodulation . . . . . . . . . . . . . . . . . . . 31
ix
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1.7.4 Transformation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 31
1.7.5 Multiplexing and Demultiplexing . . . . . . . . . . . . . . . . . 32
1.8 Some Examples of Real-World Signals and Systems . . . . . . . . . . 32
1.8.1 Audio Recording System . . . . . . . . . . . . . . . . . . . . . . . 32
1.8.2 Global Positioning System . . . . . . . . . . . . . . . . . . . . . . 33
1.8.3 Location-Based Mobile Emergency
Services System . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33
1.8.4 Heart Monitoring System . . . . . . . . . . . . . . . . . . . . . . . 34
1.8.5 Human Visual System . . . . . . . . . . . . . . . . . . . . . . . . . 36
1.8.6 Magnetic Resonance Imaging . . . . . . . . . . . . . . . . . . . . 36
1.9 Problems . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 37
1.10 MATLAB Exercises . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 39
Further Reading . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 40
2 Continuous-Time Signals and Systems . . . . . . . . . . . . . . . . . . . . . . . 41
2.1 The Representation of Signals in Terms of Impulses . . . . . . . . . . 41
2.2 Continuous-Time Systems . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 42
2.2.1 Linear Systems . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 42
2.2.2 Time-Invariant System . . . . . . . . . . . . . . . . . . . . . . . . . 43
2.2.3 Causal System . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 48
2.2.4 Stable System . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 49
2.2.5 Memory and Memoryless System . . . . . . . . . . . . . . . . . 49
2.2.6 Invertible System . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 49
2.2.7 Step and Impulse Responses . . . . . . . . . . . . . . . . . . . . . 49
2.3 The Convolution Integral . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 49
2.3.1 Some Properties of the Convolution Integral . . . . . . . . . 50
2.3.2 Graphical Convolution . . . . . . . . . . . . . . . . . . . . . . . . . 58
2.3.3 Computation of Convolution Integral
Using MATLAB . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 70
2.3.4 Interconnected Systems . . . . . . . . . . . . . . . . . . . . . . . . 74
2.3.5 Periodic Convolution . . . . . . . . . . . . . . . . . . . . . . . . . . 76
2.4 Properties of Linear Time-Invariant Continuous-Time
System . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 77
2.4.1 LTI Systems With and Without Memory . . . . . . . . . . . . 77
2.4.2 Causality for LTI Systems . . . . . . . . . . . . . . . . . . . . . . 77
2.4.3 Stability for LTI Systems . . . . . . . . . . . . . . . . . . . . . . . 77
2.4.4 Invertible LTI System . . . . . . . . . . . . . . . . . . . . . . . . . . 79
2.5 Systems Described by Differential Equations . . . . . . . . . . . . . . . 82
2.5.1 Linear Constant-Coefficient Differential Equations . . . . . 82
2.5.2 The General Solution of Differential Equation . . . . . . . . 85
2.5.3 Linearity . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 86
2.5.4 Causality . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 86
2.5.5 Time-Invariance . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 87
2.5.6 Impulse Response . . . . . . . . . . . . . . . . . . . . . . . . . . . . 88
2.5.7 Solution of Differential Equations Using
MATLAB . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 91
Contents xi
2.5.8 Determining Impulse Response and Step

Response for a Linear System Described by
a Differential Equation Using MATLAB . . . . . . . . . . . . 92
2.6 Block-Diagram Representations of LTI Systems
Described by Differential Equations . . . . . . . . . . . . . . . . . . . . . . 93
2.7 Singularity Functions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 95
2.8 State-Space Representation of Continuous-Time
LTI Systems . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 98
2.8.1 State and State Variables . . . . . . . . . . . . . . . . . . . . . . . 98
2.8.2 State-Space Representation of Single-Input
Single-Output Continuous-Time LTI Systems . . . . . . . . 99
2.8.3 State-Space Representation of Multi-input
Multi-output Continuous-Time LTI Systems . . . . . . . . . 104
2.9 Problems . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 105
Further Reading . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 110
3 Frequency Domain Analysis of Continuous-Time
Signals and Systems . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 111
3.1 Complex Exponential Fourier Series Representation
of the Continuous-Time Periodic Signals . . . . . . . . . . . . . . . . . . . 111
3.1.1 Convergence of Fourier Series . . . . . . . . . . . . . . . . . . . . 113
3.1.2 Properties of Fourier Series . . . . . . . . . . . . . . . . . . . . . . . 113
3.2 Trigonometric Fourier Series Representation . . . . . . . . . . . . . . . . 128
3.2.1 Symmetry Conditions in Trigonometric
Fourier Series . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 129
3.3 The Continuous Fourier Transform for Nonperiodic
Signals . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 133
3.3.1 Convergence of Fourier Transforms . . . . . . . . . . . . . . . . 135
3.3.2 Fourier Transforms of Some Commonly Used
Continuous-Time Signals . . . . . . . . . . . . . . . . . . . . . . . . 136
3.3.3 Properties of the Continuous-Time Fourier
Transform . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 139
3.4 The Frequency Response of Continuous-Time Systems . . . . . . . . . 159
3.4.1 Distortion During Transmission . . . . . . . . . . . . . . . . . . . 160
3.5 Some Communication Application Examples . . . . . . . . . . . . . . . . 162
3.5.1 Amplitude Modulation (AM) and Demodulation
Amplitude Modulation . . . . . . . . . . . . . . . . . . . . . . . . . . 162
3.5.2 Single-Sideband (SSB) AM . . . . . . . . . . . . . . . . . . . . . . 164
3.5.3 Frequency Division Multiplexing (FDM) . . . . . . . . . . . . . 164
3.6 Problems . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 164
Further Reading . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 170
4 Laplace Transforms . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 171
4.1 The Laplace Transform . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 171
4.1.1 Definition of Laplace Transform . . . . . . . . . . . . . . . . . . 171
xii Contents
4.1.2 The Unilateral Laplace Transform . . . . . . . . . . . . . . . . . 172

4.1.3 Existence of Laplace Transforms . . . . . . . . . . . . . . . . . . 172
4.1.4 Relationship Between Laplace Transform
and Fourier Transform . . . . . . . . . . . . . . . . . . . . . . . . . 172
4.1.5 Representation of Laplace Transform in the S-Plane . . . . 173
4.2 Properties of the Region of Convergence . . . . . . . . . . . . . . . . . . 174
4.3 The Inverse Laplace Transform . . . . . . . . . . . . . . . . . . . . . . . . . 176
4.4 Properties of the Laplace Transform . . . . . . . . . . . . . . . . . . . . . 178
4.4.1 Laplace Transform Properties of Even and
Odd Functions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 182
4.4.2 Differentiation Property of the Unilateral
Laplace Transform . . . . . . . . . . . . . . . . . . . . . . . . . . . . 183
4.4.3 Initial Value Theorem . . . . . . . . . . . . . . . . . . . . . . . . . . 186
4.4.4 Final Value Theorem . . . . . . . . . . . . . . . . . . . . . . . . . . 187
4.5 Laplace Transforms of Elementary Functions . . . . . . . . . . . . . . . 187
4.6 Computation of Inverse Laplace Transform Using
Partial Fraction Expansion . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 194
4.6.1 Partial Fraction Expansion of X(s) with Simple
Poles . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 195
4.6.2 Partial Fraction Expansion of X(s) with Multiple
Poles . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 195
4.7 Inverse Laplace Transform by Partial Fraction
Expansion Using MATLAB . . . . . . . . . . . . . . . . . . . . . . . . . . . 201
4.8 Analysis of Continuous-Time LTI Systems Using
the Laplace Transform . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 202
4.8.1 Transfer Function . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 202
4.8.2 Stability and Causality . . . . . . . . . . . . . . . . . . . . . . . . . 204
4.8.3 LTI Systems Characterized by Linear Constant
Coefficient Differential Equations . . . . . . . . . . . . . . . . . 207
4.8.4 Solution of linear Differential Equations Using
Laplace Transform . . . . . . . . . . . . . . . . . . . . . . . . . . . . 210
4.8.5 Solution of Linear Differential Equations Using
Laplace Transform and MATLAB . . . . . . . . . . . . . . . . . 216
4.8.6 System Function for Interconnections
of LTI Systems . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 217
4.9 Block-Diagram Representation of System Functions
in the S-Domain . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 218
4.10 Solution of State-Space Equations Using Laplace
Transform . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 220
4.11 Problems . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 222
Further Reading . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 225
Contents xiii
5 Analog Filters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 227

5.1 Ideal Analog Filters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 227
5.2 Practical Analog Low-Pass Filter Design . . . . . . . . . . . . . . . . . . . 232
5.2.1 Filter Specifications . . . . . . . . . . . . . . . . . . . . . . . . . . . . 232
5.2.2 Butterworth Analog Low-Pass Filter . . . . . . . . . . . . . . . . 233
5.2.3 Chebyshev Analog Low-Pass Filter . . . . . . . . . . . . . . . . . 237
5.2.4 Elliptic Analog Low-Pass Filter . . . . . . . . . . . . . . . . . . . 245
5.2.5 Bessel Filter . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 248
5.2.6 Comparison of Various Types of Analog Filters . . . . . . . . 249
5.2.7 Design of Analog High-Pass, Band-Pass,
and Band-Stop Filters . . . . . . . . . . . . . . . . . . . . . . . . . . . 252
5.3 Effect of Poles and Zeros on Frequency Response . . . . . . . . . . . . 264
5.3.1 Effect of Two Complex System Poles on
the Frequency Response . . . . . . . . . . . . . . . . . . . . . . . . . 264
5.3.2 Effect of Two Complex System Zeros on
the Frequency Response . . . . . . . . . . . . . . . . . . . . . . . . . 264
5.4 Design of Specialized Analog Filters by Pole-Zero
Placement . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 265
5.4.1 Notch Filter . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 266
5.5 Problems . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 267
Further Reading . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 269
6 Discrete-Time Signals and Systems . . . . . . . . . . . . . . . . . . . . . . . . . . 271
6.1 The Sampling Process of Analog Signals . . . . . . . . . . . . . . . . . . 271
6.1.1 Impulse-Train Sampling . . . . . . . . . . . . . . . . . . . . . . . . 271
6.1.2 Sampling with a Zero-Order Hold . . . . . . . . . . . . . . . . . 272
6.1.3 Quantization and Coding . . . . . . . . . . . . . . . . . . . . . . . 274
6.2 Classification of Discrete-Time Signals . . . . . . . . . . . . . . . . . . . 276
6.2.1 Symmetric and Anti-symmetric Signals . . . . . . . . . . . . . 276
6.2.2 Finite and Infinite Length Sequences . . . . . . . . . . . . . . . 276
6.2.3 Right-Sided and Left-Sided Sequences . . . . . . . . . . . . . 277
6.2.4 Periodic and Aperiodic Signals . . . . . . . . . . . . . . . . . . . 277
6.2.5 Energy and Power Signals . . . . . . . . . . . . . . . . . . . . . . 279
6.3 Discrete-Time Systems . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 281
6.3.1 Classification of Discrete-Time Systems . . . . . . . . . . . . 282
6.3.2 Impulse and Step Responses . . . . . . . . . . . . . . . . . . . . . 286
6.4 Linear Time-Invariant Discrete-Time Systems . . . . . . . . . . . . . . . 286
6.4.1 Input-Output Relationship . . . . . . . . . . . . . . . . . . . . . . . 286
6.4.2 Computation of Linear Convolution . . . . . . . . . . . . . . . 288
6.4.3 Computation of Convolution Sum
Using MATLAB . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 291
6.4.4 Some Properties of the Convolution Sum . . . . . . . . . . . . 291
6.4.5 Stability and Causality of LTI Systems
in Terms of the Impulse Response . . . . . . . . . . . . . . . . . 295
xiv Contents
6.5 Characterization of Discrete-Time Systems . . . . . . . . . . . . . . . . . 297

6.5.1 Non-Recursive Difference Equation . . . . . . . . . . . . . . . 298
6.5.2 Recursive Difference Equation . . . . . . . . . . . . . . . . . . . 298
6.5.3 Solution of Difference Equations . . . . . . . . . . . . . . . . . . 299
6.5.4 Computation of Impulse and Step Responses
Using MATLAB . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 304
6.6 Sampling of Discrete-Time Signals . . . . . . . . . . . . . . . . . . . . . . 305
6.6.1 Discrete-Time Down Sampler . . . . . . . . . . . . . . . . . . . . 306
6.6.2 Discrete-Time Up-Sampler . . . . . . . . . . . . . . . . . . . . . . 306
6.7 State-Space Representation of Discrete-Time LTI Systems . . . . . 307
Single-Output Discrete-Time LTI Systems . . . . . . . . . . . 307
6.7.2 State-Space Representation of Multi-input
Multi-output Discrete-Time LTI Systems . . . . . . . . . . . . 309
6.8 Problems . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 310
Further Reading . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 312
7 Frequency Domain Analysis of Discrete-Time Signals
and Systems . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 313
7.1 The Discrete-Time Fourier Series . . . . . . . . . . . . . . . . . . . . . . . . . 313
7.1.1 Periodic Convolution . . . . . . . . . . . . . . . . . . . . . . . . . . . 314
7.2 Representation of Discrete-Time Signals and Systems
in Frequency Domain . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 316
7.2.1 Fourier Transform of Discrete-Time Signals . . . . . . . . . . 316
7.2.2 Theorems on DTFT . . . . . . . . . . . . . . . . . . . . . . . . . . . . 317
7.2.3 Some Properties of the DTFT of a Complex
Sequence x(n) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 320
7.2.4 Some Properties of the DTFT of a Real
Sequence x(n) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 322
7.3 Frequency Response of Discrete-Time Systems . . . . . . . . . . . . . . 332
7.3.1 Frequency Response Computation Using MATLAB . . . . . 338
7.4 Representation of Sampling in Frequency Domain . . . . . . . . . . . . 344
7.4.1 Sampling of Low-Pass Signals . . . . . . . . . . . . . . . . . . . . 346
7.5 Reconstruction of a Band-Limited Signal from Its Samples . . . . . . 347
7.6 Problems . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 349
Further Reading . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 351
8 The z-Transform and Analysis of Discrete Time
LTI Systems . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 353
8.1 Definition of the z-Transform . . . . . . . . . . . . . . . . . . . . . . . . . . . 353
8.2 Properties of the Region of Convergence for
the z-Transform . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 355
8.3 Properties of the z-Transform . . . . . . . . . . . . . . . . . . . . . . . . . . . 360
8.4 z-Transforms of Some Commonly Used Sequences . . . . . . . . . . . 365
8.5 The Inverse z-Transform . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 371
Contents xv
8.5.1 Modulation Theorem in the z-Domain . . . . . . . . . . . . . . 372

8.5.2 Parseval’s Relation in the z-Domain . . . . . . . . . . . . . . . 372
8.6 Methods for Computation of the Inverse z-Transform . . . . . . . . . 374
8.6.1 Cauchy’s Residue Theorem for Computation
of the Inverse z-Transform . . . . . . . . . . . . . . . . . . . . . . 374
8.6.2 Computation of the Inverse z-Transform
Using the Partial Fraction Expansion . . . . . . . . . . . . . . . 375
8.6.3 Inverse z-Transform by Partial Fraction Expansion
Using MATLAB . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 379
8.6.4 Computation of the Inverse z-Transform
Using the Power Series Expansion . . . . . . . . . . . . . . . . 380
8.6.5 Inverse z-Transform via Power Series Expansion
Using MATLAB . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 383
8.6.6 Solution of Difference Equations
Using the z-Transform . . . . . . . . . . . . . . . . . . . . . . . . . 383
8.7 Analysis of Discrete-Time LTI Systems in the
z-Transform Domain . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 385
8.7.1 Transfer Function . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 385
8.7.2 Poles and Zeros of a Transfer Function . . . . . . . . . . . . . 386
8.7.3 Frequency Response from Poles and Zeros . . . . . . . . . . 388
8.7.4 Stability and Causality . . . . . . . . . . . . . . . . . . . . . . . . . 389
8.7.5 Minimum-Phase, Maximum-Phase, and
Mixed-Phase Systems . . . . . . . . . . . . . . . . . . . . . . . . . . 395
8.7.6 Inverse System . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 395
8.7.7 All-Pass System . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 397
8.7.8 All-Pass and Minimum-Phase Decomposition . . . . . . . . 399
8.8 One-Sided z-Transform . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 401
8.8.1 Solution of Difference Equations with
Initial Conditions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 404
8.9 Solution of State-Space Equations Using z-Transform . . . . . . . . . 405
8.10 Transformations Between Continuous-Time Systems
and Discrete-Time Systems . . . . . . . . . . . . . . . . . . . . . . . . . . . . 408
8.10.1 Impulse Invariance Method . . . . . . . . . . . . . . . . . . . . . . 409
8.10.2 Bilinear Transformation . . . . . . . . . . . . . . . . . . . . . . . . 411
8.11 Problems . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 413
Further Reading . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 417
Index . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 419
Chapter 1
Introduction
1.1 What is a Signal?
A signal is defined as any physical quantity that carries information and varies with
time, space, or any other independent variable or variables. The world of science and
engineering is filled with signals: speech, television, images from remote space
probes, voltages generated by the heart and brain, radar and sonar echoes, seismic
vibrations, signals from GPS satellites, signals from human genes, and countless
other applications.
1.2 What is a System?
A system is defined mathematically as a transformation that maps an input signal x(t)

into an output signal y(t) as illustrated in Figure 1.1. This can be denoted as
yð t Þ ¼ ℜ½ xð t Þ ð1:1Þ
where ℜ is an operator.
For example, a communication system itself is a combination of transmitter,
channel, and receiver. A communication system takes speech signal as input and
transforms it into an output signal, which is an estimate of the original input signal.
1.3 Elementary Operations on Signals
In many practical situations, signals related by a modification of the independent

variable t are to be considered. The useful elementary operations on signals including
time shifting, time scaling, and time reversal are discussed in the following subsections.
© Springer International Publishing AG, part of Springer Nature 2018 1

K. D. Rao, Signals and Systems, https://doi.org/10.1007/978-3-319-68675-2_1
2 1 Introduction
()
Figure 1.1 Schematic representation of a system
x(t) x(t-2)
1 1
0 2 t -2 0 2 4 t
-2
x(t+2)
(a) (b)
1
-4 -2 0 2 t
(c)
Figure 1.2 Illustration of time shifting
1.3.1 Time Shifting
Consider a signal x(t). If it is time shifted by t0, the time-shifted version of x(t) is
represented by x(t t0). The two signals x(t) and x(t t0) are identical in shape but
time shifted relative to each other. If t0 is positive, the signal x(t) is delayed (right
shifted) by t0. If t0 is negative, the signal is advanced (left shifted) by t0. Signals
related in this fashion arise in applications such as sonar, seismic signal processing,
radar, and GPS. The time shifting operation is illustrated in Figure 1.2. If the signal x
(t) shown in Figure 1.2(a) is shifted by t0 ¼ 2 seconds, x(t 2) is obtained as shown
in Figure 1.2(b), i.e., x(t) is delayed (right shifted) by 2 seconds. If the signal is
advanced (left shifted) by 2 seconds, x(t þ 2) is obtained as shown in Figure 1.2(c),
i.e., x(t) is advanced (left shifted) by 2 seconds.
1.3.2 Time Scaling
The compression or expansion of a signal is known as time scaling. The time-scaling

operation is illustrated in Figure 1.3. If the signal x(t) shown in Figure 1.3(a) is
1.3 Elementary Operations on Signals 3
x(2t)
x(t)
2
2
-1
-2 1 2 t
-4 -2 2 4 t
-2
-2
(a) (b)
x(t/2)
-4 -2 2 4 t
-2
Figure 1.3 Illustration of time scaling
x(t) x(-t)
2 2
-2 2
t -2 2
t
(a) (b)
Figure 1.4 Illustration of time reversal
compressed in time by a factor 2, x(2t) is obtained as shown in Figure 1.3(b). If the

signal x(t) is expanded by a factor of 2, x(t/2) is obtained as shown in Figure 1.3(c).
1.3.3 Time Reversal
The signal x(t) is called the time reversal of the signal x(t). The x(t) is obtained
from the signal x(t) by a reflection about t ¼ 0. The time reversal operation is
illustrated in Figure 1.4. The signal x(t) is shown in Figure 1.4(a), and its time
reversal signal x(t) is shown in Figure 1.4(b).
4 1 Introduction
Example 1.1 Consider the following signals x(t) and xi(t), i ¼ 1,2,3. Express them
using only x(t) and its time-shifted, time-scaled, and time-inverted version.
x(t)
2
2
t t
-2 2 4
-2 t 4 t
2
Solution
x(t) x(t-2)
2 2
-2 0 t 0 2 t
x(-t)
2 2
t t
0 2 0 2 4
x1 ðt Þ ¼ xðt 2Þ þ xðt þ 2Þ
x(-t)
2 2
t t
2 -2
1.4 Classification of Signals 5
x2 ðt Þ ¼ xðt 2Þ þ xðt 2Þ
t
x3 ðt Þ ¼ 2x 2
2
1.4 Classification of Signals
Signals can be classified in several ways. Some important classifications of signals

are:
1.4.1 Continuous-Time and Discrete-Time Signals
Continuous-time signals are defined for a continuous of values of the independent

variable. In the case of continuous-time signals, the independent variable t is
continuous as shown Figure 1.5(a).
Discrete-time signals are defined only at discrete times, and for these signals, the
independent variable n takes on only a discrete set of amplitude values as shown in
Figure 1.5(b).
1.4.2 Analog and Digital Signals
An analog signal is a continuous-time signal whose amplitude can take any value in
a continuous range. A digital signal is a discrete-time signal that can only have a
discrete set of values. The process of converting a discrete-time signal into a digital
signal is referred to as quantization.
a 4
b 4
3.5 3.5
3 3
2.5 2.5
Amplitude
Amplitude
2 2
1.5 1.5
1 1
0.5 0.5
0 0
−0.5 −0.5
0 2 4 6 8 10 12 14 16 0 2 4 6 8 10 12 14 16
Time Time index n
Figure 1.5 (a) Continuous-time signal, (b) discrete-time signal

6 1 Introduction
1.4.3 Periodic and Aperiodic Signals
A signal x(t) is said to be periodic with period T(a positive nonzero value), if it
exhibits periodicity, i.e., x(t þ T) ¼ x(t), for all values of t as shown in Figure 1.6(a).
Periodic signal has the property that it is unchanged by a time shift of T.
A signal that does not satisfy the above periodicity property is called an aperiodic
signal. The signal shown in Figure 1.6(b) is an example of an aperiodic signal.
Example 1.2 For each of the following signals, determine whether it is periodic or
aperiodic. If periodic, find the period.
(i) x(t) ¼ 5 sin(2πt)
(ii) x(t) ¼ 1 þ cos(4t þ 1)
(iii) x(t) ¼ e2t
(iv) xðt Þ ¼ ejð5tþ2Þ
π
(v) xðt Þ ¼ ejð5tþ2Þe

π 2t
Solution
(i) 2π ¼ 1:
It is periodic signal, period ¼ 2π
π
(ii) It is periodic, period ¼ 2π4 ¼ 2
(iii) It is aperiodic,
(iv) It is periodic. period ¼ 2π5
(v) Since x(t) is a complex exponential multiplied by a decaying exponential, it is
aperiodic.
Example 1.3 If a continuous-time signal x(t) is periodic, for each of the following
signals, determine whether it is periodic or aperiodic. If periodic, find the period.
(i) x1(t) ¼ x(2t)
(ii) x2(t) ¼ x(t/2)
Solution Let T be the period of x(t). Then, we have
xð t Þ ¼ xð t þ T Þ
x(t)
1
Amplitude
0.5
1
0
−0.5
−1
0 2 Time
0 0.05 0.1 0.15 0.2 1
Time (sec)
(a) (b)
Figure 1.6 (a) Periodic signal, (b) aperiodic signal

(i) For x1(t) to be periodic,
xð2t Þ ¼ xð2t þ T Þ

T
xð2t þ T Þ ¼ x 2 t þ
2

T
¼ x1 t þ
2

Since x1 ðt Þ ¼ x1 t þ T2 , x1(t) is periodic with fundamental period T2 .
As x1(t) is compressed version of x(t) by half, the period of x1(t) is also com-
pressed by half.
(ii) For x2(t) to be periodic,
t
xðt=2Þ ¼ x þT
2
t
1
x þ T ¼ x ðt þ 2T Þ
2 2
¼ x2 ðt þ 2T Þ
Since x2(t) ¼ x2(t þ 2T ), x2(t) is periodic with fundamental period 2T. As x2(t) is
expanded version of x(t) by two, the period of x2(t) is also twice the period of x(t).
Proposition 1.1 Let continuous-time signals x1(t) and x2(t) be periodic signals with
fundamental periods T1 and T2, respectively. The signal x(t) that is a linear combi-
nation of x1(t) and x2(t) is periodic if and only if there exist integers m and k such that
mT1 ¼ kT2 and
T1 k
¼ ¼ rational number ð1:2Þ
T2 m
The fundamental period of x(t) is given by mT1 ¼ kT2 provided that the values of m
and k are chosen such that the greatest common divisor (gcd) between m and k is 1.
Example 1.4 For each of the following signals, determine whether it is periodic or
aperiodic. If periodic, find the period.
(i) x(t) ¼ 2 cos(4πt) þ 3 sin(3πt)
(ii) x(t) ¼ 2 cos(4πt) þ 3 sin(10t)
Solution
(i) Let x1(t) ¼ 2 cos(4πt) and x2(t) ¼ 3 sin(3πt).
The fundamental period of x1(t) is
2π 1
T1 ¼ ¼
4π 2
8 1 Introduction
2π 2
T2 ¼ ¼
3π 3
The ratio TT 12 ¼ 1=2
2=3 ¼ 4 is a rational number. Hence, x(t) is a periodic signal.
3
The fundamental period of the signal x(t) is 4T1 ¼ 3T2 ¼ 2 seconds.

(ii) Let x1(t) ¼ 2 cos(4πt) and x2(t) ¼ 3 sin(10t).
2π 1
T1 ¼ ¼
4π 2
2π π
T2 ¼ ¼
10 5
1=2
The ratio TT 12 ¼ π=5 ¼ 2π
5
is not a rational number. Hence, x(t) is an aperiodic signal.
Example 1.5 Consider the signals

2πt 8πt
x1 ðt Þ ¼ cos þ 2 sin
5 5
x2 ðt Þ ¼ sin ðπt Þ
Determine whether x3(t) ¼ x1(t)x2(t) is periodic or aperiodic. If periodic, find the

period.
Solution Decomposing signals x1(t) and x2(t) into sums of exponentials gives
jð8πt=5Þ jð8πt=5Þ
x1 ðt Þ ¼ 12 ejð2πt=5Þ þ 12 ejð2πt=5Þ þ e j e j
ejðπtÞ ejðπtÞ
x2 ð t Þ ¼
2j 2j
Then,
1 jð7πt=5Þ 1 jð3πt=5Þ 1 jð3πt=5Þ 1 jð7πt=5Þ ejð13πt=5Þ

x3 ð t Þ ¼ e e þ e e
4j 4j 4j 4j 2
ejð3πt=5Þ ejð3πt=5Þ ejð13πt=5Þ
þ þ
2 2 2
It is seen that all complex exponentials are powers of ej(π/5). Hence, it is periodic.
2π
Period is π=5 ¼ 10 seconds:
x(t) x(t)
A A
-b 0 b t -b 0 b t
-A
(a) (b)
Figure 1.7 (a) Even signal, (b) odd signal
1.4.4 Even and Odd Signals
The continuous-time signal is said to be even when x(t) ¼ x(t). The continuous-
time signal is said to be odd when x(t) ¼ x(t). Odd signals are also known as
nonsymmetrical signals. Examples of even and odd signals are shown in Figure 1.7
(a) and Figure 1.7(b), respectively.
Any signal can be expressed as sum of its even and odd parts as
xð t Þ ¼ xe ð t Þ þ xo ð t Þ ð1:3Þ
The even part and odd part of a signal are
xðt Þ þ xðt Þ
x e ðt Þ ¼ ð1:3aÞ
2
xðt Þ xðt Þ
xo ð t Þ ¼ ð1:3bÞ
2
Some important properties of even and odd signals are:
(i) Multiplication of an even signal by an odd signal produces an odd signal.
Proof Let y(t) ¼ xe(t)xo(t)
yðt Þ ¼ xe ðt Þxo ðt Þ

ð1:4Þ
¼ xe ðt Þxo ðt Þ ¼ yðt Þ
Hence, y(t) is an odd signal.

(ii) Multiplication of an even signal by an even signal produces an even signal.
Proof Let y(t) ¼ xe(t)xe(t)
yðt Þ ¼ xe ðt Þxe ðt Þ

ð1:5Þ
¼ xe ðt Þxe ðt Þ ¼ yðt Þ
Hence, y(t) is an even signal.

10 1 Introduction
(iii) Multiplication of an odd signal by an odd signal produces an even signal.

Proof Let y(t) ¼ xo(t)xo(t)
yðt Þ ¼ x0 ðt Þxo ðt Þ

¼ ðxo ðt ÞÞðxo ðt ÞÞ
¼ xo ðt Þxo ðt Þ
¼ yð t Þ
Hence, y(t) is an even signal.

It is seen from Figure 1.7(a) that the even signal is symmetric about the vertical
axis, and hence
ðb ðb
xe ðt Þdt ¼ 2 xe ðt Þdt ð1:6Þ
b 0
From Figure 1.6(b), it is also obvious that

ðb
xo ðt Þdt ¼ 0 ð1:7Þ
b
Eqs. (1.6) and (1.7) are valid for no impulse or its derivative at the origin. These
properties are proved to be useful in many applications.
Example 1.6 Find the even and odd parts of x(t) ¼ ej2t.
Solution From Eq. (1.3),
ej2t ¼ xe ðt Þ þ xo ðt Þ
where
xðt Þ þ xðt Þ ej2t þ ej2t

xe ð t Þ ¼ ¼ ¼ cos ð2t Þ
2 2
xðt Þ xðt Þ ej2t ej2t
xo ð t Þ ¼ ¼ ¼ j sin ð2t Þ:
2 2
Example 1.7 If xe(t) and xo(t) are the even and odd parts of x(t), show that
ð1 ð1 ð1
x2 ðt Þdt ¼ x2e ðt Þdt þ x2o ðt Þdt ð1:8Þ
1 1 1
Solution ð 1 ð1
x2 ðt Þdt ¼ ðxe ðt Þ þ xo ðt ÞÞ2 dt
1 1
ð1 ð1 ð1
¼ x2e ðt Þdt þ2 xe ðt Þxo ðt Þdt þ x2o ðt Þdt
1 1 1
ð1 ð1
¼ x2e ðt Þdt þ x2o ðt Þdt
1 1
ð1
Since 2 xe ðt Þxo ðt Þdt ¼ 0
1
Example 1.8 For each of the following signals, determine whether it is even, odd,
or neither (Figure 1.8)
Solution By definition a signal is even if and only if x(t) ¼ x(t), while a signal is
odd if and only if x(t) ¼ x(t).
(a) It is readily seen that x(t) 6¼ x(t) for all t and x(t) 6¼ x(t) for all t; thus x(t) is
neither even nor odd.
(b) Since x(t) is symmetric about t ¼ 0, x(t) is even.
(c) Since x(t) ¼ x(t), x(t) is odd in this case.
x(t) x(t)
2 2
-4 -2 2 4 t
-2 2
(a) (b)
x(t)
2
-2
2 tt
-2
(c)
Figure 1.8 Signals of example 1.8
12 1 Introduction
x(t) x(t) x(t)
t t
t
(a) (b) (c)

Figure 1.9 (a) Causal signal, (b) noncausal signal, (c) anticausal signal
1.4.5 Causal, Noncausal, and Anticausal Signal
A causal signal is one that has zero values for negative time, i.e., t < 0. A signal is
noncausal if it has nonzero values for both the negative and positive times. An
anticausal signal has zero values for positive time, i.e., t > 0. Examples of causal,
noncausal, and anticausal signals are shown in Figure 1.9(a), 1.9(b), and 1.9(c),
respectively.
Example 1.9 Consider the following noncausal continuous-time signal. Obtain its
realization as causal signal.
x(t)
-1 -0.5
0 0.5 1 t
Solution
x(t)
1
0.5
0 1.5 2 t
-1
1.4.6 Energy and Power Signals
A signal x(t) with finite energy, which means that amplitude ! 0 as time ! 1, is
said to be energy signal, whereas a signal x(t) with finite and nonzero power is said to
be power signal. The instantaneous power p(t) of a signal x(t) can be expressed by
pð t Þ ¼ x 2 ð t Þ ð1:9Þ
The total energy of a continuous-time signal x(t) can be defined as

ð1
E¼ x2 ðt Þdt ð1:10aÞ
1
for a complex valued signal

ð1
E¼ jxðt Þj2 dt ð1:10bÞ
1
Since the power is the time average of energy, the average power is defined as
ð
1 T=2 2
P ¼ limT!1 x ðtÞdt. The signal x(t) expressed by Eq. (1.11), which is
T T=2
shown in Figure 1.10(a), is an example of energy signal.
(
t 0<t1
xð t Þ ¼ ð1:11Þ
1 1<t2
The energy of the signal is given by

ð1 ð1 ð2
1 4
E¼ x ðt Þdt ¼
2
t dt þ
2
1 dt ¼ þ1¼
1 0 1 3 3
The signal x(t) shown in Figure 1.10(b) is an example of a power signal. The
signal is periodic with period 2. Hence, averaging x2(t) over infinitely large time
interval is the same as averaging over one period, i.e., 2. Thus, the average power P is
ð1 ð1
1 1 4
P¼ x2 ðt Þdt ¼ 4t 2 dt ¼
2 1 2 1 3
Figure 1.10(a) Energy x(t)

signal
0 1 2 Time
14 1 Introduction
Figure 1.10(b) Power x(t)

signal
-4 -2 2 4 t
-2
Thus, an energy signal has finite energy and zero average power, whereas a power
signal has finite power and infinite energy.
Example 1.10 Compute energy and power for the following signals, and determine
whether each signal is energy signal, power signal, or neither.
(i) x(t) ¼ 4sin(2πt), 1 < t < 1.
2|t|
(ii) x(t) ¼ 2e
8 , 1 < t < 1.
< 2
pffi t > 1
(iii) xðt Þ ¼
: t
0 t 1:
(iv) x(t) ¼ eat for real value of a
(v) x(t) ¼ cos (t)
π
(vi) xðtÞ ¼ e jð2tþ 4Þ
Solution
ð1 ð1
(i) 2
E¼ jxðt Þj dt ¼ j4 sin ð2πt Þj2 dt
1 1
ð1
1 cos ð4πt Þ
¼ 16 dt
1 2
ð1 ð1
1
¼ 16 dt 8 cos ð4πt Þdt
1 2 1
¼1
ð ð
1 T=2 2 1 T=2
P ¼ limT!1 x ðtÞdt ¼ limT!1 16 sin 2 ð2πtÞdt
T T=2 T T=2
ð
1 T=2 1 cos ð4πtÞ

¼ limT!1 16 dt
T T=2 2
ð ð
1 T=2 1 1 T=2 cos ð4πtÞ
¼ 16 limT!1 dt 16 limT!1 dt
T T=2 2 T T=2 2
¼8
The energy of the signal is infinite, and its average power is finite; x(t) is a power
signal.
(ii) x(t) ¼ 2e2|t|
ð1 ð1
2jtj 2
E¼ jxðt Þj2 dt ¼ 2e dt
1 1
ð0 ð1
¼4 e4t dt þ 4 e4t dt
1 0
4 4t 0 4 1
¼ e 1 þ e4t 0
4 4
4 4
¼ þ ¼2
4 4
ð ð
1 T=2 2 1 T=2 2jtj 2
P ¼ limT!1 x ðt Þdt ¼ limT!1 2e dt
T T=2 T T=2
ð ð
1 0 1 T=2 4t
¼ 4 limT!1 e4t dt þ 4 limT!1 e dt
T T=2 T 0
4 1 0 4 1 T=2
¼ limT!1 e4t T=2 þ limT!1 e4t 0
4 T 4 T
4 1 4 1
¼ limT!1 1 e2T þ limT!1 e2T 1
4 T 4 T
¼0þ0¼0
The energy of the signal is finite, and its average power is zero; x(t) is an energy
signal.
8
<p2
ffi t>1
(iii) xðt Þ ¼ t
:
0 t 1:
ð1 ð1
4
E¼ jxðt Þj2 dt ¼ dt
1 1 t
¼ 4 ln ½t 1
1
¼1
16 1 Introduction
ð T=2 ð T=2
1 1
4
P ¼ lim x2 ðt Þdt ¼ lim dt
T!1 T T=2 T!1 T
1 t
1T=2

1 1 T 1
¼ 4 lim ln ½t ¼ 4 lim ln ln ½1
T!1 T T!1 T 2 T

1 T
¼ 4 lim ln
T!1 T 2
0
1
T
ln
B 2 C
¼ 4 lim B
@
C
T!1 T A
Using L’Hospital’s rule, we see that the power of the signal is zero. That is
T 2
ln 2
P ¼ 4 lim ¼ 4 lim T ¼ 0
T!1 T T!1 1
The energy of the signal is infinite and its average power is zero; x(t) is neither
energy signal nor power signal.
(iv) x(t) ¼ eat for real value of a
ð1 ð1
jeat j dt ¼ 1,
2 2
E¼ jxðt Þj dt ¼
1 1
ð ð
1 T=2 2 1 T=2 2at
P ¼ limT!1 x ðt Þdt ¼ limT!1 e dt
T T=2 T T=2
aT aT aT
e eaT e e
¼ lim ¼ lim lim
T!1 2aT T!1 2aT T!1 2aT
aT
e
¼ lim 0
T!1 2aT
Using L’Hospital’s rule, we see that the power of the signal is infinite. That is,
aT
eaT e
P ¼ lim ¼ lim ¼1
T!1 2aT T!1 2
The energy of the signal is infinite and its average power is infinite; x(t) is neither
energy signal nor power signal.
(v) x(t) ¼ cos(t)
ð1 ð1
2
E¼ jxðt Þj dt ¼ cos 2 ðt Þdt ¼ 1,
1 1
ð ð
1 T=2 2 1 T=2
P ¼ limT!1 x ðt Þdt ¼ limT!1 cos 2 ðt Þ dt
T T=2 T T=2
ð
1 T=2 1 þ cos ð2t Þ

¼ limT!1 dt
T T=2 2
ð ð
1 T=2 1 1 T=2 cos ð2t Þ
¼ limT!1 dt þ limT!1 dt
T T=2 2 T T=2 2
1
¼
2
The energy of the signal is infinite and its average power is finite; x(t) is a power
signal.
(vi) xðt Þ ¼ ejð2tþ4Þ , jxðt Þj ¼ 1.

π
ð1 ð1
2
E¼ jxðt Þj dt ¼ dt ¼ 1,
1 1
ð T=2 ð
1 1 T=2
P ¼ limT!1 x ðt Þdt ¼ limT!1
2
1 dt ¼ limT!1 1 ¼ 1
T T=2 T T=2
The energy of the signal is infinite and its average power is finite; x(t) is a power
signal.
Example 1.11 Consider the following signals, and determine the energy of each
signal shown in Figure 1.11. How does the energy change when transforming a
signal by time reversing, sign change, time shifting, or doubling it?
x(t) (t)
2 2
(t)
2 t
t t
2 -2
( ) -2
2
( )
4
2 4
t
2
t
Figure 1.11 Signals of example 1.11

18 1 Introduction
Solution (
t 0<t2
xðtÞ ¼
0 otherwise
ð1 ð2 2
2 t 3 8
Ex ¼ jxðtÞj dt ¼ t dt ¼ ¼ 2
1 0 3 0 3
(
t 2 < t 0
x1 ðtÞ ¼
0 otherwise
ð1 ð0 0
t3 8
E x1 ¼ jx1 ðtÞj2 dt ¼ t 2 dt ¼ ¼
1 2 3 2 3
(
t 0 < t 2
x2 ðtÞ ¼
0 otherwise
ð1 ð2 2
t3 8
E x2 ¼ jx2 ðtÞj2 dt ¼ t 2 dt ¼ ¼
1 0 3 0 3
(
ðt 2Þ 2 < t 4
x3 ðtÞ ¼
0 otherwise
ð1 ð2 ð4
E x3 ¼ jx3 ðtÞj2 dt ¼ ðt 2Þ2 dt ¼ ðt 2 4t þ 4Þdt
1 0 2
4
3
t
¼ 2t 2 þ 4t
3 2
8
¼
3
(
2t 0<t2
x4 ðtÞ ¼
0 otherwise
ð1 ð2 2
2 t 3 32
E x4 ¼ jx4 ðtÞj dt ¼ 4t dt ¼ 4 ¼ 2
1 0 3 0 3
The time reversal, sign change, and time shifting do not affect the signal energy.
Doubling the signal quadruples its energy. Similarly, it can be shown that the energy
of k x(t) is k2Ex.
Proposition 1.2 The sum of two sinusoids of different frequencies is the sum of the
power of individual sinusoids regardless of phase.
Proof Let us consider a sinusoidal signal x(t) ¼ Acos(Ωt + θ). The power of x(t) is
given by
ð T=2 ð T=2
1 1
P ¼ limT!1 x2 ðtÞdt ¼ limT!1 A2 cos 2 ðΩt þ θÞdt
T T=2 T T=2
ð T=2
1
¼ limT!1 A2 ½1 þ cos 2 ð2Ωt þ 2θÞdt
2T T=2
"ð ð T=2 #
A2 T=2
¼ limT!1 dt þ cos ð2Ωt þ 2θÞdt
2T T=2 T=2
A2 A2
¼ ½T þ 0 ¼
2T 2
ð1:12Þ
2
Thus, a sinusoid signal of amplitude A has a power A2 regardless of the values of
its frequency Ω and phase θ.
Now, consider the following two sinusoidal signals:
x1 ðt Þ ¼ A1 cos ðΩ1 t þ θ1 Þ
x2 ðt Þ ¼ A2 cos ðΩ2 t þ θ2 Þ
Let xs ðt Þ ¼ x1 ðt Þ þ x2 ðt Þ
The power of the sum of the two sinusoidal signals is given by
ðT
1 2
Ps ¼ limT!1 x2s ðtÞ
T T
2
ð T=2
1 2
¼ limT!1 ½A1 cos ðΩ1 t þ θ1 Þ þ A2 cos ðΩ2 t þ θ2 Þ dt
T
T=2
ðT
1 2 2
¼ limT!1 A cos 2 ðΩ1 t þ θ1 Þdt
T T 1
2
ðT
1 2 2
þ limT!1 A cos 2 ðΩ2 t þ θ2 Þdt
T T 2
2
ðT
2A1 A2 2
þ limT!1 cos ðΩ1 t þ θ1 Þcos ðΩ2 t þ θ2 Þdt
T 2
T
The first and second integrals on the right-hand side are the powers of the two
sinusoidal signals, respectively, and the third integral becomes zero since
cos ðΩ1 t þ θ1 Þ cos ðΩ2 t þ θ2 Þ ¼ cos ½ðΩ1 þ Ω2 Þt þ ðθ1 þ θ2 Þ

þ cos ½ðΩ1 Ω2 Þt þ ðθ1 θ2 Þ
Hence,
A21 A22
Ps ¼ þ ð1:13Þ
2 2
20 1 Introduction
It can be easily extended to sum of any number of sinusoids with distinct

frequencies
1.4.7 Deterministic and Random Signals
For any given time, the values of deterministic signal are completely specified as
shown in Figure 1.12(a). Thus, a deterministic signal can be described mathemati-
cally as a function of time. A random signal takes random statistically characterized
random values as shown in Figure 1.12(b) at any given time. Noise is a common
example of random signal.
1.5 Basic Continuous-Time Signals
1.5.1 The Unit Step Function
The unit step function is defined as

1 t>0
uð t Þ ¼ ð1:14Þ
0 t<0
which is shown in Figure 1.13.

It should be noted that u(t) is discontinuous at t ¼ 0.
1
1
Amplitude
0.5 0.8
0
0.6
−0.5
−1 0.4
0 0.05 0.1 0.15 0.2
Time (sec) 0.2
0
0 50 100 150
(a) (b)
Figure 1.12 (a) Deterministic signal, (b) random signal
Figure 1.13 Unit step u(t)

function 1
t
1.5 Basic Continuous-Time Signals 21
1.5.2 The Unit Impulse Function
The unit impulse function also known as the Dirac delta function, which is often
referred as delta function is defined as
δðt Þ ¼ 0, t 6¼ 0 ð1:15aÞ
ð1
δðt Þ ¼ 1: ð1:15bÞ
1
The delta function shown in Figure 1.14(b) can be evolved as the limit of the
rectangular pulse as shown in Figure 1.14(a).
δðt Þ ¼ lim pΔ ðtÞ ð1:16Þ

Δ!0
As the width Δ ! 0, the rectangular function converges to the impulse function

δ(t) with an infinite height at t ¼ 0, and the total area remains constant at one.
Some Special Properties of the Impulse Function

• Sampling property
If an arbitrary signal x(t) is multiplied by a shifted impulse function, the product is
given by
xðt Þδðt t 0 Þ ¼ xðt 0 Þδðt t 0 Þ ð1:17aÞ
implying that multiplication of a continuous-time signal and an impulse function

produces an impulse function, which has an area equal to the value of the
continuous-time function at the location of the impulse. Also, it follows that for
t0 ¼ 0,
xðt Þδðt Þ ¼ xð0Þδðt Þ ð1:17bÞ
• Shifting property
ð1
xðt Þδðt t 0 Þdt ¼ xðt 0 Þ ð1:18Þ
1
Figure 1.14 p Δ (t)

(a) Rectangular pulse,
(b) unit impulse 1/Δ
d (t)
0 t
−Δ/2 Δ/2 t
(a) (b)
22 1 Introduction
• Scaling property

1 b
δðat þ bÞ ¼ δ t þ ð1:19Þ
j aj a
• The unit impulse function can be obtained by taking the derivative of the unit step
function as follows:
duðt Þ
δ ðt Þ ¼ ð1:20Þ
dt
• The unit step function is obtained by integrating the unit impulse function as
follows:
ðt
uð t Þ ¼ δðt Þdt ð1:21Þ
1
1.5.3 The Ramp Function
The ramp function is defined as

t t>0
r ðt Þ ¼ ð1:22aÞ
0 t<0
which can also be written as
r ðt Þ ¼ tuðt Þ ð1:22bÞ
The ramp function is shown in Figure 1.15.
1.5.4 The Rectangular Pulse Function
The continuous-time rectangular pulse function is defined as
Figure 1.15 The ramp

function
Figure 1.16 The x(t)

rectangular pulse function
- 1 0
1 t
Figure 1.17 The signum x(t)=

function
1
0 t
-1

1 jt j T 1
xð t Þ ¼ ð1:23Þ
0 jt j > T 1
1.5.5 The Signum Function
The signum function also called sign function is defined as

8
< 1 t>0
sgnðt Þ ¼ 0 t¼0 ð1:24Þ
:
1 t<0
1.5.6 The Real Exponential Function
A real exponential function is defined as
xðt Þ ¼ Aeσt ð1:25Þ
where both A and σ are real. If σ is positive, x(t) is a growing exponential signal. The
signal x(t) is exponentially decaying for negative σ. For σ ¼ 0, the signal x(t) is equal
to a constant. Exponentially decaying signal and exponentially growing signal are
shown in Figure 1.18(a) and (b), respectively.
24 1 Introduction
x(t) x(t)
A A
t t
0 0
(a) (b)
Figure 1.18 Real exponential function. (a) Decaying, (b) growing
1.5.7 The Complex Exponential Function
A real exponential function is defined as
xðt Þ ¼ AeðσþjΩÞt ð1:26Þ
Hence
xðt Þ ¼ Aeσt ejΩt ð1:26aÞ
Using Euler’s identity
ejΩt ¼ cos ðΩt Þ þ j sin ðΩt Þ ð1:27Þ
Substituting Eq. (1.27) in Eq. (1.26a), we obtain
xðt Þ ¼ Aeσt ð cos ðΩt Þ þ j sin ðΩt ÞÞ ð1:28Þ
Real sine function and real cosine function can be expressed by the trigonometric
identities as
jΩt jΩt
cos ðΩt Þ ¼ e þe and sin ðΩt Þ ¼ e e
jΩt jΩt
2 2j
1.5.8 The Sinc Function
The continuous-time sinc function is defined as
sin ðπt Þ
Sincðt Þ ¼ ð1:28Þ
πt
which is shown in Figure 1.19
Figure 1.19 The sinc function
Example 1.12 State whether the following signals are causal, anticausal, or
noncausal.
(a) x(t) ¼ e2tu(t)
(b) x(t) ¼ tu(t) t(u(t 1) þ e(1t)u(t 1))
(c) x(t) ¼ et cos (2πt)u(1 t)
Solution (a)
x(t)
0.25
1
t
It is causal since x(t) ¼ 0 for t < 0

(b)
x(t)
t
1
It is causal since x(t) ¼ 0 for t < 0

26 1 Introduction
(c)
(c )
u(1-t)
1
t
It is non causal since for t<0
Example 1.13 Determine and plot the even and odd components of the following
continuous-time signal
xðt Þ ¼ tuðt þ 2Þ tuðt 1Þ
Solution
xðt Þ ¼ tuðt þ 2Þ tuðt 1Þ
xðt Þ ¼ tuðt þ 2Þ þ tuðt þ 1Þ
x(t) x(-t)
1 1
-2 1
t -2 1 2
t
-2 -2
xðtÞ þ xðtÞ
xe ðtÞ ¼
2
1
¼ t uðt þ 2Þ uðt 1Þ uðt þ 2Þ þ uðt þ 1Þ
2
( )
1 2
-2
t
-0.5
-1
xðt Þ xðt Þ
xo ðt Þ ¼
2
1
¼ t ðuðt þ 2Þ uðt 1Þ þ uðt þ 2Þ uðt þ 1ÞÞ
2
1 2
-2 t
-0.5
-1
Example 1.14 Simplify the following expressions:

2 þ3 δðt Þ
(a) tsin t
(b) 4þjt
δ ð t 1Þ
Ð3jt
1
(c) 1 ð4t 3ÞÞδðt 1Þdt
Solution

ð0Þ
(a) sin
0þ3 δðt Þ ¼ 0
3jt δðt 1Þ ¼ 3j δðt 1Þ

4þjt 4þj
(b)
Ð1 Ð1
(c) 1 ð4ð1Þ 3ÞÞδðt 1Þdt ¼ 1 1δðt 1Þdt ¼ 1:
28 1 Introduction
1.6 Generation of Continuous-Time Signals Using

MATLAB
An exponentially damped sinusoidal signal can be generated using the following

MATLAB command:
x(t) ¼ A ∗ sin (2 ∗ pi ∗ f0 ∗ t + θ) ∗ exp (a ∗ t)where a is positive for
decaying exponential.
Example 1.15 Write a MATLAB program to generate the following exponentially
damped sinusoidal signal.
x(t) ¼ 5 sin (2πt)e0.4t 10 t 10
Solution The following MATLAB program generates the exponentially damped
sinusoidal signal as shown in Figure 1.20.
MATLAB program to generate exponentially damped sinusoidal signal
clear all;clc;
x =inline('5*sin(2*pi*1*t).*exp(-.4*t)','t');
t = (-10:.01:10);
plot(t,x(t));
xlabel ('t (seconds)');
ylabel ('Ámplitude');
250
200
150
100
Ámplitude
50
-50
-100
-150
-200
-250
-10 -8 -6 -4 -2 0 2 4 6 8 10
t (seconds)
Figure 1.20 Exponentially damped sinusoidal signal with exponential parameter a ¼ 0.4.
1.6 Generation of Continuous-Time Signals Using MATLAB 29
1.5
0.5
Ámplitude
-0.5
-1
-1.5
-2
-5 -4 -3 -2 -1 0 1 2 3 4 5
t (seconds)
Figure 1.21 Unit step function
Example 1.16 Generate unit step function over [5,5] using MATLAB
Solution The following MATLAB program generates the unit step function over
[5,5] as shown in Figure 1.21.
MATLAB program to generate unit step function over [5,5]
clear all;clc;
u=inline('(t>=0)','t');
t=-5:0.01:5;
plot(t,u(t))
ylabel ('Ámplitude')
axis([-5 5 -2 2])
Example 1.17 Generate the following rectangular pulse function rect(t) using
MATLAB:
t 1, 5 < t < 5
rect 10 ¼
0, elsewhere
Solution The following MATLAB program generates the rectangular pulse func-
tion as shown in Figure 1.22.
30 1 Introduction
1.5
0.5
Ámplitude
-0.5
-1
-1.5
-2
-10 -8 -6 -4 -2 0 2 4 6 8 10
t (seconds)
Figure 1.22 Rectangular pulse function
MATLAB program to generate rectangular pulse function
clear all;clc;
u=inline('(t>=-5)& (t<5)','t');
t=-10:0.01:10;
plot(t,u(t))
ylabel ('Ámplitude')
axis([-10 10 -2 2])
1.7 Typical Signal Processing Operations
1.7.1 Correlation
Correlation of signals is necessary to compare one reference signal with one or more
signals to determine the similarity between them and to determine additional infor-
mation based on the similarity. Applications of cross correlation include cross-
spectral analysis, detection of signals buried in noise, pattern matching, and delay
measurements.
1.7 Typical Signal Processing Operations 31
1.7.2 Filtering
Filtering is basically a frequency domain operation. Filter is used to pass certain band
of frequency components without any distortion and to block other frequency
components. The range of frequencies that is allowed to pass through the filter is
called the passband, and the range of frequencies that is blocked by the filter is called
the stopband. A low-pass filter passes all low-frequency components below a certain
specified frequency Ωc, called the cutoff frequency, and blocks all high-frequency
components above Ωc. A high-pass filter passes all high-frequency components
above a certain cutoff frequency Ωc and blocks all low-frequency components
below Ωc. A band-pass filter passes all frequency components between two cutoff
frequencies Ωc1 and Ωc2 where Ωc1 < Ωc2 and blocks all frequency components
below the frequency Ωc1 and above the frequency Ωc2. A band-stop filter blocks all
frequency components between two cutoff frequencies Ωc1 and Ωc2 where Ωc1 < Ωc2
and passes all frequency components below the frequency Ωc1 and above the
frequency Ωc2. Notch filter is a narrow band-stop filter used to suppress a particular
frequency, called the notch frequency.
1.7.3 Modulation and Demodulation
Transmission media, such as cables and optical fibers, are used for transmission of
signals over long distances; each such medium has a bandwidth that is more suitable
for the efficient transmission of signals in the high-frequency range. Hence, for
transmission over such channels, it is necessary to transform the low-frequency
signal to a high-frequency signal by means of a modulation operation. The desired
low-frequency signal is extracted by demodulating the modulated high-frequency
signal at the receiver end.
1.7.4 Transformation
The transformation is the representation of signals in the frequency domain, and

inverse transform converts the signals from the frequency domain back to the time
domain. The transformation provides the spectrum analysis of a signal. From the
knowledge of the spectrum of a signal, the bandwidth required to transmit the signal
can be determined. The transform domain representations provide additional insight
into the behavior of the signal and make it easy to design and implement algorithms,
such as those for filtering, convolution, and correlation.
32 1 Introduction
1.7.5 Multiplexing and Demultiplexing
Multiplexing is used in situations where the transmitting media is having higher

bandwidth, but the signals have lower bandwidth. Thus, multiplexing is the process
in which multiple signals, coming from different sources, are combined and trans-
mitted over a single channel. Multiplexing is performed by multiplexer placed at the
transmitter end. At the receiving end, the composite signal is separated by demul-
tiplexer performing the reverse process of multiplexing and routes the separated
signals to their corresponding receivers or destinations.
In electronic communications, the two basic forms of multiplexing are time-
division multiplexing (TDM) and frequency-division multiplexing (FDM). In
time-division multiplexing, transmission time on a single channel is divided into
non-overlapped time slots. Data streams from different sources are divided into units
with same size and interleaved successively into the time slots. In frequency-division
multiplexing (FDM), numerous low-frequency narrow bandwidth signals are com-
bined for transmission over a single communication channel. A different frequency
is assigned to each signal within the main channel. Code-division multiplexing
(CDM) is a communication networking technique in which multiple data signals
are combined for simultaneous transmission over a common frequency band.
1.8 Some Examples of Real-World Signals and Systems
1.8.1 Audio Recording System
An audio recording system shown in Figure 1.23(a) takes an audio or speech as input
and converts the audio signal into an electrical signal, which is recorded on a
magnetic tape or a compact disc. An example of recorded voice signal is shown in
Figure 1.23(b).
Audio
Recording
System Audio
output
signal
(a) (b)
Figure 1.23 (a) Audio recording system, (b) the recorded voice signal “don’t fail me again”
1.8 Some Examples of Real-World Signals and Systems 33
1.8.2 Global Positioning System
The satellite-based global positioning system (GPS) consists of a constellation of

24 satellites at high altitudes above the earth. Figure 1.24 shows an example of the
GPS used in air, sea, and land navigation. It requires signals at least from four
satellites to find the user position (X, Y, and Z) and clock bias from the user receiver.
The measurements required in a GPS receiver for position finding are the ranges, i.e.,
the distances from GPS satellites to the user. The ranges are deduced from measured
time or phase differences based on a comparison between the received and receiver-
generated signals. To measure the time, the replica sequence generated in the
receiver is to be compared to the satellite sequence.
The correlator in the user GPS receiver determines which codes are being
received, as well as their exact timing. When the received and receiver-generated
sequences are in phase, the correlator supplies the time delay. Now, the range can be
obtained by multiplying the time delay by the velocity of light. For example,
assuming the time delay as 3 ms (equivalent to 3 blocks of the C/A code of satellite
12), the correlation of satellite 12 producing a peak after 3 ms [Rao06] is shown in
Figure 1.25.
1.8.3 Location-Based Mobile Emergency Services System
Mobile emergency services (MES) refer to the use of mobile positioning technology
to pinpoint mobile users for purposes of providing enhanced wireless emergency
dispatch services (including fire, ambulance, and police) to mobile phone users. In
this emergency service system, user should have assisted GPS-enabled mobile
handset unit. Network service providers will support “Mobile Location Protocol
Figure 1.24 A pictorial

representation of GPS
positioning
34 1 Introduction
Figure 1.25 The 160

correlation of satellite
12 producing a peak 140
120
100
80
60
40
20
−20
0 1000 2000 3000 4000 5000 6000 7000 8000 9000
(MLP).” The MLP serves as the interface between a location server and a location
services (LCS) client.
Whenever user requires an emergency service, he will dial the specified number
for emergency calling. Dialing of emergency service number will generate an
“emergency location immediate service (ELIS).”
ELIS is used to retrieve the position of a mobile subscriber that is involved in an
emergency call or has initiated an emergency service in some other way. The service
consists of the following messages: emergency location immediate request (ELIR)
and emergency location immediate answer (ELIA).
When user has dialed the emergency number, emergency location immediate
request is sent to network service provider.
After receiving the emergency location immediate request from the user, network
service provider extracts the position information and sends emergency location
immediate answer to the mobile user, and service provider asks him to select the
service from ambulance, police, and fire services. Mobile user selects the service,
which he actually needs.
The service provider would find the nearest emergency service center and send an
emergency location report to that center. Whenever an emergency location report is
received, a mark will appear on the corresponding digital map. This mark will
indicate the user’s location. A schematic block diagram of location-based mobile
emergency service system and tracking a mobile user are shown in Figure 1.26
(a) and (b), respectively.
1.8.4 Heart Monitoring System
In cardiac cells of the human body, a small electrical current is produced by the
movement of sodium (Naþ) and potassium (Kþ) ions. The electrical potential
1.8 Some Examples of Real-World Signals and Systems 35
Figure 1.26 (a) Schematic block diagram (b) tracking a mobile user of location-based mobile
emergency service system
Figure 1.27 One cycle of

ECG signal
generated by these ions is known as an electrocardiogram (ECG) signal. The ECG

signal is used by physicians to analyze heart conditions. The ECG signal is very
small (normally 0.0001 to 0.003 volt). These signals are within the frequency range
of 0.05 to 100 Hz. A typical one cycle ECG tracing of a normal heartbeat consists of
a P wave, a QRS complex, and a T wave as shown in Figure 1.27. A small U wave is
normally visible in 50 to 75% of ECGs.
The processing of ECG signal yields information, such as amplitude and timing,
required for a physician to analyze a patient’s heart condition. Detection of R-peaks
and computation of R-R interval of an ECG record are important requirements of
comprehensive arrhythmia analysis systems. Heart rate is computed as
Heart rate ¼ RR interval1 in seconds 60.
An ECG signal with variations in heart rate is shown in Figure 1.28.
36 1 Introduction
Recorded ECG signal

Amplitude 0.5
0
−0.5
0 0.2 0.4 0.6 0.8 1 1.2 1.4 1.6 1.8 2
⫻104
Filtered ECG signal
Amplitude
1
0
−1
0 0.2 0.4 0.6 0.8 1 1.2 1.4 1.6 1.8 2
⫻104
ECG signal R peaks
Amplitude
0.4
0.2
0
0 0.2 0.4 0.6 0.8 1 1.2 1.4 1.6 1.8 2
⫻104
Heart rate
Amplitude
100
50
0
0 0.2 0.4 0.6 0.8 1 1.2 1.4 1.6 1.8 2
⫻104
Figure 1.28 An ECG signal with variations in heart rate
1.8.5 Human Visual System
The human visual system (HVS) can widely perform a number of image processing
operations in a manner superior to anything we are currently able to execute with
computers. To perform such signal processing operations, we have to understand the
way HVS works.
When the reflection from an object (light ray) is observed by the eye, first, it
passes through the cornea, eventually through the aqueous humor, the iris, the lens,
the vitreous humor, and finally reaching the retina. The retina consists photosensitive
cells called cones and rods, which are responsible to convert the incident light energy
into neural signals that are carried to human brain by the optic nerve (Figure 1.29).
1.8.6 Magnetic Resonance Imaging
When an oscillating strong magnetic field is applied at a certain frequency on a

certain part of the human body, the hydrogen atoms in the body emit radio-frequency
waves to form image of the particular part of the body, which is captured by the MRI
machine. An MRI imaging system and a MRI image with brain tumor are shown in
Figure 1.30(a) and (b), respectively.
1.9 Problems 37
Figure 1.29 Human visual

system
Figure 1.30 (a) MRI imaging system, (b) MRI image with brain tumor
1.9 Problems
1. Classify the following continuous-time signals as periodic or aperiodic. If

periodic, determine the period.
π
(i) xðt Þ ¼ cos 2π 3 t þ 2 sinp2ffiffi
t
ffi
(ii) xðt Þ ¼ cos ð2πt Þ þ sin 2πt
(iii) xðt Þ ¼ 12 12 cos ð2t Þ

(iv) xðt Þ ¼ 1 þ sin ð4t Þ þ cos 6t þ π3
(v) x(t) ¼ ej(4t + π/5)
(vi) xðt Þ ¼ cos 2t þ π4
(vii) x(t) ¼ cos(2πt)u(t)
(viii) x(t) ¼ cos2(t)
2. A periodic signal x1(t) has a period 2, and another periodic signal x2(t) has a
period 3. Find the fundamental frequency and period for the signal y
(t) ¼ x1(t) þ x2(t).
38 1 Introduction
3. Classify the following continuous-time signals as even or odd signals or neither

even nor odd. Determine power and energy for each incase of power or energy
signal.
(i) x(t) ¼ (1þ t2)cos2(5t)
(ii) x(t) ¼ u(t)
(iii) x(t) ¼ tu(t)
(iv) x(t) ¼ tsin(2t)
(v) x(t) ¼ t + cos(2t)
(vi) x(t) ¼ e2tsin(2t))
4. Consider the following continuous-time signal:

2π ðt T Þ
xðt Þ ¼ 2 sin
10
Determine the values of T for which the signal is

(i) An even function
(ii) An odd function
5. Classify the following continuous-time signals as power or energy signals or
neither. Determine power and energy for each incase of power or energy signal.
(i) x(t) ¼ sin(2πt)cos(πt)
(ii) x(t) ¼ tu(t)
(iii) x(t) ¼ e3tu(t)
(iv) x(t) ¼ ej3t
6. Determine energy for each of the following signals and comment on the results.
x(t)
x(t) x(t)
1 1
0 1 2
t
t t
0 1 2 0 1 2 3
(a) (b) -1
x(t)
(c)
t
0 1 2
(d)
1.10 MATLAB Exercises 39
7. What is the energy of the signal x(t) ¼ cx(at b), where a 6¼ 0?

8. Verify that ect is neither energy nor a power signal for a complex value of c
with nonzero real part.
9. Show that the energy of x(t) y(t) is Ex + Ey, if x(t) and y(t) are orthogonal.
10. Derive an expression for the power of the following continuous-time signal x
(t) ¼ A1cos(Ω1t + θ1) þ A2cos(Ω2t + θ2) for Ω1 ¼ Ω2.
11. Determine the power of the signal x(t) ¼ AejΩt.
12. Find power for each of the following signals:
(i) x(t) ¼ (5 þ 3sin(2t))cos(5t)
(ii) x(t) ¼ 5 cos(5t) cos (10t)
(iii) x(t) ¼ 2sin(5t) cos (10t)
13. Find odd and even components for each of the following signals:
(a) x(t) ¼ u(t)
(b) x(t) ¼ eatu(t)
14. Evaluate the following expressions:
ð1
π
(i) cos ðt 5ÞÞδð2t 3Þdt
1 2
(ii) e cos 50π t δðt þ 1Þ
2t
ð1
π 50
(iii) 2t
e cos t δðt þ 1Þdt
ð1 2 π
1
(iv) ðt þ cos ð2πt ÞÞδðt 1Þdt
ð1
1
dδðt Þ
(v) et dt
ð1 dt
1
(vi) et δðt 1Þdt
1
1.10 MATLAB Exercises
1. Use MATLAB to generate the continuous-time signal shown in Figure 1.p1.1.

2. Generate and plot each of the following continuous-time signals using
MATLAB:
(i) x(t) ¼ 10 sin(2πt) cos (πt 4) for 10 t 10
(ii) x(t) ¼ 2e0.1t sin(2πt) for 5 t 5
40 1 Introduction
1
Ámplitude
-1
-2
-3
-4
-5
-10 -8 -6 -4 -2 0 2 4 6 8 10
t (seconds)
Figure p1.1 Signal of MATLAB exercise 1
Further Reading
1. Pierce, J.R., Noll, A.M.: Signals: The Science of Telecommunications. American Library, New
Delhi (1960)
2. Lathi, B.P.: Linear Systems and Signals, 2nd edn. Oxford University Press, New York (2005)
3. Mandal, M., Asif, A.: Continuous and Discrete Time Signals and Systems. Cambridge Univer-
sity Press, Cambridge (2007)
Chapter 2
Continuous-Time Signals and Systems
This chapter presents time-domain analysis of continuous-time systems. It develops

representation of signals in terms of impulses. The notions of linearity, time-
invariance, causality, stability, memorability, and invertibility are introduced. It
has shown that the input-output relationship for linear time-invariant (LTI) contin-
uous systems is described in terms of a convolution integral. The differential
equation representation of LTI continuous systems and classical solutions of differ-
ential equations are also presented. Next, block-diagram representation of LTI
continuous-time systems is introduced. Furthermore, a brief discussion on singular-
ity functions is provided. Finally, the state-space representation of continuous-time
LTI systems is described.
2.1 The Representation of Signals in Terms of Impulses
Consider pulse or staircase approximation bx ðt Þ to continuous-time signal x(t) as

shown in Figure 2.1. Then, the approximation signal can be expressed as sum of all
these pulse signals. Define
8
< 1, 0 < t < Δ
δΔ ðt Þ ¼ Δ ð2:1Þ
:
0, otherwise
Since δΔ(t)Δ ¼ 1, b
x ðt Þ can be expressed as
X1
b
x ðt Þ ¼ k¼1
xðkΔÞ δΔ ðt kΔÞΔ ð2:2Þ

42 2 Continuous-Time Signals and Systems
Figure 2.1 Representation x(t)

of a signal in terms of
impulses
........ ........ ........

t
0
As Δ approaches zero, the above approximation b

x ðt Þ can be written as
X1
b
x ðt Þ ¼ limΔ!0 k¼1
xðkΔÞ δΔ ðt kΔÞΔ ð2:3Þ
Also, as Δ ! 0, the summation approaches an integral, and the pulse approaches

unit impulse. Therefore, Eq. (2.3) can be rewritten as
ð1
b
x ðt Þ ¼ xðτÞδðt τÞ dτ ð2:4Þ
1
Thus, a continuous-time signal can be represented as weighted superposition of

shifted impulses. Here the superposition is integration due to nature of the
continuous-time input. The weight x(τ) dτ on the impulse δ(t τ) is determined
from the value of the input signal x(t) at the time of occurrence of each impulse.
2.2 Continuous-Time Systems
2.2.1 Linear Systems
Let x1(t) and x2(t) are the inputs applied to a system characterized by the transfor-
mation operator ℜ[] and y1(t) and y2(t) are the system outputs. A linear system
should satisfy the principles of homogeneity and superposition. Hence, the following
equations hold for a linear system
y1 ðt Þ ¼ ℜ½x1 ðt Þ, ð2:5aÞ
y2 ðt Þ ¼ ℜ½x2 ðt Þ, ð2:5bÞ
Principle of homogeneity:
ℜ½ax1 ðt Þ ¼ ay1 ðt Þ, ð2:6aÞ
ℜ½bx2 ðt Þ ¼ by2 ðt Þ ð2:6bÞ

2.2 Continuous-Time Systems 43
Principle of superposition:
ℜ½x1 ðt Þ þ ℜ½x2 ðt Þ ¼ y1 ðt Þ þ y2 ðt Þ ð2:7Þ
Linearity:
ℜ½ax1 ðt Þ þ ℜ½bx2 ðt Þ ¼ ay1 ðt Þ þ by2 ðt Þ ð2:8Þ
where a and b are arbitrary constants.
2.2.2 Time-Invariant System
A system is time invariant if the behavior and characteristics of the system are fixed
over time. A system is time invariant if a time shift in the input signal results in an
identical time shift in the output signal. For example, a time-invariant system should
produce y(t t0) as the output when x(t t0) is the input. Mathematically it can be
specified as
yð t t 0 Þ ¼ ℜ½ xð t t 0 Þ ð2:9Þ
Example 2.1 Check for linearity and time-invariance of the following system
yðt Þ ¼ txðt Þ
Solution
Linearity:
Let x1(t) and x2(t) be two distinct inputs applied to the system, then
y1 ðt Þ ¼ ℜ½x1 ðt Þ ¼ tx1 ðt Þ, y2 ðt Þ ¼ ℜ½x2 ðt Þ ¼ tx2 ðt Þ
If an input equal to sum of the inputs ax1(t), bx2(t),

x(t) ¼ ax1(t) þ bx2(t) is applied, then
yðt Þ ¼ tax1 ðt Þ þ tbx2 ðt Þ ¼ ay1 ðt Þ þ by2 ðt Þ
Hence, the system is linear.
Time-invariance:
yðt Þ ¼ ℜ½xðt Þ ¼ txðt Þ
The output y(t) of the system delayed by t0 can be written as
yðt t 0 Þ ¼ ðt t 0 Þxðt t 0 Þ
For example, for an input x1(t) ¼ x(t t0), the output y1(t) can be written as
y1 ðt Þ ¼ ℜ½x1 ðt Þ ¼ tx1 ðt Þ ¼ txðt t 0 Þ

yðt t 0 Þ 6¼ y1 ðt Þ
Hence, it is a time variant system.

Example 2.2 Check for linearity and time-invariance of the following system:
yðt Þ ¼ sin ðxðt ÞÞ
Solution
Linearity:
y1 ðt Þ ¼ ℜ½x1 ðt Þ ¼ sin ðx1 ðt ÞÞ, y2 ðt Þ ¼ ℜ½x2 ðt Þ ¼ sin ðx2 ðt ÞÞ
If an input equal to sum of the inputs ax1(t), bx2(t),x(t) ¼ ax1(t) þ bx2(t) is applied,
then
yðt Þ ¼ sin ðax1 ðt ÞÞ þ sin ðbx2 ðt ÞÞ 6¼ ay1 ðt Þ þ by2 ðt Þ
Hence, the system is nonlinear.
Time-invariance:
yðt Þ ¼ ℜ½xðt Þ ¼ sin ðxðt ÞÞ
yðt t 0 Þ ¼ sin ðxðt t 0 ÞÞ
y1 ðt Þ ¼ ℜ½x1 ðt Þ ¼ sin ðx1 ðt ÞÞ ¼ sin ðxðt t 0 ÞÞ

yð t t 0 Þ ¼ y1 ð t Þ
Hence, it is a time-invariant system.

Example 2.3 Determine if the following continuous-time systems are linear or
nonlinear:
dyðt Þ
(i) dt þ 2tyðt Þ ¼ t 2 xðt Þ
(ii) 2y(t) þ 3 ¼ x(t)

ðt
(iii) yðt Þ ¼ xðτÞdτ
1
dyðt Þ
(iv) dt þ 3yðt Þ ¼ xðt Þ dxdtðtÞ
Solution
(i) Let x1(t) and x2(t) be two distinct inputs applied to the system, then
dy1 ðt Þ
þ 2ty1 ðt Þ ¼ t 2 x1 ðt Þ
dt
dy2 ðt Þ
þ 2ty2 ðt Þ ¼ t 2 x2 ðt Þ
dt
dy1 ðt Þ dy ðt Þ
a þ 2aty1 ðt Þ þ b 2 þ 2bty2 ðt Þ ¼ at 2 x1 ðt Þ þ bt 2 x2 ðt Þ
dt dt
(ii) 3
yð t Þ ¼ xð t Þ
2
y1 ðt Þ ¼ ℜ½x1 ðt Þ ¼ x1 ðt Þ 32 , y2 ðt Þ ¼ ℜ½x2 ðt Þ ¼ x2 ðt Þ 32
If an input equal to sum of the inputs ax1(t), bx2(t),x(t) ¼ ax1(t) þ bx2(t) is applied,
then
3 3
yðt Þ ¼ ax1 ðt Þ þ bx2 ðt Þ 6¼ ay1 ðt Þ þ by2 ðt Þ
2 2
Hence, the system is nonlinear.
(iii) Let x1(t) and x2(t) be two distinct inputs applied to the system, then
ðt ðt
y1 ðt Þ ¼ ℜ½x1 ðt Þ ¼ x1 ðτÞdτ, y2 ðt Þ ¼ ℜ½x2 ðt Þ ¼ x2 ðτÞdτ
1 1

ðt ðt
yð t Þ ¼ ax1 ðτÞdτ þ bx2 ðτÞdτ ¼ ay1 ðt Þ þ by2 ðt Þ
1 1

(iv) Let x1(t) and x2(t) be two distinct inputs applied to the system, then
dy1 ðt Þ dx1 ðt Þ
þ 3y1 ðt Þ ¼ x1 ðt Þ
dt dt
dy2 ðt Þ dx2 ðt Þ
þ 3y2 ðt Þ ¼ x2 ðt Þ
dt dt

dy1 ðt Þ dy ðt Þ dx1 ðt Þ dx2 ðt Þ

a þ 3ay1 ðt Þ þ b 2 þ 3by2 ðt Þ 6¼ a2 x1 ðt Þ þ b2 x 2 ð t Þ
dt dt dt dt
The system is nonlinear.
Example 2.4 Determine if the following continuous-time systems are time invariant
or time variant:
(i) y(t) ¼ x(t),
ðt
(ii) yðt Þ ¼ xðτÞdτ,
1
(iii) y(t) ¼ x(4t)
(iv) yðt Þ ¼ ð2 þ sin ðt ÞÞxðt ÞðvÞyðt Þ ¼ dxdtðtÞ
Solution
(i) y(t) ¼ ℜ[x(t)] ¼ x(t)

yðt Þ ¼ ℜ½xðt Þ ¼ xðt Þ
yðt t 0 Þ ¼ xððt t 0 ÞÞ ¼ xðt þ t 0 Þ
y1 ðt Þ ¼ ℜ½x1 ðt Þ ¼ x1 ðt Þ ¼ xðt t 0 Þ

Hence, it is a time-varying system.

ð3
(ii) Let x(t) ¼ δ(t), then yðt Þ ¼ ℜ½xðt Þ ¼ δðt Þdt ¼ 1
3
Now, for an input x1(t) ¼ x(t 6), the output y1(t) can be written as
Ð3
y1 ð t Þ ¼ ℜ½ x 1 ð t Þ ¼ 3 δðt 6Þdt ¼ 0
yðt 6Þ 6¼ y1 ðt Þ

(iii) The output y(t) of the system delayed by t0 can be written as
yðt Þ ¼ ℜ½xðt Þ ¼ xð4t Þ

yðt t 0 Þ ¼ xð4ðt t 0 ÞÞ ¼ xð4t 4t 0 Þ
y1 ðt Þ ¼ ℜ½x1 ðt Þ ¼ x1 ð4t Þ ¼ xð4t t 0 Þ


(iv) y(t) ¼ ℜ[x(t)] ¼ (2 þ sin (t))x(t)
yðt t 0 Þ ¼ ð2 þ sin ðt t 0 ÞÞxðt t 0 Þ
y1 ðt Þ ¼ ℜ½x1 ðt Þ ¼ ð2 þ sin ðt ÞÞxðt t 0 Þ


(v)
dxðt Þ
yðt Þ ¼ ℜ½xðt Þ ¼
dt
dxðt t 0 Þ
yð t t 0 Þ ¼
dt
dxðt t 0 Þ
y1 ð t Þ ¼ ℜ½ x 1 ð t Þ ¼
dt
yð t t 0 Þ ¼ y1 ð t Þ

Example 2.5 Consider an LTI system with the response y(t) as shown in Figure 2.2
to the input signal x(t) ¼ u(t) u(t 2).
Figure 2.2 Response y(t) y(t)

to the input x(t)
2 4 t
(t)
2 4 8 t
-2
Figure 2.3 Response y1(t) to the input x1(t)
Figure 2.4 Response y2(t) (t)

to the input x2(t)
-2 2 4 t
Determine and sketch the response of the system to the following inputs:
(i) x1(t) ¼ x(t) x(t 4)
(ii) x2(t) ¼ x(t) þ x(t þ 2)
Solution
(i) x1(t) ¼ x(t) x(t 4)
Since it is an LTI system, the response y1(t) to the input x1(t) is given by y1(t) ¼
y(t) y(t 4) as depicted in Figure 2.3.
(ii) x2(t) ¼ x(t) þ x(t þ 2)
Since it is an LTI system, the response y2(t) to the input x2(t) is given by y2(t) ¼ y
(t) þ y(t þ 2) as depicted in Figure 2.4.
2.2.3 Causal System
The causal system generates the output depending upon present and past inputs only.
A causal system is non-anticipatory.
2.3 The Convolution Integral 49
2.2.4 Stable System
When the system produces bounded output for bounded input, then the system is
called bounded-input and bounded-output stable. If the signal is bounded, then its
magnitude will always be finite.
2.2.5 Memory and Memoryless System
The output of a memory system at any specified time depends on the inputs at that
specified time and at other times. Such systems have memory or energy storage
elements. The system is said to be static or memoryless if its output depends upon the
present input only.
2.2.6 Invertible System
A system is said to be invertible if the input can be recovered from its output.
Otherwise the system is noninvertible system.
2.2.7 Step and Impulse Responses
If the input to the system is unit impulse input δ(t), the system output is called the
impulse response and denoted by h(t):
hðt Þ ¼ ℜ½δðt Þ ð2:10Þ
If the input to the system is a unit step input u(t), then the system output is called
the step response s(t);that is,
s ð t Þ ¼ ℜ½ uð t Þ ð2:11Þ
2.3 The Convolution Integral
The output of a system for an input expressed as weighted superposition as in

Eq. (2.4) is given by
ð 1
yð t Þ ¼ ℜ½ xð t Þ ¼ ℜ xðτÞδðt τÞdτ ð2:12Þ
1
From the linearity property of the system, Eq. (2.12) can be rewritten as
ð1
yð t Þ ¼ xðτÞℜ½δðt τÞdτ ð2:13Þ
1
For a time-invariant system, ℜ[δ(t τ)] ¼ h(t τ). Hence, we obtain

ð1
yð t Þ ¼ xðτÞhðt τÞdτ ð2:14Þ
1
Thus, the output y(t) of a linear time-invariant system to an arbitrary input x(t) is
obtained in terms of the unit impulse input δ(t). Eq. (2.14) is referred to as the
convolutional integral and is denoted by the symbol * as
ð1
yðt Þ ¼ xðt Þ∗hðt Þ ¼ xðτÞhðt τÞdτ ð2:15Þ
1
2.3.1 Some Properties of the Convolution Integral
2.3.1.1 The Commutative Property
x1 ðt Þ∗x2 ðt Þ ¼ x2 ðt Þ∗x1 ðt Þ ð2:16Þ
Proof This property can be proved by a change of variable.

By the definition of the convolution integral
Ð1
x1 ðt Þ∗x2 ðt Þ ¼ 1 x1 ðτÞx2 ðt τÞdτ
ð2:17Þ
Let V ¼ t τ so that τ ¼ t V, and dτ ¼ dV:
Then
Ð 1
x1 ðt Þ∗x2 ðt Þ ¼ 1 x1 ðt V Þx2 ðV ÞdV
Ð1
¼ 1 x1 ðt V Þx2 ðV ÞdV ð2:18Þ
¼ x2 ðt Þ∗x1 ðt Þ
2.3.1.2 The Distributive Property
x1 ðt Þ∗½x2 ðt Þ þ x3 ðt Þ ¼ x1 ðt Þ∗x2 ðt Þ þ x1 ðt Þ∗x3 ðt Þ ð2:19Þ

Proof By the definition of the convolution integral

ð1

x1 ðt Þ∗½x2 ðt Þ þ x3 ðt Þ ¼ x1 ðτÞ x2 ðt τÞ þ x3 ðt τÞdτ
1
ð1 ð1
ð2:20Þ
¼ x1 ðτÞx2 ðt τÞdτ þ x1 ðτÞx3 ðt τÞdτ
1 1
¼ x1 ðt Þ∗x2 ðt Þ þ x1 ðt Þ∗x3 ðt Þ
2.3.1.3 The Associative Property
½x1 ðt Þ∗x2 ðt Þ∗x3 ðt Þ ¼ x1 ðt Þ∗½x2 ðt Þ∗x3 ðt Þ ð2:21Þ
Proof The left-hand side of the property can be expressed by

ð1
½x1 ðt Þ∗x2 ðt Þ∗x3 ðt Þ ¼ x1 ðτ1 Þx2 ðt τ1 Þdτ1 ∗x3 ðt Þ ð2:22Þ
1
where x1(t) * x2(t) is expressed as a convolution integral. Expanding the second

convolution gives
ð 1 ð 1
½x1 ðt Þ∗x2 ðt Þ∗x3 ðt Þ ¼ x1 ðτ1 Þx2 ðτ2 τ1 Þdτ1 x3 ðt τ2 Þdτ2 ð2:23Þ
1 1
Reversing the order of integration gives

ð1 ð1
dx1 ðt Þ∗x2 ðt Þe∗x3 ðt Þ ¼ x1 ðτ1 Þx2 ðτ2 τ1 Þ x3 ðt τ2 Þdτ1 dτ2 ð2:24Þ
1 1
Similarly, the right-hand side of the property can be written as

Ð 1
x1 ðt Þ∗dx2 ðt Þ∗x3 ðt Þe ¼ x1 ðt Þ∗ 1 x2 ðτ2 Þ x3 ðt τ2 Þdτ2
Ð1 Ð1 ð2:25Þ
¼ 1 1 x1 ðτ1 Þx2 ðτ2 Þx2 ðt τ1 τ2 Þ dτ1 dτ2
Now, it is to be shown that

ð1 ð1 ð1 ð1
x1 ðτ1 Þx2 ðτ2 τ1 Þx3 ðt τ2 Þ dτ1 dτ2 ¼ x1 ðτ1 Þx2 ðτ2 Þ
1 1 1 1 ð2:26Þ
x2 ðt τ1 τ2 Þdτ1 dτ2
In the right hand τ1 integration, let v ¼ τ1 + τ2 and dτ1 ¼ dv.

Then
ð1 ð1 ð1 ð1
x1 ðτ1 Þx2 ðτ2 τ1 Þ∗x3 ðt τ2 Þdτ1 dτ2 ¼ x1 ðV τ2 Þx2 ðτ2 Þ
1 1 1 1
x3 ðt V Þ dV dτ2
ð2:27Þ
Next, let u ¼ V τ2 anddτ2 ¼ du

Then
ð1 ð1 ð 1 ð 1
x1 ðτ1 Þx2 ðτ2 τ1 Þx3 ðt τ2 Þdτ1 dτ2 ¼ x1 ðuÞx2 ðV uÞ
1 1 1 1
x3 ðt V ÞdV d u
ð2:28Þ
ð1 ð1 ð1 ð1
x1 ðτ1 Þx2 ðτ2 τ1 Þx3 ðt τ2 Þdτ1 dτ2 ¼ x1 ðuÞx2 ðV uÞ
1 1 1 1 ð2:29Þ
x3 ðt V ÞdV d u
The above right-hand side and left-hand side integrals are the same except for
change of the variables. Hence, the associative property is proved.
2.3.1.4 Convolution with an Impulse
xðt Þ∗δðt Þ ¼ xðt Þ ð2:30Þ
Proof By definition
ð1
xðt Þ∗δðt Þ ¼ xðτÞδðt τÞ dτ ð2:31Þ
1
Since δ(t τ) is an impulse at τ ¼ t and by sampling property of the impulse,

Ð1
1 xðτÞδðt τÞdτ ¼ xðτÞjτ¼t
ð2:32Þ
¼ xð t Þ
Hence
xðt Þ∗δðt Þ ¼ xðt Þ

2.3.1.5 Convolution with Delayed Input and Delayed Impulse Response
If y(t) ¼ x(t) * h(t), then
xðt t 1 Þ∗hðt t 2 Þ ¼ yðt t 1 t 2 Þ ð2:33Þ
Proof By the convolution integral, we have

ð1
yðt Þ ¼ xðt Þ∗hðt Þ ¼ xðτÞhðt τÞ dτ ð2:34Þ
1
and
ð1
xðt t 1 Þ∗hðt t 2 Þ ¼ xðτ t 1 Þhðt τ t 2 Þ dτ ð2:35Þ
1
Let τ t1 ¼ υ. Then τ ¼ υ + t1, and Eq. (2.35) becomes

ð1
xðt t 1 Þ∗hðt t 2 Þ ¼ xðυÞhðt t 1 t 2 υÞ dυ ð2:36Þ
1
It is observed that replacing t by t t1 t2 in Eq. (2.34), we obtain Eq. (2.36).

Thus, it is proved that
xðt t 1 Þ∗hðt t 2 Þ ¼ yðt t 1 t 2 Þ
Example 2.6 Determine the continuous-time convolution of x(t) and h(t) for the
following:
(i) x(t) ¼ u(t)
h(t) ¼ u(t)
(ii) x(t) ¼ u(t a)
h(t) ¼ u(t b)
(iii) x(t) ¼ u(t þ 1) u(t 1)
h(t) ¼ u(t þ 1) u(t 1)
(iv) x(t) ¼ e(t2)u(t 2)
h(t) ¼ u(t þ 2)
x(t)
h(t)
1
(v)
(t-1)
t 2 4
t
1
(vi) x(t) ¼ u(t)

h(t) ¼ etu(t)
(vii) x(t) ¼ 2(u(t) u(t 2))
h(t) ¼ et/2u(t)
Solution
Ð1
(i) yðt Þ ¼ xðτÞhðt τÞ ¼ xðt( 1Þ
1 Ðt
Ð1 0 1dτ, t>0
¼ 1 uðτÞuðt τÞdτ ¼
( 0, t<0
t, t > 0
¼ ¼ tuðt Þ
0, t < 0
(ii) uðt aÞ∗uðt bÞ ¼ ðuðt Þ∗δðt aÞÞ∗ðuðt Þ∗δðt bÞÞ

¼ ðuðt Þ∗uðt ÞÞ∗ðδðt aÞ∗δðt bÞÞ
¼ ðuðt Þ∗uðt ÞÞ∗δðt a bÞ
Since u(t)*u(t) ¼ tu(t),
uðt aÞ∗uðt bÞ ¼ ðtuðt ÞÞ∗δðt a bÞ
¼ ðt a bÞuðt a bÞ
(iii) xðt Þ∗hðt Þ ¼ ½uðt þ 1Þ uðt 1Þ∗½uðt þ 1Þ uðt 1Þ
¼ uðt þ 1Þ∗uðt þ 1Þ uðt þ 1Þ∗uðt 1Þ
uðt 1Þ∗uðt þ 1Þ þ uðt 1Þ∗uðt 1Þ
¼ uðt þ 1Þ∗uðt þ 1Þ 2uðt þ 1Þ∗uðt 1Þ
þuðt 1Þ∗uðt 1Þ
¼ ðt þ 2Þuðt þ 2Þ 2tuðt Þ þ ðt 2Þuðt 2Þ
as shown in Figure 2.5
Figure 2.5 The x(t)*h(t)

convolution of x(t) and h(t)
-2 0 2 tt
Ð1
(iv) yð t Þ ¼ 1 xðτÞδðt τ 1Þ ¼ xðt 1Þ
Ð1 ðτ2Þ
¼ 1 e uðτ 2Þuðt τ þ 2Þdτ
( Ð tþ2
2 eðτ2Þ dτ, t > 0
¼
0, t<0
Letting τ 1 ¼ τ – 2,
( Ð tþ2 (
2 eτ1 dτ1 2 et , t>0
yðt Þ ¼ ¼
0 0, t<0
ð1
(v) yð t Þ ¼ xðτÞδðt τ 1Þ ¼ xðt 1Þ
1
Hence, y(t) is a shifted version of x(t) as shown in Figure 2.6.

Ð1
(vi) yð t Þ ¼ 1 xðτÞhðt τÞdτ
Ð1
1 uðτ Þe uðt τÞdτ
ðtτÞ
¼
Ð t ðtτÞ
¼ 0e dτ, t > 0,

¼ eðtτÞ=2 0 ¼ ð1 et Þ, t
t
> 0,
yð t Þ ¼ 0 t<0
Ð t ðtτÞ=2
(vii) yð t Þ ¼ dτ, 2 t 0,
0 2e

t=2

¼ 4 1e , 2 t 0,
Ð 2 ðtτÞ=2
yð t Þ ¼ 0 2e dτ, t 2,

4eðtτÞ=2 0 ¼ 4 eðt2Þ=2 et=2 ,
2
¼ t 2,
¼ 4et=2 ð e 1 Þ, t 2,
yð t Þ ¼ 0 t 0
Example 2.7 Consider LTI system with the impulse response h(t); for an input x(t),
the output y(t) is as shown in Figure 2.7.
Figure 2.6 The shifted y(t)

version of x(t)
1 2 4 5 t
Figure 2.7 Response y(t) y(t)

to the input x(t)
1 2 4 5 t
y(t-2)
t
1 2 4 5
Figure 2.8 Response y(t 2) to the input x(t 2)
Figure 2.9 Response y1(t) (t)

to the input x1(t)
t
1 2 4 5
Determine the output of the system for an input

x1(t) ¼ x(t) x(t 2))
Solution Since the system is LTI, for an input x(t 2), the output is y(t 2) as
shown in Figure 2.8.
The output y1(t) for the input x1(t) ¼ x(t) x(t 2) is given by
y1(t) ¼ y(t) y(t 2), which is shown in Figure 2.9.
Example 2.8 Consider a LTI system with input and output related through the
equation
ðt
yðt Þ ¼ eðtτÞ xðτ 3Þ dτ
1
(i) Determine the impulse response h(t) of the system.

(ii) Determine the output y(t) of the system for the input
x(t) ¼ u(t þ 1) u(t 3).
Solution
ðt
(i) yðt Þ ¼ eðtτÞ xðτ 3Þ dτ
1
Let τ 1 ¼ τ 3, then ðt
yð t Þ ¼ eðt3τ 1 Þ xðτ 1 Þ dτ 1
1
Hence,
h(t) ¼ e(t 3)u(t 3)
ðt
(ii) yð t Þ ¼ eðt3τ 1 Þ ½uðt τ 1 þ 1Þ uðt τ 1 3Þ dτ 1
1
ðt
¼ eðt3τ 1 Þ ½uðt τ 1 þ 1Þ uðt τ 1 3Þ dτ 1
3
x(t τ) and h(τ) are shown in Figure 2.10.

Using Figure 2.10, y(t) can be written as
8
> 0, t 2,
>
>ð
>
>
>
<
tþ1
eðτ 1 3Þ dτ1 ¼ 1 eðt2Þ , 2 < t 6,
yð t Þ ¼
>
>
3
ð tþ1
>
>
>
>
: eðτ 1 3Þ dτ ¼ eðt6Þ 1 e4 ,
1 t > 6:
t23
1
1
t-3 0 t+1 0
Figure 2.10 x(t τ) and h(τ)

2.3.2 Graphical Convolution
An understanding of graphical interpretation of convolution is very useful in com-

puting the convolution of more complex signals. The stepwise procedure for graph-
ical convolution is as follows:
Step 1: Make x(τ) fixed.
Step 2: Invert h(τ) about the vertical axis (t ¼ 0) to obtain h(τ).
Step 3: Shift the h(τ) along the τ axis by t0 seconds so that the shifted h(τ) is
representing h(t0 τ).
Step 4: The area under the product of x(τ) and h(t0 τ) is y(t0), the value of
convolution at t ¼ t0.
Step 5: Repeat steps 3 and 4 for different values of positive and negative to obtain y
(t) for all values of t.
Example 2.9 Graphically determine the continuous-time convolution of h(t) and x(t)
for the following:
(
1, 0 t 4
(i) xðt Þ ¼
0, otherwise
(
1, 0 t 4
hð t Þ ¼
0, otherwise
Solution
1
1
0 4 0 4
To compute y(t) ¼ x(t) * h(t), first h(τ) is to be obtained by inverting h(τ) about
the vertical axis. Then, the product of x(τ) and h(t τ) is formed, point by point, and
this product is integrated to compute y(t). Thus, the overlap area between the
rectangles forming x(τ) and h(t τ) is y(t).
Clearly, y(0) ¼ 0 because there is no overlap between the rectangles forming x(τ)
and h(t τ) at t ¼ 0. For 0 < t < 8, there is overlap between the rectangles forming x
(τ) and h(t τ). For t 8, there is no overlap, and hence, y(8) ¼ 0. These are
illustrated in Figure 2.11 with the final result for y(t). The shaded portion represents
the overlap area of the product x(τ) and h(t τ).
(0)= ( )h(0 − ) = 0
h( − )| =0 ( )
1
-4 0 4 8
h(1 − ) ( ) (1)= ( )h(1 − ) = 1
-4 -3 0 4 8
h(2 − ) ( ) (2)= ( )h(2 − ) = 2
-4 -2 0 4 8
h(3 − ) ( ) (3)= ( )h(3 − ) = 3
-4 -1 0 4 8
h(4 − )
( ) (4)= ( )h(4 − ) = 4
-4 -1 0 4 8
Figure 2.11 Steps in the convolution and the final result

h(5 − )
( ) (5)= ( )h(5 − ) = 3
-4 -1 0 4 8
h(2 − ) ( ) (2)= ( )h(2 − ) = 2
-4 -2 0 4 8
h(3 − ) ( ) (3)= ( )h(3 − ) = 3
-4 -1 0 4 8
h(4 − )
( ) (4)= ( )h(4 − ) = 4
-4 -1 0 4 8
h(5 − )
( ) (5)= ( )h(5 − ) = 3
-4 -1 0 4 8
Figure 2.11 (continued)

h(6 − )
( )
(6)= ( )h(6 − ) = 2
1
-4 -1 0 4 8
h(7 − )
( )
(7)= ( )h(7 − ) = 1
1
-4 -1 0 4 8
h(8 − )
(8)= ( )h(8 − ) = 0
( )
1
-4 -1 0 4 8
y(t)
2 4 6 8 t
Example 2.10 Determine graphically y(t) ¼ x(t) * h(t) for the following x(t) and h(t)
shown.
1
1
-1 0 1 3
0 1 3
-1
Solution
1
1
-3 -1 0 1 3 0 1 3
-1
There is no overlap area between x(τ) and h(τ) at t ¼ 0, y(0) ¼ 0.

For 0 < t < 6, there is overlap between the rectangles forming x(τ) and h(τ). For
t 6, there is no overlap, and hence, y(6) ¼ 0. These are illustrated in Figure 2.12
with the final result for y(t). The shaded portion represents the overlap area of the
products x(τ) and h(t τ).
Example 2.11 Consider the RC circuit shown in Figure 2.13. Determine the Vout(t)
for Vin(t) ¼ u(t 1) u(t 2), and assume the time constant RC ¼ 1 sec. Assume
the capacitor is initially discharged.
Solution The impulse response of the RC low-pass filter is
t
Ð 1 hðtÞ ¼ e uðtÞ
V out ðt Þ ¼ 1 V in ðτÞhðt τÞ ¼ V in ðt Þ∗hðtÞ
h(1 − ) ( )
(1)= ( )h(1 − ) = 1
1
-2 -1 0 3
h(2 − )
( )
-1
(2)= ( )h(2 − ) = 2
1
h(1 − )
-1 0
3
-1
h(2 − )
( ) h(3 − )
(3)= ( )h(3 − ) = 1 − 1=0
1
-1 0
3
-1
h(3 − )
( ) h(4 − )
(4)= ( )h(4 − ) = − 2
1
-1 0 3 5
-1
h(4 − )
Figure 2.12 Steps in the convolution and the final result
( ) h( 5 − )
( 5) = ( ) h ( 5 − ) = − 1
1
0 4 6
-1
( ) h( 6 − )
( 6) = ( ) h ( 6 − ) = 0
-1 0 4 6 7
-1
h( 6 − )
y(t)
2 4 6 8 t
-2
The steps involved in the convolution are illustrated in Figure 2.14.

V out ðt Þ ¼ 0, t<1
Figure 2.13 RC circuit
Figure 2.14 Illustration of steps in the convolution
Ðt
V out ðt Þ ¼ eðtτ Þ dτ, 1 t 2,
1
t

¼ eðtτÞ 1 ¼ 1 eðt1Þ , 1 t 2,
Ð
V out ðt Þ ¼ 12 eðtτ Þ dτ, 2 t,
2

¼ eðtτÞ 1 ¼ eðt2Þ eðt1Þ , 2 t:

Example 2.12 If (t) ¼ x(t) * h(t), then show that
y(2t) ¼ 2x(2t) * h(2t)
Solution ð1
yð2t Þ ¼ xð2t τÞhðτÞdτ
1
τ
Letting τ1 ¼ , we have
2
Ð1 Ð1
yð2t Þ ¼ 1 xð2t 2τ1 Þhð2τ1 Þ2dτ1 ¼ 2 1 xð2t 2τ1 Þhð2τ1 Þdτ1
¼ 2xð2t Þ∗hð2t Þ
Example 2.13 If x(t) and h(t) are odd signals, then show that
y(t) ¼ x(t) * h(t) is an even signal.
0.7
0.6
0.5
Amplitude
0.4
0.3
0.2
0.1
0
0 0.5 1 1.5 2 2.5 3 3.5 4
Time
Figure 2.15 Times versus Vout(t)
Solution y(t) ¼ x(t) * h(t)
yðt Þ ¼ xðt Þ∗hðt Þ

Ð1
¼ 1 xððt τÞÞhðτÞdτ
Ð1
¼ 1 xðt þ τÞhðτÞdτ
Since x(t) and h(t) are odd signals,

Ð1
yðtÞ ¼ 1 xðt τÞhðτÞdτ
¼ yð t Þ
Hence, y(t) is even because y(t) ¼ y(t).

Example 2.14 Consider an LTI system with the impulse response h(t) ¼ etu(t).
Find the system response for the input x(t) ¼ sin2tu(t).
Solution Ð1
yð t Þ ¼ 1 xðτÞhðt τÞdτ
Ð1
¼ 0 sin ð2τÞeðtτÞ dτ
Ð1
¼ sin ð2τÞeðtτÞ dτ
0
h
t Ð1 i
¼ sin ð2τÞeðtτÞ τ¼0 0 2 cos ð2τÞeðtτÞ dτ uðtÞ
Ð1
¼ sin ð2t Þuðt Þ uðt Þ 0 2 cos ð2τÞeðtτÞ dτ
Hence,
Ð1 Ð1
0 sin ð2τÞeðtτÞ dτ ¼ sin ð2t Þuðt Þ uðt Þ 0 2 cos ð2τÞeðtτÞ dτ

t Ð1
¼ sin ð2t Þuðt Þ uðt Þ 2 cos ð2τÞeðtτÞ τ¼0 0 4 sin ð2τÞeðtτÞ dτ
Ð1
¼ sin ð2t Þuðt Þ ð2 cos ð2t Þ þ et Þuðt Þ 0 4 sin ð2τÞeðtτÞ dτ
The above equation can be rewritten as

ð1
5 sin ð2τÞeðtτÞ dτ ¼ ½ sin ð2t Þ 2 cos ð2t Þ þ et uðt Þ
0
Therefore,
ð1
1
yð t Þ ¼ sin ð2τÞeðtτÞ dτ ¼ ½ sin ð2t Þ 2 cos ð2t Þ þ 2et uðt Þ
0 5
Example 2.15 If the response of an LTI system to input x(t) is the output y(t),
dy
show that the response of the system to dx
dt is dt , and using this result determines the
impulse response of an LTI system having the response y(t) ¼ sin2t for an input
x(t) ¼ e4tu(t).
Solution
yðtÞ ¼ xðt Þ∗hðt Þ
Ð1
¼ 1 hðτÞxðt τÞdτ
Differentiating both sides with respect to t,

ð1
dy dx
¼ hðτÞ ðt τÞdτ
dt 1 dt
dy dx
¼ hðtÞ∗
dt dt
For given y(t), dy dx
dt ¼ 2 sin 2t, and for given xðt Þ, dt ¼ 4e
4t
þ e4t δðt Þ:
From sampling property impulse function, it is known that x(t) δ (t) ¼ x(0)δ(t).
Since e4tδ(t) ¼ e0δ(t) ¼ δ(t), dx
dt can be written as
dx
¼ 4e4t þ δðt Þ
dt
Let x1(t) ¼ 4e4tu(t), then by homogeneity, the corresponding output
y1 ðt Þ ¼ 4yðt Þ ¼ 4 sin 2t
Let x2 ðt Þ ¼ dx
dt ¼ 4e
4t
þ δðt Þ the corresponding output
y2 ðt Þ ¼ 2 sin 2t
As it is LTI system, if (x1(t) þ x2(t)) is the input to the system, the corresponding
output is(y1(t) þ y2(t))
since x1(t) þ x2(t) ¼ 4e4t 4e4t + δ(t) ¼ δ(t), the impulse response h
(t) ¼ y1(t) þ y2(t) ¼ 4 sin 2t þ 2 sin 2t
Example 2.16 Consider a continuous-time LTI system with the unit step response s(t):
(i) Deduce that the response y(t) of the system to the input x(t) is
ð1
dxðτÞ
yð t Þ ¼ sðt τÞdτ
1 dt
and also show that

ð1
dxðτÞ
xð t Þ ¼ uðt τÞdτ:
1 dt
(ii) Determine the response of an LTI system with step response

sðt Þ ¼ e2t et þ 1 uðt Þ
to an input x(t) ¼ etu(t).

Solution
(i) The step response s(t) is
sðt Þ ¼ hðt Þ∗uðt Þ

Ð1
¼ 1 hðτÞuðt τÞdτ
Ðt
¼ 1 hðτÞdτ
Consider the following equivalence.
x(t) h(t) y(t) x(t) h(t) y(t)

=
From the above equivalence, we obtain

h(t) y(t)
Thus, ð t
dxðt Þ
yð t Þ ¼ ∗ hðτÞdτ
dt 1
dxðt Þ
¼ ∗sðt Þ
dt
Since y(t) ¼ x(t) ∗ h(t) and if h(t) ¼ δ(t) and y(t) ¼ ð tx(t) as x(t)∗ δ(t) ¼ x(t),thus,
dxðt Þ
putting h(t) ¼ δ(t) in yð t Þ ¼ ∗ hðτÞdτ , it becomes
Ð dt 1
t
xðt Þ ¼ dxdtðtÞ ∗ 1 δðτÞdτ
Ð
t
1 δ ð τ Þdτ ¼ uð t Þ
Since dxðt Þ
xðt Þ ¼ ∗uðt Þ
dt
(ii) xðt Þ ¼ et uðt Þ
dxðt Þ
¼ et uðt Þ þ δðtÞet
dt
δðtÞet ¼ δðtÞe0 ¼ δðtÞ

Since dxðt Þ
¼ et uðt Þ þ δðtÞ
dt

sðt Þ ¼ e2t et þ 1 uðt Þ
dxðtÞ
yðtÞ ¼ ∗sðtÞ
dt
Ð 1 dxðτÞ
¼ 1 sðt τÞdτ
Ð 1 dt τ
¼ 1 fe uðτÞ þ δðτÞgfe2ðtτÞ eðtτÞ þ 1guðt τÞdτ
Ðt
¼ 0 et fe2ðtτÞ eðtτÞ þ 1gdτ þ fe2t et þ 1guðtÞ
Ðt
¼ 0 fe2tþ3τ etþ2τ þ eτ gdτ þ sðtÞ
1 1
¼ e2t ðe3t 1Þ et ðe2t 1Þ þ ðet 1Þ þ sðtÞ
3 2
1 2t 1 t 5 t
¼ e þ e þ e 1 þ sðtÞ
3 2 6
Example 2.17 Consider h(t) be the triangular pulse and x(t) be the unit impulse train
as shown in Figure 2.16. Determine y(t) ¼ x(t) * h(t) for T ¼ 2.
h (t) x (t)
1
1
t -2T -T 0 T 2T t
-1 1
Figure 2.16 x(t) and h(t) of Example 2.17
y(t)
-3 -2 -1 0 1 2 3
t
Figure 2.17 Time versus y(t)
Solution
X
1
xðt Þ ¼ δðt nT Þ
n¼1
X
1 X
1
yðt Þ ¼ xðt Þ∗hðt Þ ¼ hðt Þ∗δðt nT Þ ¼ hðt nT Þ
n¼1 n¼1
2.3.3 Computation of Convolution Integral Using MATLAB
MATLAB provides a function conv() that performs a discrete-time convolution of

two discrete-time sequences. A new function convint() that uses conv() to numeri-
cally integrate the continuous-time convolution is as follows.
function[y,ty]=convint(x,tx,h,th)
%Inputs:
%x is the input signal vector
%tx is the times of the samples in x
%h is the impulse response vector
%th is times of the samples in h
%outputs:
%y is the output signal vector,
%length(y)=length(x)+length(h)-1
%ty is the time of the samples in y
dt=tx(2)-tx(1);
y=conv(x,h)*dt;
ty=(tx(1)+th(1))+[0:(length(y)-1)]*dt;
The computation of convolution of continuous-time signals using MATLAB is

illustrated through the following numerical examples.
Example 2.18 (i) Verify the result of Example 2.9 using MATLAB.
(ii) Verify the result of Example 2.10 using MATLAB.
Solution (i) The following MATLAB program 2.1 is used to compute the convo-
lution of x(t) and h(t) of Example 2.9.
Program 2.1
clc;
clear all;
close all;
tx=[0:0.01:4];
x=ones(1,length(tx));
th=[0:0.01:4];
h=ones(1,length(th));
[y ty]=convint(x,tx,h,th);
figure;
plot(ty,y);
xlabel('Time')
ylabel('Amplitude');
axis([0 8 0 4]);
The output y(t) ¼ x(t) * h(t) of the above program is shown in Figure 2.18. It is
observed to be the same as that shown in Figure 2.11. Thus, it is verified.
(ii) The following MATLAB program 2.2 is used to compute the convolution of x(t)
and h(t) of Example 2.10.
3.5
2.5
Amplitude
1.5
0.5
0
0 1 2 3 4 5 6 7 8
Time
1.5
0.5
Amplitude
-0.5
-1
-1.5
-2
0 1 2 3 4 5 6

Program 2.2
clc;
clear all;
close all;
tx=[0:0.01:3];
tx1=[0:0.01:1];
x=[zeros(1,length(tx1)) ones(1,(length(tx)-length(tx1)))];
th1=[-1:0.01:1];
th2=[1.01:0.01:3];
h=[ones(1,length(th1)) -1*ones(1,length(th2))];
th=[-1:0.01:3];
figure;
plot(ty,y);
xlabel('Time')
axis([0 6 -2 2]);
The output y(t) ¼ x(t) * h(t) of tie above program is shown in Figure 2.19 It is
observed to be the same as that shown in Figure 2.12. Thus, it is verified
Example 2.19 Consider the RC circuit of Example 2.11 with time constant
RC ¼ 13 sec . Determine the Vout(t) using MATLAB for Vin(t) ¼ (u(t 3) u
(t 5)). Assume the capacitor is initially discharged.
Solution The impulse response of the RC circuit is given by
ð1
1 t=RC
hð t Þ ¼ e uðtÞ ¼ 3e3t uðtÞV out ðt Þ ¼ V in ðτÞhðt τÞ ¼ V in ðt Þ∗hðtÞ
RC 1
The following MATLAB program 2.3 is used to compute the convolution of vin(t)
and h(t).
Program 2.3
clc;
clear all;
close all;
tx=[0:0.01:5];
tx1=[0:0.01:3];
x=[zeros(1,length(tx1)) ones(1,(length(tx)- length(tx1)))];
th=[0:0.01:5];
h =(3)* exp(-3*th);
figure;
plot(ty,y);
xlabel('Time')
0.12
0.1
0.08
Amplitude
0.06
0.04
0.02
0
0 1 2 3 4 5 6 7 8 9 10
Time
Figure 2.20 Time versus Vout(t)
(a) (b)
Figure 2.21 (a) Cascade connection of two systems. (b) Equivalent system
The output Vout(t) ¼ Vin(t) * h(t) of the above program is shown in Figure 2.20.
2.3.4 Interconnected Systems
2.3.4.1 Cascade Connection of Systems
The system shown in Fig. 2.21 is formed by connecting two systems in cascade. The
impulse responses of the systems are given by h1(t) and h2(t), respectively. Let y(t)
be the output of the first system. By the definition of convolution
y1 ðt Þ ¼ xðt Þ∗h1 ðt Þ ð2:37Þ
Then, the output of the overall system y(t) is given by
yðt Þ ¼ y1 ðt Þ∗h2 ðt Þ ¼ ½xðt Þ∗h1 ðt Þ∗h2 ðt Þ ð2:38Þ

By the associativity property of convolution, Eq. (2.38) can be rewritten as
yðt Þ ¼ y1 ðt Þ∗h2 ðt Þ ¼ xðt Þ∗½h1 ðt Þ∗h2 ðt Þ ð2:39Þ
Hence, the impulse response of the overall system is given by

Ð1
hðt Þ ¼ h1 ðt Þ∗h2 ðt Þ ¼ 1 h1 ðτÞ h2 ðt τÞdτ
Ð1 ð2:40Þ
¼ 1 h2 ðτÞ h1 ðt τÞdτ
2.3.4.2 Parallel Connection of Two LTI Systems
The system shown in Fig. 2.22 is formed by connecting two systems in parallel. The
impulse responses of the systems are given by h1(t) and h2(t), respectively. Let y1(t)
and y2(t) be the outputs of the first system and second system, respectively. By the
definition of convolution

Then, the output of the overall system y(t) is given by
yðt Þ ¼ y1 ðt Þ þ y2 ðt Þ ¼ xðt Þ∗h1 ðt Þ þ xðt Þ∗h2 ðt Þ ð2:43Þ
By the distributive property of convolution, Eq. (2.43) can be rewritten as
yðt Þ ¼ y1 ðt Þ þ y2 ðt Þ ¼ xðt Þ∗½h1 ðt Þ þ h2 ðt Þ
Hence, the impulse response of the overall system is given by
hð t Þ ¼ h1 ð t Þ þ h2 ð t Þ ð2:44Þ
(a) (b)
Figure 2.22 (a) Parallel connection of two systems. (b) Equivalent system
Example 2.20 An LTI system consists of two subsystems in cascade. The impulse
responses of the subsystems are, respectively, given by
h1 ðt Þ ¼ e3t uðt Þ; h2 ðt Þ ¼ et uðt Þ
Find the impulse response of the overall system.

Solution The overall impulse response of the system is given by
hðt Þ ¼ h1 ðt Þ∗h2 ðt Þ
Ð1
¼ 1 e3τ uðτÞeðtτÞ uðt τÞdτ
Ðt
¼ 0 e3τ eðtτÞ dτ
Ðt
¼ et 0 e2τ dτ
1

¼ et e2t uðtÞ
2
2.3.5 Periodic Convolution
If the signals x1(t) and x2(t) are periodic with common period T, it can be easily
shown that the convolution of x1(t) and x2(t) does not converge. In such a case, the
periodic convolution of x1(t) and x2(t) is defined as
O ðT
yðt Þ ¼ x1 ðt Þ x2 ð t Þ ¼ x1 ðτÞx2 ðt τÞdτ ð2:45Þ
0
Example 2.21 Let y(t) be the periodic convolution of x1(t) and x2(t). Show that
yð t Þ ¼ yð t þ T Þ
ðT
Solution (i) yðt þ T Þ ¼ x1 ðτÞx2 ðt þ T τÞdτ
0
Since x2(t) is periodic with period T, x2(t þ T τ) ¼ x2(t τ)

ÐT ÐT
0 x1 ðτÞx2 ðt þ T τÞdτ ¼ 0 x1 ðτÞx2 ðt τÞdτ
yðt þ T Þ ¼ yðt Þ:
2.4 Properties of Linear Time-Invariant Continuous-Time System 77
2.4 Properties of Linear Time-Invariant Continuous-Time

System
2.4.1 LTI Systems With and Without Memory
The output y(t) of a memoryless system depends only on the present input x(t). If the
system is LTI, then the relationship between the y(t) and x(t) for a memoryless
system is
yðt Þ ¼ cxðt Þ ð2:46Þ
where c is an arbitrary constant. Since the output of a continuous-time system can be

written as
ð1
yð t Þ ¼ hðτÞxðt τÞdτ
1
the corresponding impulse response is h(t) ¼ cδ(t).

Thus, a continuous-time system is memoryless if and only if
hðt Þ ¼ cδðtÞ ð2:47Þ
2.4.2 Causality for LTI Systems
The output of a continuous-time system can be written as

ð1
1
Since the impulse response h(τ) ¼ 0 for τ < 0 for a causal continuous-time system,
the output of a causal system can be expressed by the following convolution integral:
ð1
yð t Þ ¼ hðτÞxðt τÞdτ ð2:48Þ
0
2.4.3 Stability for LTI Systems
A continuous-time system is BIBO stable if and only if the impulse response is

absolutely integrable, that is,
ð1
hðτÞdτ < 1 ð2:49Þ
1
Example 2.22 Check stability of continuous-time system having the following

impulse responses:
(i) h(t) ¼ etu(t)
(ii) h(t) ¼ et cos (2t)u(t)
(iii) h(t) is periodic and nonzero
Solution
ð1 ð1
(i) jhðτÞjdτ ¼ eτ dτ ¼ 1
1 0
Indicating that h(t) is absolutely integrable and, hence, h(t) is the impulse
response of a stable system,
ð1 ð1
(ii) jhðτÞjdτ ¼ eτ j cos ð2τÞjdτ
1 0
τ
Since e |cos (2τ)| is exponentially decaying for 0 t 1 , h(t) is absolutely
summable, and hence, h(t) is the impulse response of a stable system.
(iii) If h(t) is periodic with period T, then
ð1 ð T=2
jhðτÞjdτ ¼ N jhðτÞjdτ
1 T=2
where N ! 1 ð1
Since h(t) is nonzero jhðτÞjdτ ! 1, hence, h(t) is absolutely summable and,
1
hence, h(t) is the impulse response of an unstable system.
Example 2.23 Determine if each of the following system is causal or stable:
(i) h(t) ¼ etu(t 1)
(ii) h(t) ¼ etu(t þ 1)
(iii) h(t) ¼ e2tu(t þ 10)
(iv) h(t) ¼ tetu(t)
(v) h(t) ¼ e+tu(t 1)
(vi) h(t) ¼ e2|t|
Solution ð1
(i) Causal because h(t) ¼ 0 for t < 0. Stable because jhððτÞjdτ < 1.
1 1
(ii) Not causal because h(t) 6¼ 0 for t < 0. Unstable because jhðτÞjdτ ¼ 1.
ð 1 1
(iii) Not causal because h(t) 6¼ 0 for t < 0. Stable because jhðτÞjdτ < 1.
1
ð1
(iv) Causal because h(t) ¼ 0 for t < 0. Stable because jhðτÞjdτ < 1.
ð1
1
(v) Not causal because h(t) 6¼ 0 for t < 0. Stable because jhðτÞjdτ < 1.
1ð
1
(vi) Not causal because h(t) 6¼ 0 for t < 0. Unstable because jhðτÞjdτ ¼ 1.
1
2.4.4 Invertible LTI System
A system is invertible if its input x(t) can be recovered from its output y(t) ¼ x(t) * h
(t). The cascade of an LTI system having impulse response h(t) with a LTI inverse
system having impulse response g(t) ¼ h1(t) is shown in Figure 2.23.
The process of recovering x(t) from x(t) * h(t) is called deconvolution as it
corresponds to reverse of the convolution operation.
The overall impulse response of the invertible system shown in Figure 2.23 is the
convolution of h(t) and g(t). It is required that the output of the invertible system is
equivalent to the input:

xðt Þ∗ hðt Þ∗h1 ðt Þ ¼ xðt Þ ð2:50Þ
implying that
hðt Þ∗h1 ðt Þ ¼ δðt Þ ð2:51Þ
As an example, it is verified that the inverse system for a continuous-time

integrator is a differentiator as follows:
ð t
d
xðτÞdτ ¼ xðtÞ ð2:52Þ
dt 1
Hence, the input-output relation for the inverse system shown in Figure 2.24 is
dyðt Þ
xðtÞ ¼ ð2:53Þ
dt
x(t) y(t) x(t)

h(t) (t)
Figure 2.23 Cascade connection of an LTI system and its inverse

x(t) y(t) x(t)

ò
Figure 2.24 Input-output relation for the inverse system
Example 2.24
(i) An echo of an auditorium can be modeled as a LTI system with an impulse

response consisting of a train of impulses:
X
1
hð t Þ ¼ hk δðt kT Þ
k¼0
The inverse LTI system with impulse response g(t) is
yðt Þ∗gðt Þ ¼ xðt Þ
where g(t) is also an impulse train that is modeled as
X
1
gð t Þ ¼ gk δðt kT Þ
k¼0
Obtain the relationship between hkand gk.

Solution
y(t) ¼ x(t) * h(t) and x(t) ¼ g(t) *y(t), then
gðt Þ∗hðt Þ ¼ δðt Þ:
However,
Ð1 X1 X
1
gðt Þ∗hðt Þ ¼ 1 gk δðt τ kT Þ hm δðτ mT Þdτ
k¼0 m¼0
1 X
X 1
¼ gk hm δðt τ kT Þδðt ðm þ kÞT Þ
k¼0 m¼0
Let n ¼ m þ k, then m ¼ n-k, and g(t) * h(t) can be rewritten as

!
X
1 X
1
gðt Þ∗hðt Þ ¼ gk hnk δðt nT Þ
n¼0 k¼0
Hence,
(
X
1 1, n ¼ 0,
gk hnk ¼
k¼0 0, n 6¼ 0:
Implying that
g0 h0 ¼ 1,
g0 h1 þ g1 h0 ¼ 0,
g0 h2 þ g1 h1 þ g2 h0 ¼ 0,
and so on, solution of the above equations leads to
1
g0 ¼ ,
h0
g0 h1 h1 h1
g1 ¼ ¼ ¼ 2 ,
h0 h0 h0 h0
! !
1 1 h1 1 h2 h21
g0 h2 þ g1 h1 þ g2 h0 ¼ h2 2 h1 ¼
h0 h0 h0 h0 h0 h20
(ii) Consider the following echo generation model characterized by
yðt Þ ¼ xðt Þ þ ayðt T Þ
where 0 < a < 1 and T is delay.

Construct the corresponding inverse system and obtain its impulse response.
Solution Assuming y(t) ¼ 0 for t < 0 and x(t) ¼ 0 for t < 0, the impulse response h(t)
of the echo generation system is given by
X
1
hðtÞ ¼ ak δðt kTÞ
k¼0
Thus, h0 ¼ 1, h1 ¼ a, hi ¼ 0 for i > 2.

The inverse system has to obtain x(t) from the output y(t). Hence, the inverse
system is characterized by
xðt Þ ¼ yðt Þ ayðt T Þ
and represented as depicted in Figure 2.25.

Figure 2.25 An inverse

system
y(t) x(t)
Delay
T -a
The impulse response g(t) of the inverse system is given by
X
1
gð t Þ ¼ ðaÞk δðt kT Þ
k¼0
Hence, g0 ¼ 1, g1 ¼ a.
Example 2.25 Check y(t) ¼ x(2t) for causality and invertibility.
Solution
yðt Þ ¼ xð2t Þ
At time t ¼ 1
yð1Þ ¼ xð2Þ
indicating that the value of y(t) at time t ¼ 1 depends on x(t) at a time t ¼ 2.

Therefore, y(t) ¼ x(2t) is not causal.
y(t) is invertible;
xðt Þ ¼ yðt=2Þ
2.5 Systems Described by Differential Equations
2.5.1 Linear Constant-Coefficient Differential Equations
A general Nth-order linear constant-coefficient differential equation is given by

XN dn yðt Þ X M d n xð t Þ
a
n¼0 n n ¼ b n ð2:54Þ
dt k¼0 dt n
where coefficients an and bn are real constants. The order N refers to the highest
derivative of y(t) in Eq. (2.54). For example, consider the RC circuit considered in
Example 2.11, the input x(t) and the output y(t) ¼ vo (t). If the current flowing
through the RC circuit is i(t), using Kirchhoff’s voltage law, we write
2.5 Systems Described by Differential Equations 83
ð
1
xðt Þ þ Riðt Þ þ iðt Þdt ¼ 0 ð2:55Þ
c
which can be rewritten as
ð
1
Riðt Þ þ iðt Þdt ¼ xðt Þ ð2:56Þ
c
Since iðt Þ ¼ c dvdt0 ðtÞ ¼ c dydtðtÞ , substituting iðt Þ ¼ c dydtðtÞ in Eq. (2.56), we obtain the
following first-order constant-coefficient differential equation
dyðt Þ 1 1
þ yð t Þ ¼ xð t Þ ð2:57Þ
dt RC RC
relating the voltage across the capacitor y(t) and the input x(t).
Example 2.26 Find the differential equation relating the current y(t) and the input
voltage x(t) for the RLC circuit shown in Figure 2.26 assuming R ¼ 3 Ohms, L ¼ 1
Henry, and C ¼ 12 Farad:
Solution Using Kirchhoff’s voltage law, we write the following loop equation for
the given RLC circuit:
ð
dyðt Þ 1
xðt Þ þ Ryðt Þ þ L þ yðt Þdt ¼ 0
dt c
For R ¼ 3, L ¼ 1, and C ¼ 12 , the above equation becomes
ð
dyðt Þ
þ 3yðtÞ þ 2 yðtÞdt ¼ xðtÞ
dt
Differentiating this equation, we obtain
d 2 yð t Þ dyðt Þ dxðt Þ
2
þ3 þ 2yðtÞ ¼
dt dt dt
Figure 2.26 RLC circuit

Figure 2.27 Operational amplifier circuit
Example 2.27 Find the differential equation relating the input voltage V i ðt Þ and the
output voltage V o ðt Þ for the operational amplifier circuit shown in Figure 2.27.
Solution
‘I r1 ¼ I r2 þ I c1 ; I r2 ¼ I c2 ; V 2 ¼ V 0
Rewriting the current node equations, we get

Vi V1 V1 V0 d
¼ þ c1 ð V 1 V 0 Þ
r1 r2 dt
dV 1 dV 0
r 1 r 2 c1 r 1 r 2 c1 þ ðr 1 þ r 2 ÞV 1 r 1 V 0 ¼ V i r 2
dt dt
V1 V0 dV 0
¼ c2 ;
r2 dt
dV 0
V 1 ¼ r 2 c2 þ V0
dt
Substituting the above equation for V 1 , the input-output relation can be written as
d2 V 0 dV 0
c2 c1 r 2 r 1 2
þ c2 ðr 1 þ r 2 Þ þ V0 ¼ Vi
dt dt
which is rewritten as
d2 V 0 r 1 þ r 2 dV 0 V0 Vi
2
þ þ ¼
dt r r c
1 2 1 dt r r c c
1 2 1 2 r 1 2 c1 c2
r
2.5.2 The General Solution of Differential Equation
The general solution of Eq. (2.54) for a particular input x(t) is given by
yð t Þ ¼ yc ð t Þ þ yp ð t Þ ð2:58Þ
where yc(t) is called the complementary solution and yp(t) is called the particular
solution. The complementary solution yc(t) is obtained by setting x(t) ¼ 0 in
Eq. (2.54). Thus yc(t) is the solution of the following homogeneous differential
equation
XN dn yðt Þ
a
n¼0 n
¼0 ð2:59Þ
dt n
Example 2.28 Consider the RC circuit of Example 2.11 with time constant
RC ¼ 1 sec. Determine the voltage across the capacitor for an input x(t) ¼ e2tu(t).
Assume the capacitor is initially discharged.
Solution As the time constant RC ¼ 1, the input x(t) and the output yðt Þ ¼ V 0 ðt Þ of
the RC circuit are related by
dyðt Þ
þ yðt Þ ¼ e2t uðtÞ yð0Þ ¼ 0
dt
The particular solution for the exponential input is of the form
yp ðt Þ ¼ Ae2t t>0
Substituting yp(t) in the above differential equation, we get
2Ae2t þ Ae2t ¼ e2t t>0
Solving for A, we obtain A ¼ 1 and
yp ðt Þ ¼ e2t
To obtain complementary solution, let us assume
yc ðt Þ ¼ Bekt
Substituting this into
dyc ðt Þ
þ yc ð t Þ ¼ 0
dt
yields
Bkekt þ Bekt ¼ 0
ðk þ 1ÞBekt ¼ 0
Thus, k ¼ 1 and
yc ðt Þ ¼ Bet
Now,
yðt Þ ¼ yc ðt Þ þ yp ðt Þ ¼ Bet e2t
at t ¼ 0 y(0) ¼ B – 1.
Since the capacitor is initially discharged, y(0) ¼ 0, and we obtain B ¼ 1.
Hence, the voltage across the capacitor is given by

yðt Þ ¼ et e2t uðt Þ
2.5.3 Linearity
The system specified by Eq. (2.54) is linear only if all of the initial conditions
are zero.
For instance, in the Example 2.11, if the capacitor is not assumed to be discharged
initially, then yð0Þ ¼ V 0 ð0Þ 6¼ 0
A linear system has the property that zero input produces zero output.
However, if we let x(t) ¼ 0, then
yðt Þ ¼ yc ðt Þ ¼ yð0Þet ð2:60Þ
Thus, this system is nonlinear if y(0) 6¼ 0.

If the capacitor is assumed to be discharged initially, then yð0Þ ¼ V 0 ð0Þ ¼ 0:
Then for x(t) ¼ 0,
yðt Þ ¼ yc ðt Þ ¼ 0 ð2:61Þ
the system is linear
2.5.4 Causality
A linear system described by Eq. (2.54) is causal when it is initially relaxed. It

implies that if x(t) ¼ 0 for t t0, then y(t) ¼ 0 for t t0, thus, the response for t > to
with the initial conditions
dyðt 0 Þ dN1 yðt 0 Þ

yð t 0 Þ ¼ ... ¼ ¼0 ð2:62aÞ
dt dt N1

dn yðt 0 Þ dn yðt Þ
¼ ð2:62bÞ
dt n dt n t¼t0
2.5.5 Time-Invariance
For a linear causal system, initial rest also implies time-invariance.

For example, consider the system described by
dyðt Þ
þ yðt Þ ¼ xðt Þ yð0Þ ¼ 0 ð2:63Þ
dt
Let y1(t) be the response to an input x1(t) and
x1 ðt Þ ¼ 0 t 0 ð2:64Þ
so that
dy1 ðt Þ
þ y1 ðt Þ ¼ x1 ðt Þ ð2:65Þ
dt
and
y 1 ð 0Þ ¼ 0 ð2:66Þ
Now, let x2(t) ¼ x1(t τ) and y2 (t) be the corresponding response. From
Eq. (2.64), we get
x2 ð t Þ ¼ 0 tτ ð2:67Þ
Then y2(t) should satisfy
dy2 ðt Þ
þ y2 ðt Þ ¼ x2 ðt Þ ð2:68Þ
dt
and
y2 ð τ Þ ¼ 0 ð2:69Þ
From Eq. (2.65), we write
dy1 ðt τÞ
þ y1 ð t τ Þ ¼ x1 ð t τ Þ ¼ x2 ð t Þ
dt
By letting y2(t) ¼ y1(t τ), we obtain from Eq. (2.66)
y2 ðτÞ ¼ y1 ðτ τÞ ¼ y1 ð0Þ ¼ 0:
Eqs. (2.68) and (2.69) are satisfied and thus the system is time invariant.
2.5.6 Impulse Response
The impulse response h(t) of the continuous-time LTI system described by

Eq. (2.54) satisfies the differential equation
XN d n hð t Þ X M dn δðt Þ
a
n¼0 n
þ b n ð2:70Þ
dt n n¼0 dt n
with the initial rest condition.
For example, let us consider the RC circuit of Example 2.11 with the time
constant RC ¼ 1 sec described by the following differential equation:
dyðt Þ
þ yðt Þ ¼ xðt Þ
dt
The impulse response h(t) should satisfy the differential equation
dhðt Þ
þ hðt Þ ¼ δðt Þ
dt
Then, the complimentary solution hc(t) satisfies

dhc ðt Þ
þ hc ð t Þ ¼ 0
dt
yc ðt Þ ¼ Bekt

dhc ðt Þ
þ hc ð t Þ ¼ 0
dt
yields
Bkekt þ Bekt ¼ 0
ðk þ 1ÞBekt ¼ 0
Thus, k ¼ 1 and
hc ðt Þ ¼ Bet uðtÞ
The particular solution hp(t) is zero since hp(t) cannot contain δ(t). Thus,
hðt Þ ¼ Bet uðtÞ
To find the constant B, substituting h(t) ¼ Betu(t)into
dhðt Þ
dt
yields
duðtÞ
Bet uðtÞ þ Bet þ Bet uðtÞ ¼ δðtÞ
dt
duðtÞ
Bet ¼ Bet δðtÞ ¼ δðtÞ
dt
such that B ¼ 1 and hence
hðt Þ ¼ et uðtÞ:
Example 2.29 Consider the RL circuit shown in Figure 2.28:

(i) Determine the impulse response.
(ii) Determine the step response.
Solution Writing the loop equation using Kirchhoff’s voltage law assuming i(t) is
the current flowing through the circuit, we obtain
diðt Þ
xðtÞ þ RiðtÞ þ L ¼0
dt
But iðtÞ ¼ yRðtÞ :
Figure 2.28 RL circuit

Substituting iðtÞ ¼ yRðtÞ in the above equation, we get
L dyðtÞ
þ yðtÞ ¼ xðtÞ
R dt
dyðt Þ R R
þ yðtÞ ¼ xðt Þ
dt L L
(i) The impulse response h(t) should satisfy the differential equation
dhðt Þ R R
dt L L
Then, the complimentary solution hc(t) satisfies
dhc ðtÞ R
þ hc ðtÞ ¼ 0
dt L
yc ðt Þ ¼ Bekt
dhc ðtÞ R
þ hc ðtÞ ¼ 0
dt L
yields
R
Bkekt þ Bekt ¼ 0
L
R
kþ Bekt ¼ 0
L
Thus, k ¼ RL and

R
hc ðt Þ ¼ BeLt uðtÞ
The particular solution hp(t) is zero since hp(t)) cannot contain δ(t). Thus,
R
hðt Þ ¼ BeLt uðtÞ
R
To find the constant B, substituting hðt Þ ¼ BeLt uðtÞinto
dhðt Þ R R
dt L L
yields
R R R duðtÞ R R R
B eLt uðtÞ þ BeLt þ B eLt uðtÞ ¼ δðtÞ
L dt L L
L t duðtÞ Lt R R
R R
Be ¼ Be δðtÞ ¼ δðtÞ
dt L L
such that B ¼ RL and hence
R R t
hð t Þ ¼ e L uðtÞ:
L
(ii) The step response s(t) is given by
Ðt ÐtR R
sðtÞ ¼ hðτÞdτ ¼ 0 eLτ dτ
0
L
Lτ t
R R
Lt
¼ e j0 ¼ 1 e uðtÞ
2.5.7 Solution of Differential Equations Using MATLAB
The response y(t) of a system described by Eq. (2.54) for an input x(t) can be
determined by using the MATLAB command dsolve(‘eqn1’,‘eqn2’, ...) which
accepts symbolic equations representing ordinary differential equations and initial
conditions. Several equations or initial conditions may be grouped together, sepa-
rated by commas, in a single input argument.
Example 2.30 Verify the result of Example 2.28 using MATLAB.
Solution The relation between the output y(t) and the input x(t) is related by
dyðt Þ
þ yðt Þ ¼ e2t uðtÞ yð0Þ ¼ 0
dt
The output response y(t) is determined and displayed by the MATLAB
commands
y = dsolve('Dy+y=exp(-2*t)', ,'y(0)=0', 't');

disp (['y(t) = (',char(y), ')u(t) ' ]);
The displayed output is

yðt Þ ¼ et e2t uðt Þ
Example 2.31 Consider the RLC circuit of Example 2.26 and determine the current
y(t) for the input voltage x(t) ¼ 10e3t u(t) where the initial inductor current is zero
and the initial capacitor voltage is equal to 5 volts.
Solution The relation between the output y(t) and the input x(t) is related by
d2 y dy dx
2
þ 3 þ 2y ¼
dt dt dt
dyðt Þ
yð0Þ ¼ 0, ¼ 5:
dt t¼0
The current y(t) is determined and displayed by the MATLAB commands
y = dsolve('D2y+3*Dy+2*y=-30*exp(-3*t)', 'y(0)=0',
'Dy(0)=5', 't');
disp (['y(t) = (', char(y), ')u(t) ' ]);
The displayed y(t) is

yðt Þ ¼ 25e2t 10et 15e3t uðt Þ
2.5.8 Determining Impulse Response and Step Response

for a Linear System Described by a Differential
Equation Using MATLAB
In general, for a system described by Eq. (2.54), the impulse response can be
determined by using the following MATLAB command:
h ¼ impluseðb; a; t Þ
For example, for the differential equation given by
d2 y dy
þ 3 þ 2y ¼ xðt Þ
dt 2 dt
we use the following MATLAB program 2.4 to determine impulse response and step
response.
2.6 Block-Diagram Representations of LTI Systems Described by Differential Equations 93
Program 2.4
clc;
clear all;
close all;
th=0:.01:4;
b = [1];
a = [1 3 2];
h=impulse(b,a,th);
tx=[0:0.01:4];
x=[ones(1,length(tx))];
figure,plot(th,h)
xlabel('Time')
figure, plot(ty,y);
xlabel('Time')
The impulse response and step response obtained from the above program are
shown in Figure 2.29(a) and (b), respectively.
2.6 Block-Diagram Representations of LTI Systems

Described by Differential Equations
The block diagram of a continuous system describes how the internal operations are
ordered, whereas the differential equation description gives only the input and output
relation. Hence, the block diagram is more detailed representation of continuous-
time systems than the differential equation description. Integrators are preferred to
differentiators in the block-diagram representation of continuous-time systems as the
integrators can be easily built from analog components and noise in a system will be
smoothed out.
Let us define the following three basic elements adder, scalar multiplier, and
integrator used in the block-diagram representation of continuous-time systems
(Figure 2.30).
Consider the system described by Eq. (2.54), which is repeated here for conve-
nience assuming M ¼ N:
XN dk yðtÞ X N d n xðtÞ
a
n¼0 n
¼ b n ð2:71Þ
dt k n¼0 dt n
0.25
0.2
0.15
Amplitude
0.1
0.05
0
0 0.5 1 1.5 2 2.5 3 3.5 4
Time
(a)
0.5
0.45
0.4
0.35
0.3
Amplitude
0.25
0.2
0.15
0.1
0.05
0
0 1 2 3 4 5 6 7 8
Time
(b)
Figure 2.29 (a) impulse response, (b) step response
n
If it is assumed that the system is at rest, then the Nth integral of d dtyðntÞ is yðNnÞ ðt Þ,
n
and the Nth integral of is d dtxðntÞ is xðNnÞ ðt Þ. Hence, taking the Nth integral of
Eq. (2.71), we obtain the integral description of the system as
2.7 Singularity Functions 95
Figure 2.30 Block-

diagram representation of
basic elements (a) adder,
(b) multiplier, (c) integrator
(a)
(b)
XN XN
n¼0
an yðNkÞ ðt Þ ¼ n¼0
bn xðNnÞ ðt Þ ð2:72Þ
Since y(0)(t) ¼ y(t), Eq. (2.72) can be rewritten as
1 hX N XN1 i
yð t Þ ¼ b n x ðNnÞ ð t Þ a n y ðNn Þ ð t Þ ð2:73Þ
aN n¼0 n¼0
The direct form I and the direct form II implementations of Eq. (2.73) are shown
in Figure 2.31(a) and (b), respectively.
Example 2.32 Obtain the block-diagram representation of a system described by
d2 y dy d 2 x dx
2
þ 2 þ y ¼ 2 6xðt Þ
dt dt dt dt
Solution
2.7 Singularity Functions
The unit impulse δ(t) is one of a class of signals known as singularity functions.
Consider a LTI system for which the input and the output are related by
dxðt Þ
yðt Þ ¼ ð2:74Þ
dt
The unit impulse response of the considered system is the derivative of the unit
impulse which is referred as the unit doublet u1(t).
(a)
(b)
Figure 2.31 Block-diagram representation for continuous-time system described by integral
Eq. (2.73) (a) direct form I, (b) direct form II
2.7 Singularity Functions 97
Figure 2.32 Block diagram

representation in direct form
II for a second order
continuous time system
described by the above
differential equation is
shown in Figure 2.32
By the convolution representation of LTI systems, we represent
dxðt Þ
¼ xðt Þ∗u1 ðt Þ ð2:75Þ
dt
for any input signal x(t). Similarly, for an LTI system described by
d 2 xð t Þ
yð t Þ ¼ , ð2:76Þ
dt 2
we obtain

d 2 xðt Þ d dxðt Þ
¼ ¼ xðt Þ∗u1 ðt Þ∗u1 ðt Þ
dt 2 dt dt ð2:77Þ
¼ xðt Þ∗u2 ðt Þ
where u2(t) ¼ u1(t) ∗ u1(t) is the second derivative of unit impulse.

Thus, uk(t) for k > 0 is the kth derivative of the unit impulse and is the impulse
response of a LTI system that takes the kth derivative of the input:
uk ðt Þ ¼ u1 ðt Þ∗u1 ðt Þ . . . . . . k times ð2:78Þ
If we consider a system described by Eq. (2.75) with input x(t) ¼ 1,then

ð1 ð1
dxðt Þ
¼ xðt Þ∗u1 ðt Þ ¼ u1 ðτÞxðt τÞdτ ¼ u1 ðτÞdτ ¼ 0 ð2:79Þ
dt 1 1
Hence, the unit doublet has zero area.

In addition to the singularity functions, the successive integrals of the unit
impulse function, it is known that
ðt
uð t Þ ¼ δðτÞdτ, ð2:80Þ
1
and hence the unit step function is the impulse response of an integrator, and we have
ðt
xðt Þ∗uðt Þ ¼ xðτÞdτ ð2:81Þ
1
Similarly, the impulse response of a system consisting of two integrators

in cascade can be denoted by u1(t) which can be expressed as the convolution of
u(t) with itself:
ðt
u1 ðt Þ ¼ uðt Þ∗uðt Þ ¼ uðτÞdτ ð2:82Þ
1
Since u(t) ¼ 0 for t < 0 and u(t) ¼ l, u1(t) can be expressed as
u1 ðt Þ ¼ tuðt Þ ð2:83Þ
Example 2.33 For a system if y(t) ¼ x(t) * h(t), show that

ð t
dhðt Þ
(i) yðtÞ ¼ xðτÞdτ ∗
1 dt
ðt
dxðt Þ
(ii) yðtÞ ¼ ∗ hðτÞdτ
dt 1
Solution
(i) yðtÞ ¼ xðt Þ∗hðt Þ,
¼ xðt Þ∗u1 ðt Þ∗u1 ðt Þ∗hðt Þ
Ðt dhðt Þ
¼ 1 xðτÞdτ∗
dt
(ii)
yðtÞ ¼ xðt Þ∗hðt Þ,
¼ xðt Þ∗u1 ðt Þ∗hðt Þ∗u1 ðt Þ
ðt
dxðtÞ
¼ ∗ hðτÞdτ
dt 1
2.8 State-Space Representation of Continuous-Time LTI

Systems
2.8.1 State and State Variables
The state of a system at time t0 is the minimal information required that is sufficient
to determine the state and the output of the system for all times t t0 for the known
system input at all times t t0.The variables that contain this information are called
state variables.
2.8 State-Space Representation of Continuous-Time LTI Systems 99
2.8.2 State-Space Representation of Single-Input Single-

Output Continuous-Time LTI Systems
Consider a single-input single-output continuous-time LTI system described by the

following Nth-order differential equation:
d N yð t Þ dN1 yðt Þ dyðt Þ

N þ a N1 N1
þ . . . þ a1 þ a 0 y ð t Þ ¼ ℧ð t Þ ð2:84Þ
dt dt dt
where y(t)) is the system output and ʊ(t) is the system input.
Define the following useful set of state variables
dyðt Þ d 2 yð t Þ dN1 yðt Þ

x1 ðt Þ ¼ yðt Þ, x2 ðt Þ ¼ , x3 ðt Þ ¼ , . . . , x N ð t Þ ¼ ð2:85Þ
dt dt 2 dt N1
Taking derivatives of the first N 1 state variables of the above, we get
dx1 ðt Þ dx2 ðt Þ dxN1 ðt Þ

¼ x2 ðt Þ, ¼ x3 ðt Þ, . . . , ¼ xN : ð2:86Þ
dt dt dt
Rearranging Eq. (2.84) and using Eq. (2.86), we obtain
dxN ðt Þ
¼ a0 x1 ðt Þ a1 x2 ðt Þ aN1 xN1 ðt Þ þ ℧ðt Þ ð2:87Þ
dt
yð t Þ ¼ x1 ð t Þ ð2:88Þ
Denoting dxðtÞ _,
dt by x
Eqs. (2.86), (2.87), and (2.88) can be written in matrix form as
2 3 2 32 3 2 3
x_ 1 ðtÞ 0 1 0 ... 0 x1 ðtÞ 0
6 7 6 76 7 6 7
6 2 x_ ðtÞ 7 6 0 0 1 ... 76 x2 ðtÞ 7 6 0 7
0
6 7 6 76 7 6 7
6 ⋮ 7¼6 ⋮ ⋮ ⋮ ⋱ 76 ⋮ 7 þ 6 0 7℧ðtÞ
⋮
6 7 6 76 7 6 7
6 7 6 76 7 6 7
4 x_ N1 ðtÞ 5 4 0 0 0 ... 1 54 xN1 ðtÞ 5 4 ⋮ 5
x_ N ðtÞ a0 a1 a2 . . . aN1 xN ðtÞ 1
ð2:89aÞ
2 3
x1 ð t Þ
6 7
6 x2 ð t Þ 7
6 7
6 7
yð t Þ ¼ ½ 1 0 0 ... 0 6 ⋮ 7 ð2:89bÞ
6 7
6 x ðt Þ 7
4 N1 5
xN ð t Þ
Define a Nx1 dimensional vector called state vector as

2 3
x1 ð t Þ
6 7
6 x2 ð t Þ 7
6 7
6 7
X ðt Þ ¼ 6 ⋮ 7 ð2:90Þ
6 7
6 x ðt Þ 7
4 N1 5
xN ð t Þ
The derivative of X(t) becomes

2 3
x_ 1 ðtÞ
6 7
6 x_ 2 ðtÞ 7
6 7
dXðtÞ 6 7
¼ X ðtÞ ¼ 6 ⋮ 7
_ 6
7 ð2:91Þ
dt 6 7
6 x_ N1 ðtÞ 7
4 5
x_ N ðtÞ
More compactly Eqs. (2.89a) and (2.89b) can be written as
X_ ðt Þ ¼ AXðt Þ þ b℧ðt Þ ð2:92Þ

yðt Þ ¼ cX ðt Þ ð2:93Þ
where
2 3 2 3
0 1 0 ... 0 0
6 7 6 7
6 0 0 1 ... 0 7 6 0 7
6 7 6 7
6 7 6 7
A¼6 ⋮ ⋮ ⋮ ⋱ ⋮ 7; b ¼ 6 0 7; c ¼ ½ 1 0 0 ... 0
6 7 6 7
6 0 ... 7 6⋮7
4 0 0 1 5 4 5
a0 a1 a2 . . . aN1 1
Eqs. (2.92) and (2.93) are called N-dimensional state-space representation or state
equations of the system.
In general state equations of a system are described by
X_ ðt Þ ¼ AXðt Þ þ b℧ðt Þ ð2:94Þ

yðt Þ ¼ cXðt Þ þ d℧ðt Þ ð2:95Þ
Example 2.34 Obtain the state-space representation of a system described by the

following differential equation:
d3 yðt Þ d 2 yð t Þ dyðt Þ
3
þ2 þ3 þ 4yðt Þ ¼ ℧ðt Þ
dt dt 2 dt
Solution: The order of the differential equation is three. Hence, the three-state
variables are
dyðt Þ d 2 yð t Þ
x1 ðt Þ ¼ yðt Þ, x2 ðt Þ ¼ , x3 ð t Þ ¼
dt dt 2
The first derivatives of the state variables are
x_ 1 ðt Þ ¼ x2 ðt Þ
x_ 2 ðt Þ ¼ x3 ðt Þ
x_ 3 ðt Þ ¼ 4x1 ðt Þ 3x2 ðt Þ 2x3 ðt Þ þ ℧ðt Þ
The state-space representation in matrix form is given by

2 3 2 32 2 3 3
x_ 1 ðt Þ 0 1 0 x1 ð t Þ 0
6 7 6 76 7 6 7
4 x_ 2 ðt Þ 5 ¼ 4 0 0 1 54 x2 ðt Þ 5 þ 4 0 5℧ðt Þ
x_ 3 ðt Þ -4 -3 -2 x3 ð t Þ 1
2 3
x1 ð t Þ
6 7
yð t Þ ¼ ½ 1 0 0 4 x2 ðt Þ 5
x3 ð t Þ
Example 2.35 Obtain the state-space representation for the electrical circuit shown
in Figure 2.33 considering V c1 , i1, and V c2 as state variables and vc1 as output y(t).
+ vc1 – i1
1H i2
1F i3
+
vs(t) + Vc2
1W 1F
– -
Figure 2.33 Third-order electrical circuit

Solution The state variables for the circuit are

x1 ¼ V c1
x2 ¼ i1
x3 ¼ V c2
From the relationship between the voltage V c1 and current i1, we obtain
dx1 ðt Þ
¼ x2
dt
Kirchhoff’s voltage equation around the closed loop gives
dx2 ðt Þ
V s þ x1 þ þ x3 ¼ 0
dt
This equation can be rewritten as
dx2 ðt Þ
¼ x1 x3 þ V s
dt
The current i3 ¼ x3 and the current i2 ¼ dxdt3 ðtÞ
By Kirchhoff’s current law
i1 ¼ i2 þ i3
implying that
dx3 ðt Þ
x2 ¼ þ x3
dt
Hence,
dx3 ðt Þ
¼ x2 x3
dt
The voltage V c1 is taken as the output y(t):
yð t Þ ¼ x1
The state-space representation of the circuit is given by

1Ω
i1(t ) iL(t )
1H
+
+
vs (t ) 1F vc (t )
−
1Ω
Figure 2.34 Electrical circuit of Example 2.36
2 3 2 32 2 3 3
x_ 1 ðt Þ 0 1 0 x1 ð t Þ
0
6 7 6 76 7 6 7
6 x_ 2 ðt Þ 7 ¼ 6 1 1 76 7 6 7
4 5 4 0 54 x2 ðt Þ 5 þ 4 0 5V s
x_ 3 ðt Þ 0 1 1 x3 ð t Þ 1
2 3
x1 ð t Þ
6 7
yð t Þ ¼ ½ 1 0 0 6 7
4 x2 ð t Þ 5
x3 ð t Þ
Example 2.36 Obtain state-space representation of the circuit shown in Figure 2.34
considering the current through the inductor and voltage across the capacitor as state
variables and voltage across the capacitor as the output y(t).
Solution The state variables for the circuit are
x1 ð t Þ ¼ i L ð t Þ
x2 ð t Þ ¼ V c ð t Þ
Since the voltage across the capacitor is equal to the voltage across the series
inductor and resistor branch,
we obtain
dx1 ðt Þ
þ x1 ðt Þ ¼ x2 ðt Þ
dt
dx1 ðt Þ
¼ x1 ðt Þ þ x2 ðt Þ
dt
Kirchhoff’s voltage equation around the closed loop gives
V s ðt Þ þ i1 ðt Þ þ x2 ðt Þ ¼ 0
Hence
i1 ðt Þ ¼ x2 ðt Þ þ V s ðt Þ
By Kirchhoff’s current law
dx2 ðt Þ
i1 ðt Þ ¼ x1 ðt Þ þ
dt
Therefore
dx2 ðt Þ
x2 ðt Þ þ V s ðt Þ ¼ x1 ðt Þ þ
dt
dx2 ðt Þ
¼ x1 ðt Þ x2 ðt Þ þ V s ðt Þ
dt
The voltage V c ðt Þ is taken as the output y(t):
yð t Þ ¼ x2 ð t Þ
The state-space representation of the circuit is given by

" # " #" # " #
x_ 1 ðt Þ 1 1 x1 ðt Þ 0
¼ þ vs ð t Þ
x_ 2 ðt Þ 1 1 x2 ðt Þ 1
" #
x1 ð t Þ
yð t Þ ¼ ½ 0 1
x2 ð t Þ
2.8.3 State-Space Representation of Multi-input Multi-output

Continuous-Time LTI Systems
The state-space representation of continuous-time system with m inputs and l output

and N state variables can be expressed as
2.9 Problems 105
X_ ðt Þ ¼ AX ðt Þ þ B℧ðt Þ ð2:96Þ
yðt Þ ¼ CX ðt Þ þ D℧ðt Þ ð2:97Þ
where
2 3 2 3
a11 a12 ... a1N b11 b12 ... b1m
6 7 6 7
6 a21 a22 ... a2N 7 6 b21 b22 ... b2m 7
A¼6
6⋮
7 B¼6 7
4 ⋮ ⋱ ⋮ 7 5
6⋮
4 ⋮ ⋱ ⋮ 7 5
aN1 aN2 ... aNN NN bN1 bN2 . . . bNm Nm
2 3 2 3
c11 c12 . . . c1N d11 d12 . . . d 1m
6 7 6 7
6 c21 c22 . . . c2N 7 6 d21 d22 . . . d 2m 7
C¼6
6⋮ ⋮
7 6
D¼6 7
4 ⋱ ⋮7 5 4⋮ ⋮ ⋱ ⋮5
7
cl1 cl2 ... clN lN dl1 dl2 ... d lm lm
2.9 Problems
1. Check the following for linearity and time-invariance:
(i) yðt Þ ¼ dxdtðtÞ

(ii) y(t) ¼ tx(t)
(iii) y(t) ¼ t2x2(t)
dyðt Þ
(iv) dt þ tyðt Þ ¼ xðt Þ
(v) y(t) ¼ ln (x(t))
(vi) y(t) ¼ x(t) þ cons tan t
2. Determine which of the following systems are linear and which are nonlinear:
dyðt Þ
(i) dt þ 3yðt Þ ¼ x2 ðt Þ
dyðt Þ
(ii) þ y2 ð t Þ ¼ xð t Þ
dt
2
dyðt Þ
(iii) dt þ 3yðt Þ ¼ xðt Þ
dyðt Þ
(iv) dt þ sin ðt Þyðt Þ ¼ dxdtðtÞ þ 3xðt Þ
3. An amplifier has an output y(t) ¼ cos (ωt). If y(t) is limited by clipping resulting
in the clipped output yc(t), check for linearity and time-invariance of yc(t).
4. An input signal x(t) and two possible outputs of a linear time-invariant system are
shown in Figure P2.1. Which outputs are possible given that the system is linear
and time invariant?
x (t ) y (t )
(i)
(ii)
Figure P2.1 Input signal and two possible outputs of problem 4
5. If x(t) ¼ u(t + 1) u(t 1), compute (x *x *x)(t) and sketch.

6. Determine graphically the convolution y(t) ¼ x(t) * h(t) for the following:
xðt Þ ¼ uðt Þ-uðt-4Þ
(i)
hðt Þ ¼ uðt-4Þ-uðt-6Þ
xðt Þ ¼ et uðt Þ
(ii)
hðt Þ ¼ e2t uðt Þ
xðtÞ ¼ 2uðt 1Þ 2uðt 2Þ
(iii)
hðtÞ ¼ uðt þ 1Þ 2uðt 1Þ þ uðt 2Þ
xðt Þ ¼ uðt Þ
(
(iv) et , t 0,
hð t Þ ¼
et , t 0
xðt Þ ¼ uðt þ 1Þ uðt 1Þ
(v) 1
hðt Þ ¼ t ðuðt Þ uðt 3ÞÞ
3
7. Consider a continuous-time LTI system with the step response
sðtÞ ¼ et uðtÞ
Determine and sketch the output of this system to the input
xðtÞ ¼ uðt 1Þ uðt 3Þ
8. Consider h(t) be the triangular pulse and x(t) be the unit impulse train as shown in
Figure 2.16 of Example 2.17. Determine y(t) ¼ x(t) * h(t) and sketch it.
(i) for T ¼ 3 (ii) for T ¼ 32
9. An LTI system consists of two subsystems in cascade. The impulse responses of
the subsystems are, respectively, given by
h1 ðt Þ ¼ δðt Þ 2e2t uðt Þ; h2 ð t Þ ¼ e t uð t Þ
Find the impulse response of the overall system.

2.9 Problems 107
10. If y(t) ¼ x(t) * h(t) shows that the area of the convolution y(t) is the product of the
areas of the signals that are being convolved x(t) and h(t), that is,
ð1 ð 1 ð 1
yðτÞdτ ¼ xðτÞdτ hðτÞdτ
1 1 1
11. Compute and sketch the periodic convolution of the square-wave signal x(t)
shown in Fig. P2.2 with itself.
Figure P2.2 Square wave x(t)

signal of problem 11
t
-2 -1 0 1 2
12. Consider an LTI system with impulse response h(t):

(i) Check for its stability, if h(t) is periodic and nonzero.
(ii) Check for causality of the inverse of the LTI system if h(t) is causal.
(iii) Check for its stability if h(t) is causal.
13. Consider the system described by
dyðt Þ dxðt Þ
þ 3yðt Þ ¼ xðt Þ þ
dt dt
Determine the impulse response h(t) of the system.
14. Consider the system described by
dyðt Þ
þ yðt Þ ¼ xðt Þ yð0Þ ¼ 0
dt
(i) Determine the step response of the system.

(ii) Determine the impulse response from the step response.
15. Determine the response y(t) of the OP-Amp circuit shown in Figure P2.3 for an
input x(t) ¼ u(t)
Figure P2.3 OP-Amp circuit of problem 15
16. Determine the impulse response y(t) of the OP-Amp circuit shown in
Figure P2.4.
Figure P2.4 OP-Amp circuit of problem 16
17. Determine the impulse response h(t) for a system described by
d2 y dy dx
þ 3 þ 2y ¼
dt 2 dt dt
18. Draw block diagrams for direct form II implementation of the corresponding
systems
d2 y
(i) dt 2
þ 5dy dx
dt þ 4y ¼ dt þ x
2.10 MATLAB Exercises 109
(ii) dy dx
dt þ 3y ¼ 2 dt þ xðt Þ
2 2
(iii) d y/dt ady/dt ¼ a dx/dt + abx(t)
19. For a given signal x(t)

(i) Show that xðt Þu1 ðt Þ ¼ xð0Þu1 ðt Þ dxdtðtÞ δðt Þ:
t¼0
(ii) Determine the value of
ð1
xðτÞu2 ðτÞdτ:
1
(iii) Find an expression for x(t) u2(t) similar to (i).

20. Obtain the state-space representation for the electrical circuit shown
In Figure P2.5 considering i1, V 1, and V 2 as state variables and current through the
inductor as the output y(t)
Figure P2.5 Electrical − v1 +

circuit of prblem 20
i3 i2
1Ω 1F
+ 1F +
i1 v2
1H
−
1Ω
1. Verify the solution of problem 2 using MATLAB.

2. Write a MATLAB program to compute the convolution of the input x(t) and the
impulse response h(t) shown in Figure M2.1.
1 1
0 5 0 0.5 1.5
Figure M2.1 Input and Impulse response of MaTLAB exercise 2

3. If x(t) ¼ u(t 1) u(t 2), write a MATLAB program to compute the result y10(t)
convolving ten x(t) functions together, that is
y10 ðt Þ ¼ xðt Þ∗xðt Þ∗xðt Þ∗xðt Þ∗xðt Þ∗xðt Þ∗xðt Þ∗xðt Þ∗xðt Þ∗xðt Þ
and comment on the result.

4. Write a MATLAB program to find the impulse response h(t) and step response for
a system described by
d 2 yð t Þ dyðt Þ dxðt Þ
þ5 þ 6yðt Þ ¼ þ xðt Þ:
dt 2 dt dt
Further Reading
1. Oppenheim, A.V., Willsky, A.S.: Signals and Systems. Prentice-Hall, Englewood Cliffs (1983)
2. Hsu, H.: Signals and Systems Schaum’s Outlines, 2nd edn. McGraw-Hill, New York (2011)
3. Kailath, T.: Linear Systems. Prentice-Hall, Englewood Cliffs (1980)
4. Zadeh, L., Desoer, C.: Linear System Theory. McGraw-Hill, New York (1963)
Chapter 3
Frequency Domain Analysis of Continuous-
Time Signals and Systems
The continuous-time signals and systems are often characterized conveniently in a

transform domain. This chapter describes the transformations known as Fourier
series and Fourier transform which convert time-domain signals into frequency-
domain (or spectral) representations. The frequency domain representation of
continuous-time signals are described along with the conditions for the existence
of Fourier series for periodic signals and Fourier transform for nonperiodic signals
and their properties. Finally, the frequency response of continuous-time systems is
discussed.
3.1 Complex Exponential Fourier Series Representation

of the Continuous-Time Periodic Signals
It is recalled from Chapter 1 that a signal x(t) is periodic if
xðt Þ ¼ xðt þ T Þ for all t ð3:1Þ
The fundamental period T0 is small minimum, positive nonzero value of T for

which Eq. (3.1) is satisfied, and Ω0 ¼ T2π0 is referred to as the fundamental angular
frequency.
A sinusoidal signal x(t) ¼ cos (ω0t) and the complex exponential signal xðt Þ
¼ ejΩ0 t are the two examples of periodic signals.
The complex exponentials related harmonically are expressed by
xn ðt Þ ¼ ejnΩ0 t , n ¼ 0, 1, 2, ð3:2Þ
The fundamental frequency of each of these signals is a multiple of Ω0, and hence
each is periodic with period T0.

112 3 Frequency Domain Analysis of Continuous-Time Signals and Systems
Thus, a linear combination of complex exponentials related harmonically can be

written as
X1
xðt Þ ¼ n¼1
an ejnΩ0 t ð3:3Þ
Eq. (3.3) is the Fourier series representation of a periodic signal x(t).

The Fourier coefficients ak can be determined as follows:
Multiplying Eq. (3.3) both sides by ejmΩ0 t , we obtain
X1
xðt ÞejmΩ0 t ¼ n¼1
an ejnΩ0 t ejmΩ0 t ð3:4Þ
Integrating Eq. (3.4) both sides from 0 to T0, we have

ðT0 ðT0 X
jmΩ0 t 1
xðt Þe ¼ n¼1
an ejnΩ0 t ejmΩ0 t dt ð3:5Þ
0 0
Interchanging the integration and summation, Eq. (3.5) can be rewritten as

ðT0 X1 ð T 0
xðt ÞejmΩ0 t dt ¼ a
n¼1 k
ejðnmÞΩ0 t dt ð3:6Þ
0 0
ðT0 ðT0 ðT0
ejðnmÞΩ0 t dt ¼ cos ððn mÞΩ0 t Þ dt þ j sin ððn mÞΩ0 t Þ dt ð3:7Þ
0 0 0
Hence,
ðT0
T0 m¼n
ejðnmÞΩ0 t dt ¼ ð3:8Þ
0 0 m 6¼ n
Thus, Eq. (3.6) becomes

ðT0
xðt ÞejmΩ0 t dt ¼ an T 0 ð3:9Þ
0
Therefore, the Fourier coefficients an are given by

ðT0
1
an ¼ xðt ÞejnΩ0 t dt ð3:10Þ
T0 0
Thus, the Fourier series of a periodic signal is defined by Eq. (3.3) referred to as
the synthesis equation and Eq. (3.10) as the analysis equation.
3.1 Complex Exponential Fourier Series Representation of the Continuous. . . 113
3.1.1 Convergence of Fourier Series
The sufficient conditions for guaranteed convergence of Fourier series are the
following Dirichlet conditions:
1. x(t) must be absolutely integral over any period, that is,
ð
j xðt Þ j dt < 1: ð3:11Þ
T0
which guarantees that each Fourier coefficient ak has finite value.

2. x(t) must have finite number of maxima and minima during any single period of it.
3. x(t) must have finite number of discontinuities in any finite interval of time and
each of these discontinuities being finite.
3.1.2 Properties of Fourier Series
Linearity Property If x1(t) and x2(t) are two continuous-time signals with Fourier
series coefficients an and bn, then the Fourier series coefficients of a linear combi-
nation of x1(t) and x2(t), that is, c1x1(t) þ c2x2(t), are given by
c 1 an þ c 2 bn
where c1 and c2 are arbitrary constants.

Time Shifting Property If x(t) is a continuous-time periodic signal with period T0
and the Fourier coefficients an, then the Fourier series coefficients of the x(t t0) are
jnT2π t 0
given by e 0 an .
Proof Since Ω0 ¼ T2π0 , the Fourier coefficients an of a periodic signal x(t) are given
by
ðT0
1 jnT2π t
an ¼ xðt Þe 0 dt
T0 0
Let the Fourier series coefficients of x(t t0) be b

an
ðT0
1 jnT2π t
b
an ¼ xðt t 0 Þe 0 dt
T0 0
Letting τ ¼ t t0
ðT0
jn T τ jn T t 0
1 2π 2π
b
an ¼ xðτÞe 0 e 0 dτ
T0 0
ðT0
jn T τ
2π 1 2π
jn T t 0
b
an ¼ e 0 xðτÞe 0 dτ
T0 0
2π
jn T t 0
b
an ¼ e 0 an
Conjugate Property If x(t) is a continuous-time periodic signal with the Fourier

coefficients an, then the Fourier series coefficients of the x*(t) are given by a∗
n .
ðT0
1 jn 2π t
Proof an ¼ xðt Þe T 0 dt, then replacing n by n, we get
T0 0
ðT0
1 jnT2π t
an ¼ xðt Þe 0 dt
T0 0
Taking both sides conjugate of this equation, we have

ðT0
1 jnT2π t
a∗
n ¼ x∗ ðtÞe 0 dt
T0 0
Symmetry for Real Valued Signal If x(t) is a continuous-time real valued signal
with the Fourier coefficients an, then
an ¼ a∗
n
where * stands for the complex conjugate.

Proof From conjugate property, we know that
ðT0
1 jnT2π t
a∗
n ¼ x∗ ðtÞe 0 dt
T0 0
Since xðt Þ is real x∗ ðt Þ ¼ xðt Þ, we get

ð
1 T0 jn
2π
t
a∗n ¼ xðt Þe T 0 dt
T0 0
¼ an
implies that Re(an) ¼ Re(a–n), i.e., the real part of an is even, andIm(an) ¼ (Im
(an)), i.e., the imaginary part of an is odd.
If x(t) is a continuous-time real and even signal, i.e., x*(t) ¼ x(t) x(t) ¼ x(t), it
can be easily shown that an¼ an and an ¼ a∗ n.
Similarly, if x(t) is a continuous-time real and odd signal, i.e., x*(t) ¼ x(t) x(t) ¼
x(t), it can be easily shown that an ¼ an and an ¼ a∗ n .
Frequency Shifting Property If x(t) is a continuous-time periodic signal with

period T0 and the Fourier coefficients an, then the Fourier series coefficients of the
jK T2π t
e 0 xðt Þ are given by anK.
Proof The Fourier coefficients an of a periodic signal x(t) are given by
ðT0
1 jnT2π t
an ¼ xðt Þe 0 dt
T0 0
jK T2π t
Let dn be the Fourier coefficients of e 0 xðt Þ, then
ðT0 2π 2π
1 jK T t jn T t
dn ¼ xðt Þe 0 e 0 dt
T0 0
ð
1 T0 2π
jðnK Þ T t
¼ xðt Þe 0 dt
T0 0
¼ anK
Thus, it is proved.
Time Reversal Property If x(t) is a continuous-time periodic signal with the
Fourier coefficients an, then the Fourier series coefficients of the x(t) are given
by an.
Time Scaling Property If x(t) is a continuous-time periodic signal with period T0
and the Fourier coefficients an, then the Fourier series coefficients of the x(αt) α >
0 are given by an with period Tα0 .
Proof Since Ω0 ¼ T2π0 , the Fourier coefficients an of a periodic signal x(t) are
given by
ðT0
1 jnT2π t
an ¼ xðt Þe 0 dt
T0 0
Let the Fourier series coefficients of x(αt) be b

a n , then
ðT0
α jn2παt
b
an ¼ xðαt Þe T 0 dt
T0 0
Letting τ ¼ αt
ðT0
jn T τ
1 2π
b
an ¼ xðτÞe 0 dτ
T0 0
¼ an
b 0 ¼ T0.
Hence, T
α
Differentiation in Time If x(t) is a continuous-time periodic signal with period T0

and the Fourier coefficients an, then the Fourier series coefficients of the dtd xðt Þ are
given by jn T2π0 an
Proof The Fourier series representation of of x(t) is

X1
xð t Þ ¼ n¼1
an ejnΩ0 t
Differentiating this equation both sides with respect to t, we obtain
d X1
xð t Þ ¼ jnΩ0 an ejnΩ0 t
dt n¼1
Since Ω0 ¼ T2π0
d X1
2π
xð t Þ ¼ jn an ejnΩ0 t
dt n¼1
T0
giving the Fourier series representation of dtd xðt Þ. Comparing this with the Fourier
series representation of coefficients an, it is clear that the Fourier coefficients of
dt xðt Þ are jn T 0 an
d 2π
Integration Property If x(t) is a continuous-time periodic signal withÐ period T0 and

the Fourier coefficients an, then the Fourier series coefficients of the x(t) are given
T0
by jn2π an
Proof The Fourier series representation of x(t) is

X1
xðt Þ ¼ n¼1
an ejnΩ0 t
Integrating this equation both sides with respect to t, we obtain

ð ð !
X
1
xðt Þdt ¼ an e jnΩ0 t
dt
n¼1
X1
an jnΩ0 t
¼ e
n¼1
jnΩ0
Since Ω0 ¼ T2π0
ð X1
T0
xðt Þdt ¼ an ejnΩ0 t
n¼1
jn2π
Ð
giving the Fourier series representation of x(t)dt. Comparing this with the Fourier Ð
series representation of coefficients an, it is clear that the Fourier coefficients of x(t)
T0
dt are jn2π an
Periodic Convolution If x1(t) and x2(t) are two continuous-time signals with
common period T0 and Fourier coefficients an and bn, respectively, then the Fourier
series coefficients of the convolution integral of x1(t) and x2(t) are given by T0anbn.
Proof The periodic convolution integral of two signals with common period T0 is
defined by
ð
yðt Þ ¼ x1 ðτÞx2 ðt τÞdτ ð3:12Þ
T0
Let cn be the Fourier series coefficients of y(t), then

ðT0 ð T 0 ð
1 1
cn ¼ yðt ÞejnΩ0 t dt ¼ x1 ðτÞx2 ðt τÞdτ ejnΩ0 t dt ð3:13Þ
T0 0 T0 0 T0
Letting t τ ¼ t1, Eq. (3.13) becomes

ð T 0 ð
1
cn ¼ x1 ðτÞx2 ðt 1 Þdτ ejnΩ0 ðτþt1 Þ dt 1
T0 0 T0
Interchanging the order of integration, the above equation can be rewritten as

ð ð
1 jnΩ0 τ jnΩ0 t 1
cn ¼ x1 ðτÞe dτ x2 ðt 1 Þe dt 1 ð3:14Þ
T0 T0 T0
By definition of Fourier series

ð
1
x1 ðτÞejnΩ0 τ dτ ¼ an
T0 T0
hÐ i
jnΩ0 t 1
T0 x 2 ðt 1 Þe dt 1 ¼ T 0 bn
Hence cn¼ T0anbn.. Thus, it is proved.

Multiplication Property
If x1(t) and x2(t) are two continuous-time signals with Fourier coefficients an and bn,
respectively,
X1 then the Fourier series coefficients of a new signal x1(t)x2(t) are given
by a b implying that signal multiplication in the time domain is equiv-
l¼1 l nl
alent to discrete-time convolution in the frequency domain.
Proof Let dn be the Fourier coefficients of the new signal x1(t)x2 (t). By definition of
Fourier series representation of a signal, we have
ðT0
1
dn ¼ x1 ðτÞx2 ðt ÞejnΩ0 t dt
T0 0
ðT0 X
1 1
¼ al ejlΩ0 t x2 ðt ÞejnΩ0 t dt
T0 0
l¼1
ðT0
P1 1
¼ l¼1 al x2 ðt ÞejðnlÞΩ0 t dt
T0 0
P1
¼ l¼1 al bnl
Parseval’s Theorem If x(t) is a continuous-time signal with period T0 and Fourier

coefficients an, then the average power P of x(t) is given by
ð X1
1
P¼ jxðt Þj2 dt ¼ n¼1 n
ja j2 ð3:15Þ
T 0 T0
Proof The average power P of a periodic signal x(t) is defined as

ð
1
P¼ jxðt Þj2 dt ð3:16Þ
T0 T0
Assuming that x(t) is complex valued x(t)x*(t) ¼ |x(t)|2 and x*(t) can be expressed
in terms of its Fourier series as
X1
x∗ ð t Þ ¼ n¼1
a∗
n e
jnΩ0 t
ð3:17Þ
Eq. (3.16) can be rewritten as

ð
1
P¼ xðt Þx∗ ðt Þ dt ð3:18Þ
T0 T0
Substituting Eq. (3.17) in Eq. (3.18), we obtain

ð hX1 i
1 ∗ jnΩ0 t
P¼ xð t Þ a
n¼1 n
e dt ð3:19Þ
T0 T0
Interchanging the order of integration, Eq. (3.19) can be rewritten as
X1 ðT0
1 jnT2π t
P¼ a∗
n¼1 n T
xðt Þe 0 dt ð3:20Þ
0 0
By definition of the Fourier series

ðT0
1 jnT2π t
an ¼ xðt Þe 0 dt
T0 0
Thus,
Table 3.1 Some properties of continuous-time Fourier series

Property Periodic signal Fourier series coefficients
Linearity c1x1(t) þ c2x2(t) c1an + c2bn
Time shifting x(t t0) jn 2π t
e T 0 0 an
Time reversal x(t) an
Conjugate x*(t) a∗
8n
Symmetry x(t) real >
> an ¼ a∗ n
>
>
xe(t) (x(t) real) < Re½an ¼ Re½an
xo(t) (x(t) real) Im½an ¼ Im½an
>
>
>
> j a j¼j an j
: n
arg½an ¼ arg½an
Re[an]
jIm[an]
Frequency jK T2π t anK
e 0 xðt Þ
shifting
Time scaling x(αt), α > 0 an
(periodic with period Tα0 )
Differentiation in time d
xðt Þ jn T2π0 an
Ðdt T0
Integration x(t)dt jn2π an
Ð
T 0 x1 ðτ Þx2 ðt τ Þdτ
Periodic convolution property T0anbn.
Multiplication property x1(t)x2(t) X1
¼ al bnl
l¼1
ð X
1
1
P¼ jxðt Þj2 dt ¼ jan j2
T0 T0 n¼1
The properties of continuous-time Fourier series are summarized in Table 3.1.

ð1 X1
1
Parseval’ s Theorem jxðt Þj2 dt ¼ jan j2
T0 1
n¼1
Half-Wave Symmetry
If the two halves of one period of a periodic signal are of identical shape, except that
one is the negation of the other, the periodic signal is said to have a half-wave
symmetry. Formally, if x(t) is a periodic signal with period T0, then x(t) has half-

wave symmetry if x t T20 ¼ xðt Þ
Example 3.1 Prove that Fourier series representation of a periodic signal with half-
wave symmetry has no even-numbered harmonics.
Proof A periodic signal x(t) with half-wave symmetry is given by
(
xð t Þ 0 t < T 0 =2
xð t Þ ¼
xðt Þ T 0 =2 t < T 0
ð
1
an ¼ xðt ÞejnΩ0 t dt
T 0 T0
ð ð
1 T 0 =2 1 T0
¼ xðt ÞejnΩ0 t dt þ xðt ÞejnΩ0 t dt
T0 0 T 0 T 0 =2

T0
For T 0 =2 t < T 0 xðt Þ ¼ x t
2

Substituting xðt Þ ¼ x t T20 in the above equation, we get
ð T 0 =2 ðT0
1 1 T 0 jnΩ0 t
an ¼ xðt ÞejnΩ0 t dt x t e dt
T0 0 T0 T 0 =2 2
By using time shifting property, we obtain

ð T 0 =2 ð T 0 =2
1 T0 1
an ¼ xðt ÞejnΩ0 t dt ejnΩ0 2 xðt ÞejnΩ0 t dt
T0 0 T0 0
Since Ω0 ¼ T2π0
ð T 0 =2 ð T 0 =2
1 1
an ¼ xðt ÞejnΩ0 t dt ejnπ xðt ÞejnΩ0 t dt
T0 0 T0 0
ð T 0 =2
ð1 ejnπ Þ
¼ xðt ÞejnΩ0 t dt
T0 0
ð
ð1 ð1Þn Þ T 0 =2
T0 0
8
< 0 for even n
¼ 2 Ð T 0 =2
: xðt ÞejnΩ0 t dt for odd n
T0 0
Example 3.2 Find Fourier series of the following periodic signal with half-wave
symmetry as shown in Figure 3.1.
Figure 3.1 Periodic signal

with half-wave symmetry
π
Solution The period T0 ¼ 6 and Ω0 ¼ 2π 6 ¼ 3
Fourier coefficients are given by
8
>
< 0 for even n
an ¼ ð T 0 =2
> 2
T0 0
For odd n
ð
2 T 0 =2
T0 0
ð
2 3
6 0
ð
1 2 jnπt
¼ e 3 dt
3 1
1 j2nπ=3
¼ e ejnπ=3
jnπ
Example 3.3 Find Fourier series of the following periodic signal with half-wave
symmetry (Figure 3.2)
π
Solution The period T0 ¼ 8 and Ω0 ¼ 2π
8 ¼ 4
Fourier coefficients are given by
8
< 0 for even n
an ¼ 2 Ð T 0 =2
T0 0
For odd n
x(t)
2 4 6 8 t
-1
Figure 3.2 Periodic signal with half-wave symmetry

ð T 0 =2
2
T0 0
ð4
2
8 0
ð2
1 t jnπ t
¼ e 4 dt
4 02
ð
1 2 jnπ t
¼ te 4 dt
8 0
" ð2 #
1 t jnπ t 2 1 jn
π
t
¼ e 4 þ e 4 dt
8 jnπ=4 0 jnπ=4 0
" 2 #
1 8j k 16 jnπt
¼ j þ 2 2e 4
8 nπ nπ 0
jðkþ1Þ 2
¼ þ 2 2 jðkÞ 1
nπ n π
Example 3.4 Consider the periodic signal x(t) given by
xðt Þ ¼ ð2 þ j2Þej3t j3ej2t þ 6 þ j3ej2t þ ð2 j2Þej3t
(i) Determine the fundamental period and frequency of x(t)

(ii) Show that x(t) is a real signal
(iii) Find energy of the signal
Solution (i) x(t) is the sum of two periodic signals with periods T 1 ¼ 2π
3 and T 2
¼ 2π2 The ratio T1
T2 ¼ 2
3 is a rational number.
The fundamental period of the signal x(t) is 3T1 ¼ 2T2 ¼ 2π. The fundamental
frequency Ω0 ¼ 2π
2π ¼ 1.
(ii) x(t) is exponential Fourier series representation of the form
X
1
xð t Þ ¼ an ejnΩ0 t
n¼1
where
a3 ¼ 2 þ j2; a2 ¼ j3; a0 ¼ 6; a3 ¼ 2 j2; a2 ¼ j3
and an¼ 0 for all other. It is noticed that an ¼ a∗

n for all n. Hence, x(t) is a real signal.
(iii) The average power of the signal x(t) is

ð X
1
1
E¼ jxðt Þj2 dt ¼ j an j 2
T0 T0 n¼1
By Parseval’s theorem,
X
1
E¼ j an j 2
n¼1
pffiffiffiffiffiffiffiffiffiffiffiffiffiffiffi
2 pffiffiffiffiffiffiffiffiffiffiffiffiffiffiffi
2
¼ 22 þ 22 þ 32 þ 62 þ 32 þ 22 þ 22
¼ 8 þ 9 þ 36 þ 9 þ 8 ¼ 70
Example 3.5 Find the Fourier series coefficients for each of the following signals:
(i) x(t) ¼ cos(Ω0t) þ sin (2Ω0t)
(ii) x(t) ¼ 2 cos(Ω0t) þ sin2 (2Ω0t)
X1
Solution (i) xðt Þ ¼ n¼1 n
a ejnΩ0 t
1 1 1 1
xðt Þ ¼ ejΩ0 t þ ejΩ0 t þ ej2Ω0 t ej2Ω0 t
2 2 2j 2j
1 1 1 1
a1 ¼ ; a1 ¼ ; a2 ¼ ; a2 ¼
2 2 2j 2j
an ¼ 0 for all other n.
(ii) xðt Þ ¼ 2 cos ðΩ0 t Þ þ sin 2 ð2Ω0 t Þ ¼ 2 cos ðΩ0 t Þ þ 12 ½1 cos ð4Ω0 t Þ
X
1
n¼1

1 jΩ0 t 1 jΩ0 t 1 1 1 j4Ω0 t 1 j4Ω0 t
xð t Þ ¼ 2 e þ e þ e þ e
2 2 2 2 2 2

1 1 1 j4Ω0 t 1 j4Ω0 t
¼ ejΩ0 t þ ejΩ0 t þ e þ e
2 2 2 2
1 1 1
a1 ¼ 1; a0 ¼ ; a1 ¼ 1; a4 ¼ ; a4 ¼
2 4 4
an ¼ 0 for all other n.
Example 3.6 Find Fourier series coefficients of the following continuous-time

periodic signal and plot the magnitude and phase spectrum of it:
xðt Þ ¼ 2s inð2πt 3Þ þ s inð6πt Þ
Solution
2 jð2πt3Þ 2 jð2πt3Þ 1 jð6πtÞ 1 jð6πtÞ
xðt Þ ¼ e e þ e e
2j 2j 2j 2j
1 1 1 1
¼ ej3 ejð2πtÞ ej3 ejð2πtÞ þ ejð6πtÞ ejð6πtÞ
j j 2j 2j
1 1 1 1
¼ ejð6πtÞ ej3 ejð2πtÞ þ ej3 ejð2πtÞ þ ejð6πtÞ
2j j j 2j
Since Ω0¼ 2π
1 1 1 1
xðt Þ ¼ ejð3Ω0 tÞ ej3 ejðΩ0 tÞ þ ej3 ejðΩ0 tÞ þ ejð3Ω0 tÞ
2j j j 2j
X
1
n¼1
1 j 1 π
a3 ¼ ¼ ¼ ejð2Þ
2j 2 2
ej3
a1 ¼ ¼ jej3 ¼ ej1:7124
j
ej3
a1 ¼ ¼ jej3 ¼ ej1:7124
j
1 j 1
a3 ¼ ¼ ¼ e j ð 2 Þ
π
2j 2 2
an¼ 0 for all other n.
The magnitudes of Fourier coefficients are
ja3 j ¼ ja3 j ¼ 0:5

ja1 j ¼ ja1 j ¼ 1:0
The magnitude spectrum and phase spectrum are shown in Figures 3.3 and 3.4,
respectively.
Since x(t) is a real valued, its magnitude spectrum is even and the phase spectrum
is odd.
Example 3.7
(i) Obtain x(t) for the following non-zero Fourier series coefficients of a continuous-
time real valued periodic signal x(t) with fundamental period of 8.
a1 ¼ a∗
1 ¼ j, a5 ¼ a5 ¼ 1
Figure 3.3 Magnitude spectrum of x(t)
Figure 3.4 Phase spectrum of x(t)
Figure 3.5 Magnitude

spectrum of a signal
(ii) Consider a continuous periodic signal with the following magnitude spectra
shown in Figure 3.5. Find the DC component and average power of the signal.
Solution (i)
X
1
n¼1
xðt Þ ¼ ej5Ω0 t þ ej5Ω0 t þ jðejΩ0 t ejΩ0 t Þ

¼ 2 cos ð5Ω0 t Þ 2 s inðΩ0 t Þ
π
¼ 2 cos ð5Ω0 t Þ þ 2 cos Ω0 t þ

2
(ii) The DC component is given by
a0 ¼ 1:
By
X using Parseval’s relation, the average power is computed as
1
n¼1 n
ja j2 ¼ 12 þ 22 þ 12 þ 22 þ 12 ¼ 11
Example 3.8 Which of the following signals cannot be represented by the Fourier
series?
(i) x(t) ¼ 4 cos(t) þ 6 cos(t)
(ii) x(t) ¼ 3 cos(πt) þ 6 cos(t)
(iii) x(t) ¼ cos(t) þ 0.75
(iv) x(t) ¼ 2 cos(3πt) þ 3 cos(7πt)
(v) x(t) ¼ e|t| sin (5πt)
Solution (i) x(t) ¼ 4 cos(t) þ 6 cos(t) is periodic with period 2π.
(ii) x(t) ¼ 3 cos(πt) þ 6 cos(t)
The first term has period
2π
T1 ¼ ¼2
π
The second term has period
2π
T2 ¼ ¼ 2π
1
The ratio TT 12 ¼ π2 is not a rational number. Hence, x(t) is not a periodic signal.
(iii) x(t) ¼ cos(t) þ 0.75 is periodic with period 2π.
(iv) x(t) ¼ 2 cos(3πt) þ 3 cos(7πt)
The first term has period
2π 2
T1 ¼ ¼
3π 3
The second term has period
2π 2
T2 ¼ ¼
7π 7
The ratio TT 12 ¼ 73 is a rational number. Hence, x(t) is a periodic signal.
(v) Due to decaying exponential function, it is not periodic. So Fourier series cannot
be defined for it.
Hence, (ii) and (v) cannot be represented by Fourier series. Since the remaining
three are periodic; they can be represented by Fourier series.
Example 3.9 Find the Fourier series of a periodic square wave with period T0
defined over one period by

1 j t j< T 0 =4
xðt Þ ¼
0 T 0 =4 <j t j T 0 =2
Solution For n¼0,

ð T 0 =4
1 2T 0 =4 1
a0 ¼ dt ¼ ¼
T0 T 0 =4 T0 2
For n 6¼ o,
ð T 0 =4 T 0 =4

1
an ¼ T10 ejnΩ0 t dt ¼ ejnΩ0 t
T 0 =4 jnΩ0 T 0
T 0 =4
jnΩ0 T 0 =4 jnΩ0 T 0 =4

2 e e
¼
nΩ0 T 0 2j
2 sin ðnΩ0 T 0 =4Þ
¼
nΩ0 T 0
Since Ω0 ¼ T2π0 ,

2π
sin T 0 =4
T0
an ¼ n 6¼ 0
0 nπ 1
2π
BT 0 T 0 C
sin B
@ 4 A
C
¼

nπ
nπ
1 sin
¼ 2
2 nπ=2
Example 3.10 Find the Fourier series of the periodic signal shown in Figure 3.6.
Solution The period T0 ¼ 2. Ω0 ¼ T2π0 ¼ π:
For n¼0,
ð1
1
a0 ¼ tdt ¼ 0
2 1
For n 6¼ o,
ð1
1
an ¼ tejnπt dt
2 1
" 1 ð #
1 t
jnπt 1 1 jnπt
¼ e þ jnπ e dt
2 jnπ 1 1
" 1 #1 3
1 t e jnπt
¼ ejnπt 5
2 jnπ 1 ðjnπ Þ2 1
" 1 #
1 t
¼ ejnπt þ 0
2 jnπ 1
jnπ
1 ðe þe Þ jnπ
¼
2 jnπ
Since ejnπ + e jnπ ¼ 2(1)n
ð1Þnþ1
an ¼
jnπ
3.2 Trigonometric Fourier Series Representation
The trigonometric Fourier series representation of a periodic signal x(t) is expressed

by
a0 Xþ1
xð t Þ ¼ þ ðan cos ðnΩ0 t Þ þ bn sin ðnΩ0 t ÞÞ ð3:21Þ
2 n¼1
The Fourier coefficients an and bn are given by

ðT0
2
an ¼ xðt Þ cos ðnΩ0 t Þdt ð3:22aÞ
T0 0
ðT0
2
bn ¼ xðt Þ sin ðnΩ0 t Þdt ð3:22bÞ
T0 0
3.2 Trigonometric Fourier Series Representation 129
3.2.1 Symmetry Conditions in Trigonometric Fourier Series
If x(t) is an even periodic signal, then

ð
2 T 0=2
a0 ¼ xðt Þdt ð3:23aÞ
T0 0
ð
4 T 0=2
an ¼ xðt Þ cos ðnΩ0 t Þdt ð3:23bÞ
T0 0
bn¼ 0 for all n.

If x(t) is an odd periodic signal, thena0¼ 0;an ¼ 0 for all n
ð T 0=2
4
bn ¼ xðt Þ sin ðnΩ0 t Þdt ð3:24Þ
T0 0
Therefore, for every even signal bn ¼ 0. Hence, Fourier series of an even signal
contains DC term and cosine terms only. Fourier series of an odd signal contains sine
terms only.
Example 3.11 Find the trigonometric Fourier series representation of the periodic
signal shown in Figure 3.7 with A ¼ 3 and period T0¼ 2π.
Solution The period T0¼2π. Ω0 ¼ T2π0 ¼ 1.
The periodic signal x(t) defined over one period is

3 π t < 0
xð t Þ ¼
3 0t<π
Since x(t) has odd symmetry, a0 ¼ 0 and an ¼ 0

ð
4 0
bn ¼ 3 sin ðnt Þ dt
2π π
6 cos ðnt Þ 0
¼ π
π n
6
¼ 1 cos ððnπ ÞÞ
(
nπ
0 for n even
¼ 12
for n odd
nπ

Figure 3.8 Periodic triangle wave
The trigonometric Fourier series representation of x(t) is given by
X
1
12 X
1
sin ðð2n 1Þt Þ
xð t Þ ¼ bn sin ðnt Þ ¼
n¼1
π n¼1 ð2n 1Þ
Example 3.12 Find the trigonometric Fourier series representation of the periodic
triangle wave shown in Figure 3.8 with A ¼ 2 period T0 ¼ 2.
Solution The period T0¼2. Ω0 ¼ T2π0 ¼ π.
The periodic triangle wave x(t) defined over one period with A¼2 is

4t j t j< 1=2
xð t Þ ¼
4 ð1 t Þ 1=2 < t < 3=2
Since x(t) has odd symmetry, a0 ¼ 0 and an ¼ 0

ð 1=2
4
bn ¼ 4t sin ðnπt Þ dt
2 1=2
" ð 1=2 #
1=2 1
¼8 cos ðnπt Þ1=2 þ
t
nπ cos ðnπt Þ dt
nπ 1=2
h i
¼ 8 0 þ n21π 2 sin ðnπt Þ1=2
1=2
nπ

2
¼ 8 0 þ 2 2 sin
nπ 2
16 nπ
¼ 2 2 sin
n π 2
The trigonometric Fourier series representation of x(t) is

16 1 1 1
xð t Þ ¼ sin ðπt Þ sin ð3πt Þ þ sin ð5πt Þ sin ð7πt Þ þ
π2 9 25 49
Example 3.13 Find the trigonometric Fourier series representation of the following
periodic signal with period T0 ¼ 2.
3.2 Trigonometric Fourier Series Representation 131
8
>
> 1 t
1
>
<0 2
xð t Þ ¼ 1 1
>
> cos ð3πt Þ t<
>
: 2 2
0 1=2 t < 1
Solution The period T0¼2. Ω0 ¼ T2π0 ¼ π.

ð 1=2 ð 1=2
2 2
Since x(t) has even symmetry, bn ¼ 0 a0 ¼ 0 dt þ cos ð3πt Þ dt
2 1 2 1=2
ð1
þ22 0 dt
1=2
1=2
sin ð3πt Þ
¼ 3π 1
2
2
¼
3π
ð 1=2
2
an ¼ cos ð3πt Þ cos ðnπt Þ dt
2 1=2
For n ¼ 1,
ð 1=2
a1 ¼ cos ð3πt Þ cos ðπt Þ dt ¼ 0
1=2
For n ¼ 2,
ð 1=2
6
a2 ¼ cos ð3πt Þ cos ð2πt Þ dt ¼
1=2 5π
For n ¼ 3
ð 1=2
1
a3 ¼ cos ð3πt Þ cos ð3πt Þ dt ¼
1=2 2
For n ¼ 4,5,6,. . . .
ð 1=2
an ¼ cos ð3πt Þ cos ðnπt Þ dt
1=2
nπ
6 cos
¼ 2
n2 π 9π
The trigonometric Fourier series representation of x(t) is
a0 Xþ1
xð t Þ ¼ þ an cos ðnπt Þ
2 n¼1 nπ
1 6 1 X
1 6 cos
¼ þ cos ð2πt Þ þ cos ð3πt Þ 2 cos ðnπt Þ
3π 5π 2 n 2 π 9π
n¼4
Example 3.14 If the input to the half-wave rectifier is an AC signal x(t) ¼ cos(2πt),
find trigonometric Fourier series representation of output signal of the half-wave
rectifier.
Solution The output y(t) of half-wave rectifier is

xð t Þ for xðt Þ 0
yð t Þ ¼
0 for xðt Þ < 0
The sinusoidal input and output of half-wave rectifier are shown in Figure 3.9.
Since y(t) is a real and even function, its Fourier coefficients are real and even.
The period T0¼1. Ω0 ¼ T2π0 ¼ 2π
Input x(t)=cos(2p t)
1
0.5
−0.5
−1
−1.5 −1.25 −1 −0.75 −0.5 −0.25 0 0.25 0.5 0.75 1 1.25 1.5
Output y (t )
1
0.5
−0.5
−1
−1.5 −1.25 −1 −0.75 −0.5 −0.25 0 0.25 0.5 0.75 1 1.25 1.5
t (sec)
Figure 3.9 Input and output of half-wave rectifier

3.3 The Continuous Fourier Transform for Nonperiodic Signals 133
For n ¼ 0,
ð 1=4
1 1
a0 ¼ cos ð2πt Þdt ¼
1 1=4 π
Hence, the DC component is π1

For n6¼0
ð 1=4
2
an ¼ cos ð2πt Þ cos ðnΩ0 t Þdt
1 1=4
"ð ð 1=4 #
2 1=4
¼ cos ð2π ðn þ 1Þt Þdt þ cos ð2π ðn 1Þt Þdt
1 1=4 1=4
" #
sin ð2π ðn þ 1Þt Þ1=4 sin ð2π ðn 1Þt Þ1=4
¼ þ
2π ðn þ 1Þ 1=4 2π ðn þ 1Þ 1=4
2 π
π
3
2 sin ðn þ 1Þ 2 sin ðn þ 1Þt
¼4 2 þ 2 5
2π ðn þ 1Þ 2π ðn þ 1Þ
2 π
π
3 π
1 cos n cos n 2 cos n

¼ 4 2 2 5¼ 2
π nþ1 n1 π ð 1 n2 Þ
The trigonometric Fourier series representation of y(t) is
a0 X þ1
xð t Þ ¼ þ an cos ðnπt Þ
2 n¼1 π
1 X1 2 cos n
¼ þ 2 cos nπt
2π n¼4 π ð1 n2 Þ
3.3 The Continuous Fourier Transform

for Nonperiodic Signals
Consider a nonperiodic signal x(t) as shown in Figure 3.10 (a) with finite duration,
i.e., x(t) ¼ 0 for |t| >T1. From this nonperiodic signal, a periodic signal ~x ðt Þ can be
constructed as shown in Figure 3.10(b).
The Fourier series representation of ~x ðt Þ is
X1
~x ðt Þ ¼ n¼1
an ejnΩ0 t ð3:25aÞ
x(t)
− 1 0 t
1
(a)
~
(t)
−2 0 − − 1 0 2 0
t
0 1 0
(b)
Figure 3.10 (a) Nonperiodic signal (b) periodic signal obtained from (a)
ð T 0 =2
1
an ¼ ~x ðt ÞejnΩ0 t dt ð3:25bÞ
T0 T 0 =2
Since ~x ðt Þ ¼ xðt Þ for j t j< T20 and also since x(t) ¼ 0 outside this interval, so
we have
ð T 0 =2 ð1
1 jnΩ0 t 1
an ¼ xðt Þe dt ¼ xðt ÞejnΩ0 t dt ð3:26Þ
T0 T 0 =2 T0 1
Define
ð1
X ðjΩÞ ¼ xðt ÞejΩt dt ð3:27Þ
1
Then
1
an ¼ X ðjnΩ0 Þ ð3:28Þ
T0
and ~x ðt Þ can be expressed in terms of X( jΩ), that is,
P1 1
~x ðt Þ ¼ n¼1 X ðjnΩ0 ÞejnΩ0 t
T0 ð3:29Þ
1 X1
¼ X ðjnΩ0 ÞejnΩ0 t Ω0
2π n¼1
As T0 tends to infinity, ~x ðt Þ ¼ xðt Þ and summation becomes integration,

Eq. (3.29) becomes
ð1
1
xð t Þ ¼ X ðjΩÞejΩt dΩ ð3:30Þ
2π 1
Eq. (3.27) is referred to as the Fourier transform of x(t), and Eq. (3.30) is called
the inverse Fourier transform.
3.3.1 Convergence of Fourier Transforms
The sufficient conditions referred to as the Dirichlet conditions for the convergence
of Fourier transform are:
1. x(t) must be absolutely integrable, that is,
ð1
j xðt Þ j dt < 1 ð3:31Þ
1
2. x(t) must have a finite number of maxima and minima within any finite interval.
3. x(t) must have a finite number of discontinuities within any finite interval, and
each of these discontinuities is finite.
Although the above Dirichlet conditions guarantee the existence of the Fourier
transform for a signal, if impulse functions are permitted in the transform, signals
which do not satisfy these conditions can have Fourier transforms.
Example 3.15 Determine x(0) and X(0) using the definitions of the Fourier trans-
form and the inverse Fourier transform
Solution By the definition of Fourier transform, we have
ð1
X ðjΩÞ ¼ F½xðt Þ ¼ xðt ÞejΩt dt
1
Substituting Ω ¼ 0 in this equation, we obtain

ð1
X ð 0Þ ¼ xðt Þ dt
1
By the definition of the inverse Fourier transform, we have

ð1
1
xð t Þ ¼ X ðjΩÞ ejΩt dΩ
2π 1
Substituting t¼0, it follows that

ð1
1
x ð 0Þ ¼ X ðjΩÞ dΩ
2π 1
3.3.2 Fourier Transforms of Some Commonly Used

Continuous-Time Signals
The unit impulse

The Fourier transform of the unit impulse function is given by
ð1
F½δðt Þ ¼ δðt Þ ejΩt dt ¼ 1 ð3:32Þ
1
implying that the Fourier transform of the unit impulse contribute equally at all
frequencies
Example 3.16 Find the Fourier transform of x(t) ¼ ebtu(t) for b > 0.
Solution ð1 ð1
bt jΩt
F½xðt Þ ¼ e uðt Þe ebt ejΩt dt
dt ¼
ð1 0
1
ðjΩþbÞt 1 ðjΩþbÞt 1
¼ e dt ¼ e
0 jΩ þ b 0
1
¼ b>0
b þ jΩ
Example 3.17 Find the Fourier transform of x(t) ¼ 1.

Solution By definition of the inverse Fourier transform and sampling property of
the impulse function, we have
ð1
1 1
F1 ½δðΩÞ ¼ δðΩÞejΩt dΩ ¼
2π 1 2π
1
Hence, F 2π ¼ δðΩÞ and thus F½1 ¼ 2πδðΩÞ
Example 3.18 Find the Fourier transforms of the following
(i) sin (Ω0t)
(ii) cos (Ω0t)
Solution
jΩ0 t
0ejΩ0 t
(i) sin ðΩ0 tÞ ¼ e 2j
By the sampling property of the impulse function, we have

ð
1 1 1 1
F ½δðΩ Ω0 Þ ¼ δðΩ Ω0 ÞejΩt dΩ ¼ ejΩ0 t
2π 1 2π
1 jΩ t
Hence, F 2π e 0 ¼ δ ð Ω Ω0 Þ
jΩ t
Thus F e 0
̀ ¼ 2πδðΩ Ω0 Þ, F ejΩ0 t ̀ ¼ 2πδðΩ þ Ω0 Þ
Therefore, F½ sin ðΩ0 tÞ ¼ πj ½δðΩ Ω0 Þ δðΩ þ Ω0 Þ
jΩ0 t
þejΩ0 t
(ii) cos ðΩ0 tÞ ¼ e 2
jΩ0 t
e þ ejΩ0 t
F½ cos ðΩ0 tÞ ¼ F ¼ π½δðΩ Ω0 Þ þ δðΩ þ Ω0 Þ
2
Example 3.19 Find the Fourier transform of the rectangular pulse signal shown in
Figure 3.11
Solution
1 jt j T 1
xð t Þ ¼
0 jt j > T 1
ð1
XðjΩÞ ¼ F½xðtÞ ¼ xðtÞejΩt dt
ð1
T1
¼ ejΩt dt
T 1
1 jΩt T 1
¼ e jT 1
jΩ
ejΩT1 ejΩT1
¼2
2jΩ
sin ðΩT 1 Þ
¼2
Ω
sin ðπt Þ sin ðΩT 1 Þ
Since sinc ðt Þ ¼ , 2 can be written in terms of the sinc
πt Ω
function as

sin ðΩT 1 Þ ΩT 1
2 ¼ 2T 1 sin c
Ω π
Figure 3.11 Rectangular x(t)

pulse signal
− 1 0
1
t
Figure 3.12 Fourier ( Ω)

transform of a signal
1
Ω
-Ω Ω
Hence,

ΩT 1
X ðjΩÞ ¼ F½xðt Þ ¼ 2T 1 sinc
π
Example 3.20 Consider the Fourier transform X( jΩ) of a signal shown in Fig-
ure 3.12. Find the inverse Fourier transform of it.
Solution
1 jΩj Ω
X ðjΩÞ ¼
0 jΩj > Ω
By the inverse Fourier transform definition, we have

ð1
1
xðt Þ ¼ X ðjΩÞejΩt dΩ
2π 1
ðΩ
1
¼ ejΩt dΩ
2π Ω

1 1 jΩt Ω
¼ e Ω
2π jΩ

1 ejΩt þ ejΩt
¼
2π jt

sin ðΩt Þ Ω Ωt
¼ ¼ sinc
πt π π
Example 3.21 Determine the Fourier transform of Gaussian signal xðt Þ ¼ et2σ2
2
Solution ð1
t 2 jΩt
XðjΩÞ ¼ F½xðt Þ ¼ e dt
1 e2σ2
Letting b ¼ 1
2σ 2
Ð1
ebt ejΩt dt
2
XðjΩÞ ¼ 1
Ð1 bðt 2 þðjΩ=bÞÞ
¼ 1 e dt
Ð 1 cðtþðjΩ=2bÞÞ2 Ω2 =4b
¼ 1 e dt
Ð 1 2
eΩ =4b 1 ebðtðjΩ=2bÞÞ Ω =4b dt
2 2
¼
pffiffiffiffiffi pffiffiffi
Letting τ ¼ t b, dτ ¼ b dt

2
Ð
=4b 1
b tðjΩ=2bÞ Ω2 =4b
XðjΩÞ ¼ eΩ
2
1 e
2
dt
ð pffiffi
eΩ =4b 1 τðjΩ=2 bÞ
2
¼ pffiffiffi e dτ
b 1
ð1 pffiffi 2 pffiffiffi
Since eðτðjΩ=2 bÞÞ dτ ¼ π
1
eΩ =4b pffiffiffi

2
XðjΩÞ ¼ pffiffiffi π
b
Substituting b ¼ 2σ1 2
eΩ =4b pffiffiffi

2
XðjΩÞ ¼ pffiffiffi π
pffiffiffiffiffi σb2 Ω2 =2
¼ σ 2π e
3.3.3 Properties of the Continuous-Time Fourier Transform
Linearity If x1(t) and x2(t) are two continuous-time signals with Fourier transforms
X1( jΩ) and X2( jΩ), then the Fourier transform of a linear combination of x1(t) and
x2(t) is given by
F½a1 x1 ðt Þ þ a2 x2 ðt Þ ¼ a1 X 1 ðjΩÞ þ a2 X 2 ðjΩÞ ð3:33Þ
where a1 and a2 are arbitrary constants.

Example 3.22 Find the Fourier transform of an impulse train with period T as
given by
X
1
xð t Þ ¼ δðt kT Þ
k¼1
Solution A periodic signal x(t) with period T is expressed by
P1 2π
xð t Þ ¼ k¼1 ak ejkΩ0 t Ω0 ¼
T
Taking Fourier transform both sides, we obtain
" #
X
1
F½xðt Þ ¼ F ak e jkΩ0 t
k¼1

Using F ejΩ0 t ̀ ¼ 2πδðΩ Ω0 Þand the linearity property, we have
" #
X
1 X
1
F ak e jkΩ0 t
¼ 2π ak δðΩ kΩ0 Þ
k¼1 k¼1
If x(t) is an impulse train with period T as given by
X
1
xð t Þ ¼ δðt kT Þ
k¼1
X1 1 X1
Since k¼1
δðt TÞ ¼ ejkΩ0 t
T k¼1
" #
X1
2π X 1
F δðt kT Þ ¼ δðΩ kΩ0 Þ
k¼1
T k¼1
Symmetry for Real Valued Signal If x(t) is a continuous-time real valued signal
with Fourier transform X( jΩ), then
X ðjΩÞ ¼ X ∗ ðjΩÞ ð3:34Þ
where * stands for the complex conjugate.

Proof Ð 1 ∗
X ∗ ðjΩÞ ¼ Ð 1 xðt ÞejΩt dt
1
¼ 1 x∗ ðt ÞejΩt dt
Since x(t) is real x*(t) ¼ x(t), we get

ð1
∗
X ðjΩÞ ¼ xðt ÞejΩt dt ¼ X ðjΩÞ
1
The X( jΩ) can be expressed in rectangular form as


X ðjΩÞ ¼ Re½X ðjΩÞ þ jIm X ðjΩÞ
If x(t) is real, then
Re ½X ðjΩÞ ¼ Re½X ðjΩÞ

Im ½X ðjΩÞ ¼ Im ½X ðjΩÞ
implying that the real part is an even function of Ω and the imaginary part is an odd
function of Ω.
For real x(t) in polar form
jX ðjΩÞj ¼ jX ðjΩÞj
arg½X ðjΩÞ ¼ arg½X ðjΩÞ
indicating that the magnitude is an even function of Ω and the phase is an odd
function of Ω.
Symmetry for Imaginary Valued Signal If x(t) is a continuous-time imaginary
valued signal with Fourier transform X( jΩ), then
X ∗ ðjΩÞ ¼ X ðjΩÞ ð3:35Þ
Proof Ð 1 ∗
X ∗ ðjΩÞ ¼ 1 xðt Þe
jΩt
dt
Ð1 ∗
¼ 1 x ðt Þe dt
jΩt
Since x(t) is purely imaginary, we get x(t) ¼ x*(t), and we get

ð1
X ∗ ðjΩÞ ¼ xðt ÞejΩt dt ¼ X ðjΩÞ
1
Re ½X ðjΩÞ ¼ Re ½X ðjΩÞ

Im ½X ðjΩÞ ¼ Im ½X ðjΩÞ
Symmetry for Even and Odd Signals

(i) If x(t) is a continuous-time real valued and has even symmetry, then
X ∗ ðjΩÞ ¼ X ðjΩÞ ð3:36aÞ
(ii) If x(t) is a continuous-time real valued and has odd symmetry, then
X ∗ ðjΩÞ ¼ X ðjΩÞ ð3:36bÞ
Proof Since x(t) is real x(t) ¼ x*(t) and x(t) has even symmetry x(t) ¼ x(t), we get
ð 1 ∗
∗ jΩt
X ðjΩÞ ¼ xðt Þe dt
1
ð1
¼ x∗ ðt Þ ejΩt dt
1
ð1
¼ x ðt Þ ejΩt dt
1
ð1
¼ xðt Þ ejΩðtÞ dt
1
Letting τ ¼t, we obtain

Ð1
X ∗ ðjΩÞ ¼ 1 xðτÞejΩτ dτ
¼ X ðjΩÞ
The condition X*( jΩ) ¼ X( jΩ) holds for the imaginary part of X( jΩ) to be zero.
Therefore, if x(t) is real valued and has even symmetry, then X( jΩ) is real.
Similarly, for real valued x(t) having odd symmetry, it can be shown that X*
( jΩ)¼X( jΩ) and X( jΩ) is imaginary.
Time Shifting If x(t) is a continuous-time signal with Fourier transform X( jΩ), then
the Fourier transform of x(t-t0) the delayed version of x(t) is given by
F½xðt t 0 Þ ¼ ejΩt0 X ðjΩÞ ð3:37Þ
Proof ð1
F½xðt t 0 Þ ¼ xðt t 0 ÞejΩt dt
1
Letting τ ¼ tt0, we obtain

Ð1 jΩðτt 0 Þ
F½xðt t 0 Þ ¼ 1 xðτÞe dτ
jΩt 0
Ð 1 jΩτ
¼ e 1 xðτÞe dτ
jΩt 0
¼e X ðjΩÞ
Therefore, time shifting results in unchanged magnitude spectrum but introduces

a phase shift in its transform, which is a linear function of Ω.
Example 3.23 Find the Fourier transform of δ(tt0)
Solution
δðjΩÞ ¼ F½δðt Þ ¼ 1
Hence, F½δðt t 0 Þ ¼ ejΩt0 δðjΩÞ ¼ ejΩt0

Frequency Shifting If x(t) is a continuous-time signal with Fourier transform X

( jΩ), then the Fourier transform of the signal ejΩ0 t xðt Þ is given by

F ejΩ0 t xðt Þ ¼ X ðjðΩ Ω0 ÞÞ ð3:38Þ
Proof ð1
F½ejΩ0 t xðt Þ ¼ ejΩ0 t xðt ÞejΩt dt
1
ð1
¼ xðt ÞejðΩΩ0 Þt dt
1
¼ X ð j ð Ω Ω0 Þ Þ
Thus, multiplying a sequence x(t) by a complex exponential ejΩ0 t in the time

domain corresponds to a shift in the frequency domain.
Time and Frequency Scaling If x(t) is a continuous-time signal with Fourier
transform X( jΩ), then the Fourier transform of the signal x(at) is given by

1 Ω
F½xðat Þ ¼ X j ð3:39Þ
j aj a
where a is a real constant.

Proof ð1
F½xðat Þ ¼ xðat Þ ejΩt dt
1
Letting τ ¼ at, we obtain

ð Ω
1 1
F½xðatÞ ¼ xðτÞej a τ dτ for a > 0
ð a 1
1 1 Ω
¼ xðτÞej a τ dτ f or a < 0
a 1
Thus,

1 Ω
F½xðat Þ ¼ X j
j aj a
Differentiation in Time If x(t) is a continuous-time signal with Fourier transform X

( jΩ), then the Fourier transform of the dtd xðt Þ is given by

d
F xðt Þ ¼ jΩX ðjΩÞ ð3:40Þ
dt
Proof By the definition of the inverse Fourier transform, it is known that

ð1
1
xð t Þ ¼ X ðjΩÞ ejΩt dΩ
2π 1
Differentiating this equation both sides with respect to t, we obtain

ð
d 1 1
xð t Þ ¼ X ðjΩÞjΩejΩt dΩ
dt 2π 1
d
xðt Þ ¼ jΩxðt Þ
dt
Taking the Fourier transform of this equation both sides, we get

d
F xðt Þ ¼ jΩX ðjΩÞ
dt
Thus, differentiation in the time domain corresponds to multiplication by jΩ in

the frequency domain.
By repeated application of this property, we obtain

dn
F x ð t Þ ¼ ðjΩÞn X ðjΩÞ
dt n
Example 3.24 Determine the Fourier transform of x(t) ¼ u(t)

Solution Decomposing the unit step function into even and odd components, it is
written as
uðt Þ ¼ xe ðt Þ þ xo ðt Þ
where the even component xe ðt Þ ¼ 12 and the odd component xo ðt Þ ¼ uðt Þ 12

Hence, F½uðtÞ ¼ F½xe ðt Þ þ F½xe ðt Þ
¼ X e ðjΩÞ þ X o ðjΩÞ
1 1
X e ðjΩÞ ¼ F½xe ðt Þ ¼ F½1 ¼ 2πδðΩÞ ¼ πδðΩÞ
2 2
d d
xo ðt Þ ¼ uðt Þ ¼ δðt Þ
dt dt
d
Thus, F dtxo ðt Þ ¼ jΩX o ðjΩÞ
jΩX o ðjΩÞ ¼ F½δðt Þ ¼ 1

1
X o ðjΩÞ ¼
jΩ
Therefore, U ðjΩÞ ¼ F½uðtÞ ¼ X e ðjΩÞ þ X o ðjΩÞ
1
¼ πδðΩÞ þ
jΩ
Example 3.25 Determine the Fourier transform of

(i) x(t) ¼ sin(Ω0t)u(t)
(ii) x(t) ¼ cos(Ω0t)u(t)
Solution (i)

ejΩ0 t ejΩ0 t
sin ðΩ0 tÞuðtÞ ¼ uðtÞ
2j
1 1
F½ sin ðΩ0 tÞuðtÞ ¼ F ejΩ0 t uðtÞ F ejΩ0 t uðtÞ
2j 2j
since F½uðtÞ ¼ πδðΩÞ þ jΩ
1
By frequency shifting property, we get

π 1 1 1
F½ sin ðΩ0 tÞuðtÞ ¼ ½δðΩ Ω0 Þ δðΩ þ Ω0 Þ þ
2j 2j jðΩ Ω0 Þ jðΩ þ Ω0 Þ
π Ω0
¼ ½ δ ð Ω Ω0 Þ δ ð Ω þ Ω0 Þ þ 2
2j Ω0 Ω 2
(ii)

ejΩ0 t þ ejΩ0 t
cos ðΩ0 tÞuðtÞ ¼ uð t Þ
2
1 1
F½ cos ðΩ0 tÞuðtÞ ¼ F ejΩ0 t uðtÞ þ F ejΩ0 t uðtÞ
2 2
Since F½uðtÞ ¼ πδðΩÞ þ jΩ
1
By frequency shifting property, we get

π 1 1 1
F½ cos ðΩ0 tÞuðtÞ ¼ ½δðΩ Ω0 Þ þ δðΩ þ Ω0 Þ þ þ
2 2 jðΩ Ω0 Þ jðΩ þ Ω0 Þ
π jΩ
¼ ½δðΩ Ω0 Þ þ δðΩ þ Ω0 Þ þ 2
2 Ω0 Ω2
Example 3.26 Determine the Fourier transform of

(i) xðt Þ ¼ ebt sin ðΩ0 tÞuðt Þ b > 0
(ii) xðt Þ ¼ ebt cos ðΩ0 tÞuðt Þ b > 0
Solution (i)
jΩ0 t
e ejΩ0 t
F ebt sin ðΩ0 tÞuðt Þ ¼ F ebt uð t Þ
2j

ejΩ0 t ejΩ0 t
¼ F ebt uðt Þ F ebt uð t Þ
2j 2j
Let x1(t) ¼ ebtu(t)

Ðt Ðt
F½x1 ðtÞ ¼ ebt ejΩt dt ¼
0 0 eðbþjΩÞt dt
1
¼ b > 0:
b þ jΩ
By frequency shifting property

F ejΩ0 t x1 ðt Þ ¼ X 1 ðjðΩ Ω0 ÞÞ

ejΩ0 t 1 1
F ebt uð t Þ ¼
2j 2j b þ jðΩ Ω0 Þ
Similarly,

ejΩ0 t 1 1
F ebt uð t Þ ¼
2j 2j b þ jðΩ þ Ω0 Þ
Hence,

bt 1 1 1 1
F e sin ðΩ0 tÞuðt Þ ¼
2j b þ jðΩ Ω0 Þ 2j b þ jðΩ þ Ω0 Þ
Ω0
¼ b>0
ðb þ jΩÞ2 þ Ω0 2
(ii)
jΩ0 t jΩ0 t
e þe
F ebt cos ðΩ0 tÞuðt Þ ¼ F ebt uð t Þ
2

ejΩ0 t ejΩ0 t
¼ F ebt uðt Þ þ F ebt uð t Þ
2 2

1 1 1 1
¼ þ
2 b þ j ð Ω Ω0 Þ 2 b þ jðΩ þ Ω0 Þ
b þ jΩ
¼ b>0
ðb þ jΩÞ2 þ Ω0 2
Differentiation in Frequency If x(t) is a continuous-time signal with Fourier

transform X( jΩ), then the Fourier transform of the -jtx(t) is given by
d
F½jtxðt Þ ¼ X ðjΩÞ ð3:41Þ
dΩ
Proof By the definition of the Fourier transform, it is known that

ð1
X ðjΩÞ ¼ xðt ÞejΩt dt
1
Differentiating this equation with respect to Ω, we have

ð1
d
X ðjΩÞ ¼ jtxðt ÞejΩt dt
dΩ 1
implying that F½jtxðt Þ ¼ dΩ X ðjΩÞ

d
Thus, differentiation in the frequency domain corresponds to multiplication by

jt in the time domain.
F½jtxðt Þ ¼ dΩ
d
X ðjΩÞ can also be expressed as
d
F½txðt Þ ¼ j X ðjΩÞ
dΩ
Example 3.27 Find the Fourier transform of the following continuous-time signal:
t n1 bt
xð t Þ ¼ e uð t Þ
ðn 1Þ!
Solution For n ¼ 1, xðt Þ ¼ ebt uðt Þ, b>0
1
X ðjΩÞ ¼
b þ jΩ
For n ¼ 2, x(t) ¼ tebtu(t)
By differentiation in frequency property,

d 1
X ðjΩÞ ¼ F tebt uðt Þ ¼ j
dΩ ðb þ jΩÞ
d
¼ j ðb þ jΩÞ1
dΩ
j
¼ ð1Þðb þ jΩÞ2
1
1
¼
ðb þ jΩÞ2
t bt
2
For n ¼ 3, xðt Þ ¼ 2! e uð t Þ
" #
t 2 bt j d 1
X ðjΩÞ ¼ F e uðt Þ ¼
2! 2dΩ ðb þ jΩÞ2
j d
¼ ðb þ jΩÞ2
2dΩ
j
¼ ð2Þðb þ jΩÞ3 j
2
1
¼
ðb þ jΩÞ3
t bt 3
For n ¼ 4, xðt Þ ¼ 3! e uð t Þ
" #
t 3 bt j d 1
X ðjΩÞ ¼ F e uðt Þ ¼
3! 3dΩ ðb þ jΩÞ3
j d
¼ ðb þ jΩÞ3
3dΩ
j
¼ ð3Þðb þ jΩÞ4 j
3
1
¼
ðb þ jΩÞ4
Thus
t n1 bt
for n, xðt Þ ¼ ðn1 Þ! e uð t Þ
n1 " #
t bt j d 1
X ðjΩÞ ¼ F e uð t Þ ¼
ðn 1Þ! ðn 1ÞdΩ ðb þ jΩÞn1
j d
¼ ðb þ jΩÞnþ1
ðn 1ÞdΩ
j
¼ ðn 1Þðb þ jΩÞnþ11 j
ð n 1Þ
1
¼
ðb þ jΩÞn
Integration If x(t) is a continuous-time

ðt signal with Fourier transform X( jΩ), then
the Fourier transform of the xðτÞ dτ is given by
1
ð t
1
F xðτÞ dτ ¼ X ðjΩÞ ð3:42Þ
1 jΩ
ðt
Proof Letting yðt Þ ¼ xðτÞ dτ and differentiating both sides, we obtain
1
d
yð t Þ ¼ xð t Þ
dt
Now taking the Fourier transform of both sides, it yields

d
F yðt Þ ¼ F½xðt Þ ¼ X ðjΩÞ
dt
jΩY ðjΩÞ ¼ X ðjΩÞ
Hence
Ð t
F½yðt Þ ¼ F 1 xðτÞ dτ
1
¼ X ðjΩÞ
jΩ
Parseval’s theorem If x(t) is a continuous-time signal with Fourier transform X

( jΩ), then the energy E of x(t) is given by
ð1 ð1
21
E¼ jxðt Þj dt ¼ jX ðjΩÞj2 dΩ ð3:43Þ
1 2π 1
where |X( jΩ)|2 is called the energy density spectrum.

Proof The energy E of x(t) is defined as
ð1
E¼ jxðt Þj2 dt ð3:44Þ
1
Assuming that x(t) is complex value x(t)x∗(t) ¼ |x(t)|2 and x*(t) can be expressed
in terms of its Fourier transform as
ð1
∗ 1
x ðt Þ ¼ X ∗ ðjΩÞejΩt d Ω ð3:45Þ
2π 1

ð1
E¼ xðt Þx∗ ðt Þ dt ð3:46Þ
1
Substituting Eq (3.45) in Eq. (3.46), we obtain

ð1 ð1
1 ∗ jΩt
E¼ xð t Þ X ðjΩÞe d Ω dt ð3:47Þ
1 2π 1

ð1 ð 1
1 ∗ jΩt
E¼ X ðjΩÞ xðt Þe d t dΩ ð3:48Þ
2π 1 1
By definition of the Fourier transform

ð1
X ððjΩÞÞ ¼ xðt ÞejΩt dt
1
Thus,
ð1 ð1
21
E¼ jxðt Þj dt ¼ jX ðjΩÞj2 dΩ
1 2π 1
Example 3.28 Consider a signal x(t) with its Fourier transform given by
8
< 2 jΩj 1
X ðjΩÞ ¼ 1 1 < j Ωj 2
:
0 otherwise
(i) Determine the energy of the signal x(t)

(ii) Find x(t)
Solution (i) By Parseval’s theorem, the energy E of x(t) is given by
ð1
Ð1 2 1
E ¼ 1 jxðt Þj dt ¼ jX ðjΩÞj2 dΩ
2π 1
ð1 ð2 ð 1
1 1 1
¼ 4dΩ þ 1dΩ þ 1dΩ
2π 1 2π 1 2π 2
8 1 1
¼ þ þ
2π 2π 2π
5
¼
π
(ii)
8
<2 jΩj 1
X ðjΩÞ ¼ 1 1 < j Ωj 2
:
0 otherwise
can be written as
X ðjΩÞ ¼ X 1 ðjΩÞ þ X 2 ðjΩÞ
where

1 jΩj 1
X 1 ðjΩÞ ¼
0 jΩj > 1

1 jΩj 2
X 2 ðjΩÞ ¼
0 jΩj > 2
which are depicted as
1( W) 2( W)
1 1
W W
0 1 -2 0 2
h i
sin ðΩ0 t Þ 1 j Ωj Ω0
Since F ¼
πt 0 j Ωj > Ω0
By linearity property,
xðt Þ ¼ F1 ðX 1 ðjΩÞÞ þ F1 ðX 2 ðjΩÞÞ

sin ðt Þ sin ð2t Þ
¼ þ
πt πt
The Convolution Property

If x1(t) and x2(t) are two continuous-time signals with Fourier transforms X1( jΩ)
and X2( jΩ), then the Fourier transform of the convolution integral of x1(t) and x2(t)
is given by
F½x1 ðt Þ∗x2 ðt Þ ¼ X 1 ðjΩÞX 2 ðjΩÞ ð3:49Þ
Hence, convolution of two sequences x1(t) and x2(t) in the time domain is equal to
the product of their frequency spectra.
Proof By the definition of convolution integral,
ð1
yðt Þ ¼ x1 ðt Þ∗x2 ðt Þ ¼ x1 ðτÞx2 ðt τÞ dτ ð3:50Þ
1
Taking the Fourier transform of Eq. (3.50), we obtain

ð1 ð1
Y ðjΩÞ ¼ F½yðt Þ ¼ ½x1 ðτÞx2 ðt τÞ dτejΩt dt ð3:51Þ
1 1

ð1 ð1

Y ðjΩÞ ¼ x1 ð τ Þ x2 ðt τÞejΩt dt dτ ð3:52Þ
1 1
ð1

By the shifting property, x2 ðt τÞejΩt dt ¼ ejΩτ X 2 ðjΩÞ.
1
Hence, Eq. (3.52) becomes
ð1 ð1
YðjΩÞ ¼ x1 ðτÞejΩt X 2 ðjΩÞdτ ¼ X 2 ðjΩÞ x1 ðτÞejΩτ dτ ð3:53Þ
1 1
By ðthe definition of continuous-time Fourier transform,

1
jΩτ
X 1 ðjΩÞ ¼ x1 ðτÞe dτ
1
Thus
Y ðjΩÞ ¼ X 1 ðjΩÞX 2 ðjΩÞ
Example 3.29 Determine the Fourier transform of the triangular output signal y(t) of
an LTI system as shown in Figure 3.13.
Solution A triangular signal can be represented as the convolution of two rectan-
gular pulse signals x1(t) and x2(t) defined by

1 jt j < 1
x1 ð t Þ ¼ x2 ð t Þ ¼
0 jt j > 1
yðt Þ ¼ x1 ðt Þ∗x2 ðt Þ
By definition,
Figure 3.13 Triangular y(t)

output signal
-2 2 t
Ð1 Ð1
jΩt
X 1 ðjΩÞ ¼ 1 x1 ðt Þedt ¼ 1 ejΩt dt
1 jΩ sin Ω
¼ e ejΩ ¼ 2
jΩ Ω
By the convolution property, Y( jΩ) the Fourier transform of y(t) is given by
Y ðjΩÞ ¼ X 1 ðjΩÞX 2 ðjΩÞ

sin ðΩÞ sin ðΩÞ
¼2 2
Ω Ω
sin ðΩÞ
2
¼4
Ω2
Example 3.30 Consider an LTI continuous-time system with the impulse response
hðt Þ ¼ sin ðtΩ0 tÞ. Find the output y(t) of the system for an input
sin ð2Ω0 t Þ
xðt Þ ¼ :
t
h i
1 jΩj Ω0
Solution Since F sin ðΩ 0 tÞ
¼
πt 0 jΩj > Ω0

sin ðΩ0 t Þ
HðjΩÞ ¼ F½hðt Þ ¼ F
t
π j Ωj Ω 0
¼
0 j Ωj > Ω0

sin ð2Ω0 t Þ
XðjΩÞ ¼ F½xðt Þ ¼ F
t
π jΩj 2Ω0
¼
0 jΩj > 2Ω0
shown as
( W) ( W)
W
-2W0 0 2W 0 W -W 0 0 W0
The output y(t) of the system is given by
yðt Þ ¼ xðt Þ∗hðt Þ
By convolution property, we have
YðjΩÞ ¼ F½yðt Þ ¼ XðjΩÞHðjΩÞ

(
π 2 j Ωj Ω 0
¼
0 jΩj > Ω0
which is depicted as
( W)
W
-W 0 0 W0
Therefore,
sin ðΩ0 t Þ
yðt Þ ¼ F1 ðYðjΩÞÞ ¼ π
t
Duality property
For a given Fourier transform pair
F
xðt Þ $ X ðjΩÞ
By interchanging the roles of time and frequency, a new Fourier transform pair is
obtained as
F
X ðjt Þ $ 2πxðΩÞ
For example, the duality exists between the Fourier transform pairs of Examples
3.19 and 3.20 as given by

1 jt j T 1 F ΩT 1
xð t Þ ¼ $ X ðjΩÞ ¼ 2T 1 sin c
0 jt j > T 1 π

Ω Ωt F 1 jt j Ω
xðt Þ ¼ sin c $ X ðjΩÞ ¼
π π 0 jt j > Ω
The Modulation Property

Due to duality between the time domain and frequency domain, the multiplication in
the time domain corresponds to convolution in the frequency domain.
If x1(t) and x2(t) are two continuous-time signals with Fourier transforms X1( jΩ)
and X2( jΩ), then the Fourier transform of the product of x1(t) and x2(t) is given by
1
F½x1 ðt Þx2 ðt Þ ¼ ½X 1 ðjΩÞ∗X 2 ðjΩÞ ð3:54Þ
2π
This can be easily proved by dual property. Eq. (3.54) is called the modulation
property since the multiplication of two signals often implies amplitude modulation.
Example 3.31 Find the Fourier transform of ejΩ0 t xðt Þ
Solution Let x1 ðt Þ ¼ ejΩ0 t and x2(t) ¼ x(t)
X 1 ðjΩÞ ¼ F½ejΩ0 t ¼
` 2πδðΩ Ω0 Þ
X 2 ðjΩÞ ¼ F½xðtÞ ¼
` XðjΩÞ
1
F½x1 ðtÞx2 ðtÞ ¼ F½ejΩ0 t xðtÞ ¼ ½2πδðΩ Ω0 Þ∗XðjΩÞ
2π
¼ XðjðΩ Ω0 ÞÞ
Example 3.32 Let y(t) be the convolution of two signals x1(t) and x2(t) defined by
x1 ðt Þ ¼ sin cð2t Þ
x2 ðt Þ ¼ sin cðt Þ cos ð3πt Þ
Determine the Fourier transform of y(t).

Solution By dual property, the Fourier transform of sinc(t) is given by

Ω
F½sincðt Þ ¼ rect
2π
The Fourier transform of x1(t) is given by
F½x1 ðt Þ ¼ X 1 ðjΩÞ ¼ F½ sin cð2t Þ

1 Ω=2
¼ rect
2 2π

1 Ω
¼ rect
2 4π
The Fourier transform of x2(t) is given by
F½x2 ðt Þ ¼ X 2 ðjΩÞ ¼ F½sin c ðt Þ cos ð3πt Þ

j3πt
e þ ej3πt
¼ F sin c ðt Þ
2
By modulation property
j3πt
e þ ej3πt 1 ðΩ 3π Þ ðΩ þ 3π Þ
X 2 ðjΩÞ ¼ F sin c ðt Þ ¼ rect þ rect
2 2 2π 2π
By convolution property

F yðt Þ ¼ F½x1 ðt Þ∗x2 ðt Þ ¼ X 1 ðjΩÞX 2 ðjΩÞ

1 Ω ðΩ 3π Þ Ω þ 3π
¼ rect rect þ rect
4 4π 2π 2π
1 1
1( W) = rect W
2 4
4p -2p 0 2p 4p
1 1 (W − 3 ) (W + 3 )
2( W) = rect + rect
2 2 2
W
4p -2p 0 2p 4p
There is no overlap between the two transforms X1( jΩ) and X2( jΩ), and hence
X 1 ðjΩÞX 2 ðjΩÞ ¼ 0
Therefore,
YðjΩÞ ¼ F½x1 ðtÞ∗x2 ðtÞ ¼ X 1 ðjΩÞX 2 ðjΩÞ

¼0
Example 3.33 Determine the value of

ð1
sin c2 ð2t Þ dt
1
1

Ω
Solution Since the Fourier transform of sinc (2t) is 2 rect 4π , using Parseval’s
theorem,
ð1 ð 1 2
1 1 Ω
sin c2 ð2t Þdt ¼ rect2 dΩ
1 2π
1 2 4π
ð
1 2π
¼ 1dΩ
8π 2π
4π
¼
8π
1
¼
2
Example 3.34 Consider a signal x(t) with its Fourier transform given by

π j Ωj Ω0
X ðjΩÞ ¼
0 j Ωj > Ω0
Find the Fourier transform the system output y(t) given by
yðt Þ ¼ xðt Þ cos ðΩc t Þ where Ωc > Ω0
Solution By the definition of the inverse Fourier transform, we have

ð
1 Ω0
xð t Þ ¼ πejΩt dΩ
2π Ω0

1 1 jΩ0 t
¼ e ejΩ0 t
2 jt
sin ðΩ0 t Þ
¼
t
Hence, yðt Þ ¼ xðt Þ cos ðΩc t Þ ¼ sin ðtΩ0 tÞ cos ðΩc t Þ
By modulation property, we obtain
YðjΩÞ ¼ XðjΩÞ∗F½cos ðΩc tÞ

¼ XðjΩÞ∗½πδðΩ Ωc Þ þ πδðΩ þ Ωc Þ
W
−W − W 0 −W −W + W 0 0 W −W0 W W + W0
The Fourier transform properties of continuous-time signals are summarized in

Table 3.2 and 3.3.
Duality : for given Fourier transform pair xðt Þ $F X ðjΩÞ

X ðjt Þ $F 2π xðΩÞ ð
Ð1 1 1
Parseval’ s theorem 1 j x ð t Þ j2
dt ¼ jX ðjΩÞj2 dΩ
2π 1
Table 3.2 Some properties of continuous-time Fourier transforms

Property Aperiodic signal Fourier transform
Linearity a1x1(t)+a2x2(t) a1X1( jΩ)+a2X2( jΩ)
Time shifting x(t t0) ejΩt0 X ðjΩÞ
Symmetry x∗ ðtÞ 8 X ∗ ðjΩÞ
∗
xðtÞ real >
> X ðjΩÞ ¼ X ðjΩÞ
>
>
xe(t) (x(t) real) < Re ½X ðjΩÞ ¼ Re ½X ðjΩÞ
xo(t) (x(t) real) Im ½X ðjΩÞ ¼ Im X ðjΩÞ
>
>
>
> jX ðjΩÞj ¼ jX ðjΩÞj
:
arg½X ðjΩÞ ¼ arg½X ðjΩÞ
Re[X( jΩ)
jIm[X( jΩ)]
Time reversal x(n) X(ejω)
Frequency shifting ejΩ0 t xðt Þ X( j(ΩΩ0))
1
Ω
Time and frequency scaling x(at) jaj X j a
dt xðt Þ
Differentiation in time d jΩX( jΩ)
Differentiation in frequency jtx(t) dΩ X ðjΩÞ
d
ðt
jΩ X ðjΩÞ
Integration 1
xðτÞ dτ
1
Convolution x1(t) ∗ x2(t) Xl( jΩ)X2( jΩ)
property
2π ½X 1 ðjΩÞ∗X 2 ðjΩÞ
Modulation x1(t) x2(t) 1
property
3.4 The Frequency Response of Continuous-Time Systems 159
Table 3.3 Basic Fourier transform pairs

Signal Fourier transform
δ(t) 1
ebt uðt Þ b>0 1
bþjΩ
1 2πδ(Ω)
ejΩ0 t 2πδ(ΩΩ0)
X1
ak ejkΩ0 t X1
k¼1 2π ak δðΩ kΩ0 Þ
k¼1
X
1
2π X1
δðt kTÞ δðΩ kΩ0 Þ
k¼1
T k¼1
sin(Ω0t) π[δ(ΩΩ0)δ(ΩΩ0)]
cos(Ω0t) π[δ(ΩΩ0)+δ(ΩΩ0)]/j

1 jt j < T 1 2T 1 sin c ΩTπ 1
xðt Þ ¼
0 jt j > T 1
Periodicsquare wave with period T0 X1
2 sin ðkΩ0 T 1 Þ
1 jt j < T 1 δðΩ kΩ0 Þ
xðt Þ ¼ k¼1
k
0 T 1 jt j T 0 =2
and x(t) ¼ x(t+T0)
Ω
Ωt
π sin c π 1 jΩj Ω
X ðjΩÞ ¼
0 jΩj > Ω
δ(tt0) ejΩt0
u(t) πδðΩÞ þ jΩ
1
Sgn(t) 2
Ω 6¼ 0
jΩ
tebt uðt Þ b>0 1
ðbþjΩÞ2
n1
bt 1
t
ðn1Þ! e uðt Þ ðbþjΩÞn
t2 pffiffiffiffiffi
σ 2π e σ2Ω
2 2
e2σ 2
3.4 The Frequency Response of Continuous-Time Systems
As in chapter 2, the input output relation of a useful class of continuous-time LTI

systems satisfies the linear constant coefficient differential equation
XN dn yðtÞ X M d n xðtÞ
a
n¼0 n
¼ b n ð3:55Þ
dt n n¼0 dt n
where coefficients an and bn are real constants.
From convolution property, it is known that
Y ðjΩÞ ¼ H ðjΩÞX ðjΩÞ ð3:56Þ

Y ðjΩÞ
H ðjΩÞ ¼ ð3:57Þ
X ðjΩÞ
Applying Fourier transform to both sides of Eq. (3.55), we obtain

X X
N d n yð t Þ M d n xð t Þ
F a
n¼0 n dt n
¼F b
n¼0 n dt n
ð3:58Þ
By linearity property, Eq. (3.58) becomes

XN n X n
d yð t Þ M d xð t Þ
a F
n¼0 n
¼ bF
n¼0 n
ð3:59Þ
dt n dt n
From the differentiation property, Eq. (3.59) can be rewritten as

XN XM
n¼0
an ðjΩÞn Y ðjΩÞ ¼ n¼0
bn ðjΩÞn X ðjΩÞ ð3:60Þ

hX N i XM
n
Y ðjΩÞ n¼0
a n ð jΩ Þ ¼ X ð jΩ Þ b ðjΩÞn
n¼0 n
ð3:61Þ
Thus, the frequency response of a continuous-time LTI system is given by

PM
YðjΩÞ bn ðjΩÞn
HðjΩÞ ¼ ¼ Pn¼0 ð3:62Þ
XðjΩÞ N
k¼0 an ðjΩÞ
n
The function H(jΩ) is a rational function being a ratio of polynomials in (jΩ).
3.4.1 Distortion During Transmission
Eq. (3.56) implies that the transmission of an input signal x(t) through the system is
changed into an output signal y(t). The X( jΩ) and Y( jΩ) are the spectra of the input
and output signals, and H( jΩ) is the frequency response of the system.
During the transmission, the input signal amplitude spectrum |X( jΩ)| is changed
to|X( jΩ)||H(jΩ)|. Similarly, the input signal phase spectrum ∠X(jΩ) is changed to
∠X( jΩ) þ ∠ H(jΩ). An input signal spectral component of frequency Ω is modified
in amplitude by a |H(jΩ)| factor and is shifted in phase by an angle ∠H(jΩ).
Thus, the output waveform will be different from the input waveform during
transmission through the system introducing distortion.
3.4 The Frequency Response of Continuous-Time Systems 161
Example 3.35 Consider an LTI system described by the following differential

equation
d2 yðt Þ dyðt Þ dxðt Þ

þ3 þ 2yðt Þ ¼ 4 xð t Þ
dt 2 dt dt
Find the Fourier transform of the impulse response of the system.
Solution Apply the Fourier transform on both sides of the differential equation,
then we obtain
2
d yð t Þ dyðt Þ dxðt Þ
F þ3 þ 2yðt Þ ¼ F 4 xð t Þ
dt 2 dt dt
2
d yð t Þ dyðt Þ dxðt Þ
F þ 3F þ 2F ½ y ðt Þ ¼ 4F F½xðt Þ
dt 2 dt dt
ðjΩÞ2 YðjΩÞ þ 3jΩY ðjΩÞ þ 2YðjΩÞ ¼ 4jΩX ðjΩÞ XðjΩÞ

Ω2 þ j3Ω þ 2 YðjΩÞ ¼ ½j4Ω 1XðjΩÞ
The Fourier transform of the impulse response H(jΩ) is given by
YðjΩÞ 1 j4Ω
HðjΩÞ ¼ ¼
XðjΩÞ Ω2 j3Ω 2
Example 3.36 Find the frequency response H( jΩ) of the following circuit
Solution
dv0 ðt Þ v0 ðt Þ
i ðt Þ ¼ C þ
dt R
diðt Þ
vi ðt Þ þ L þ v0 ð t Þ ¼ 0
dt
diðt Þ
L ¼ vi ð t Þ v0 ð t Þ
dt
d2 v0 ðt Þ L dv0 ðt Þ
LC þ þ v0 ð t Þ ¼ vi ð t Þ
dt 2 R dt
Taking Fourier transform both sides of this equation, we obtain
L
LCðjΩÞ2 v0 ðjΩÞ þ jΩ v0 ðjΩÞ þ v0 ðjΩÞ ¼ vi ðjΩÞ
R

L
1 LCΩ þ jΩ v0 ðjΩÞ ¼ vi ðjΩÞ
2
R
v0 ðjΩÞ 1
H ðjΩÞ ¼ ¼
vi ðjΩÞ L
1 LCΩ2 þ jΩ
R
3.5 Some Communication Application Examples
3.5.1 Amplitude Modulation (AM) and Demodulation

Amplitude Modulation
In amplitude modulation, the amplitude of the carrier signal c(t) is varied in some
manner with the baseband signal (message signal) m(t) also known as the modulat-
ing signal.
The AM signal is given by
m(t) s(t)
c(t)
sðt Þ ¼ mðt Þ:cðt Þ ð3:63Þ

cðt Þ ¼ cos ðΩc t þ θc Þ ð3:63aÞ
For convenience, if it is assumed that θc ¼ 0
cðt Þ ¼ cos ðΩc t Þ ð3:64Þ

CðjΩÞ ¼ F½cðtÞ ¼ π½δðΩ Ωc Þ þ δðΩ þ Ωc Þ ð3:65Þ
1
SðjΩÞ ¼ F½mðtÞcðtÞ ¼ ½M ðjΩÞ∗CðjΩÞ ð3:66Þ
2π
M ðjΩÞ∗δðΩ Ωc Þ ¼ M ðjðΩ Ωc ÞÞ ð3:67Þ
1
SðjΩÞ ¼ ½M ðjðΩ Ωc ÞÞ þ M ðjðΩ þ Ωc ÞÞ ð3:68Þ
2
Eq. (3.61) implies that the AM shifts the message signal so that it is centered at
Ωc. The message signal m(t) can be recovered if Ωc >Ωm so that the replica spectra
3.5 Some Communication Application Examples 163
Ω
−Ω 0 Ω
Ω
−Ω Ω
0.5
Ω
−Ω − Ω −Ω −Ω Ω Ω − Ω Ω Ω Ω
Figure 3.14 Amplitude modulation
w(t) Low pass ˆ (t)

s(t)
filter
( Ω)
c(t)
Figure 3.15 Amplitude demodulation
do not overlap. The AM modulation in frequency domain is illustrated in

Figure 3.14.
Amplitude Demodulation
The message signal m(t) can be extracted by multiplying the AM signal s(t) by the
same carrier c(t) and passing the resulting signal through a low-pass filter as shown
in Figure 3.15.
wðt Þ ¼ sðt Þcðt Þ

¼ mðt Þc2 ðt Þ
¼ mðt Þcos 2 ðΩc t Þ ð3:69Þ
1 þ cos ð2Ωc t Þ
¼ m ðt Þ
2
The amplitude demodulation in frequency domain is illustrated in Figure 3.16.
3.5.2 Single-Sideband (SSB) AM
The double-sideband modulation is used in Section 3.5.1. By removing the upper

sideband by using a low-pass filter with cutoff frequency Ωc or a lower sideband by
high-pass filter with cutoff frequency Ωc, single-sideband modulation that requires
half the bandwidth can be used. The frequency domain representation of single-
sideband modulation is shown in Figure 3.17. However, the single-sideband mod-
ulation requires nearly ideal filters and increases the transmitter cost.
3.5.3 Frequency Division Multiplexing (FDM)
In frequency division multiplexing, multiple signals are transmitted over a single

wideband channel using a single transmitting antenna. Different carriers with ade-
quate separation are used to modulate for each of these signals with no overlap
between the spectra of the modulated signals. The different modulated signals are
summed before sending to the antenna. At the receiver, to recover a specific signal,
the corresponding frequency is extracted through a band-pass filter. The FDM
spectra for three modulated signals are shown in Figure 3.18.
3.6 Problems
1. Find the exponential Fourier series representation for each of the following
signals:
(i) x(t) ¼ cos(Ω0t)
(ii) x(t) ¼ sin(Ω0t)
(iii) xðt Þ ¼ cos 2t þ π6
(iv) x(t) ¼ sin2(t)
(v) x(t) ¼ cos(6t) þ sin (4t)
(vi) xðt Þ ¼ ½1 þ cos ð2πt Þ sin 5πt þ π4
3.6 Problems 165
0.5
Ω
−Ω − Ω −Ω −Ω Ω Ω − Ω Ω Ω Ω
Ω
−Ω Ω
0.5
Ω
− Ω −Ω Ω Ω
Ω
−Ω 0 Ω
Figure 3.16 Illustration of amplitude demodulation in frequency domain

0.5
Ω
−Ω −Ω Ω Ω Ω Ω
Figure 3.17 Single-sideband AM
Ω
Ω Ω
1
1 1
Ω Ω Ω
Ω
0.5
Figure 3.18 Illustration of FDM for three signals
2. What signal will have the following Fourier series coefficients
1 sin 2 ðnπ=2Þ
an ¼
4 ðnπ=2Þ2
3. If x1 (t) and x2(t) are periodic signals with fundamental period T0, find the Fourier
series representation of x(t) ¼ x1(t)x2(t).
4. Consider the periodic signal x(t) given by
xðt Þ ¼ ð2 þ j2Þej3t j3ej2t þ 6 þ j3ej2t þ ð2 j2Þej3t
Determine the trigonometric Fourier series representation of the signal x(t).

5. Find the Fourier series of a periodic signal x(t) with period 3 defined over one
period by

t þ 2 2 t 0
xð t Þ ¼
2 2t 0 t 1
6. Find the Fourier series of a periodic signal x(t) with period 6 defined over one
period by
3.6 Problems 167
8
>
> 0 3 t 2
>
>
< tþ2 2 t 1
xð t Þ ¼ 1 1 t 1
>
>
>
> t þ2 1t2
:
0 2t3
7. Find Fourier series of the periodic signal shown in Figure P3.1.
Figure P3.1 Periodic signal of problem 7
8. Determine the exponential Fourier series representation of the periodic signal

depicted in Figure P3.2
Figure P3.2 Periodic

signal of problem 8
9. Determine trigonometric Fourier series representation of the signal shown in

Figure P3.3
10. Plot the magnitude and phase spectrum of the periodic signal shown in
Figure P3.4
11. Find the Fourier transform of xðt Þ ¼

eat uðt Þ a > 0
t jtj 1
12. Find the Fourier transform of xðtÞ ¼
0 jtj > 1
13. Find the Fourier transform of rectangular pulse given by

1 jt j T
xð t Þ ¼
0 jt j > T
14. Find the Fourier transform of xðtÞ ¼ π24t2 sin 2 ð2tÞ

15. Find the Fourier transform of the complex sinusoidal pulse given by

ej5t jt j π
xð t Þ ¼
0 otherwise
16. Find Fourier transform of the following signal shown in Figure P3.5
Figure P3.5 Signal x(t)
17. Consider the following two signals x(t) and y(t) as shown in Figure P3.6.
Determine Fourier transform y(t) using the Fourier transform of x(t), time
shifting property, and differentiation property
3.6 Problems 169
Figure P3.6 Signals x(t) and y(t)
18. Find the inverse Fourier transform of

2cos ðΩÞ jΩj π
XðjΩÞ ¼
0 jΩj > π
19. Find the inverse Fourier transform of
jΩ
X ðjΩÞ ¼ 2
ðjΩÞ þ 3jΩ þ 2
20. Consider the following communication system shown in Figure P3.7 to transmit
two signals simultaneously over the same channel.
Figure P3.7 Communication system
Plot the spectra of x(t), y(t), and z(t) for given the following spectra of the two
input signal shown in Figure P3.8.
Figure P3.8 Spectra of x1(t) and x2(t)
21. Determine y(t) of an LTI system with input

xðt Þ ¼ an ejnΩ0 t and the following H( jΩ) depicted in Figure P3.9 where H( jΩ) is H
( jnΩ0) evaluated at frequency nΩ0.
Figure P3.9 H( jΩ) of LTI

system of problem 21
22. Sketch amplitude single-sideband modulation and demodulation if the message

signal m(t) ¼ cos(Ωmt).
Further Reading
1. Lanczos, C.: Discourse on Fourier Series. Oliver Boyd, London (1966)

2. Körner, T.W.: Fourier Analysis. Cambridge University Press, Cambridge (1989)
3. Walker, P.L.: The Theory of Fourier Series and Integrals. Wiley, New York (1986)
4. Churchill, R.V., Brown, J.W.: Fourier Series and Boundary Value Problems, 3rd edn. McGraw-
Hill, New York (1978)
5. Papoulis, A.: The Fourier Integral and Its Applications. McGraw-Hill, New York (1962)
6. Bracewell, R.N.: Fourier Transform and Its Applications, rev, 2nd edn. McGraw-Hill, New York
(1986)
7. Morrison, N.: Introduction to Fourier Analysis. Wiley, New York (1994)
8. Lathi, B.P.: Linear Systems and Signals, 2nd edn. Oxford University Press, New York (2005)
9. Oppenheim, A.V., Willsky, A.S.: Signals and Systems. Englewood Cliffs, NJ, Prentice- Hall
(1983)
Chapter 4
Laplace Transforms
The Laplace transform is a generalization of the Fourier transform of a continuous

time signal. The Laplace transform converges for signals for which the Fourier
transform does not. Hence, the Laplace transform is a useful tool in the analysis
and design of continuous time systems. This chapter introduces the bilateral Laplace
transform, the unilateral Laplace transform, the inverse Laplace transform, and
properties of the Laplace transform. Also, in this chapter, the LTI systems, including
the systems represented by the linear constant coefficient differential equations, are
characterized and analyzed using the Laplace transform. Further, the solution of
state-space equations of continuous time LTI systems using Laplace transform is
discussed.
4.1 The Laplace Transform

4.1.1 Definition of Laplace Transform
The Laplace transform of a signal x(t) is defined as

ð1
X ðsÞ ¼ Lfxðt Þg ¼ xðt Þest dt ð4:1Þ
1
The complex variable s is of the forms ¼ σ + jΩ, with a real part σ and an
imaginary part Ω. The Laplace transform defined by Eq. (4.1) is called as the
bilateral Laplace transform.

172 4 Laplace Transforms
4.1.2 The Unilateral Laplace Transform
The unilateral Laplace transform plays an important role in the analysis of causal
systems described by constant coefficient linear differential equations with initial
conditions.
The unilateral Laplace transform is mathematically defined as
ð1
X ðsÞ ¼ Lfxðt Þg ¼ xðt Þest dt ð4:2Þ
0þ
The difference between Eqs. (4.1) and (4.2) is on the lower limit of the integra-
tion. It indicates that the bilateral Laplace transform depends on the entire signal,
whereas the unilateral Laplace transform depends on the right-sided signal, i.e.,
x(t) ¼ 0 for t < 0.
4.1.3 Existence of Laplace Transforms
The Laplace transform is said to exist if the magnitude of the transform is finite, that
is, |X(s)| < 1.
Piecewise continuous A function x(t) is piecewise continuous on a finite interval
a t b, if x is continuous on [a,b], except possibly at finitely many points at each
of which x has a finite left and right limit.
Sufficient Condition
The sufficient condition for existence of Laplace transforms is that if x(t) is piecewise
continuous on (0, 1) and there exist some constants k and M such that |x(t)| Mekt,
then X(s) exists for s > k.
Proof As x(t) is piecewise continuous on (0, 1), x(t)est is integrable on (0, 1).
ð 1 ð1 ð1

st
jLfxðt Þgj ¼ xðt Þe dt jxðt Þjest dt Mekt est dt
0 0 0
M ðskÞt 1 M M
¼ e ¼ ð 0 1Þ ¼ ð4:3Þ
ks 0
ks sk
For s > k, |ℒ{x(t)}| < 1 .
4.1.4 Relationship Between Laplace Transform and Fourier

Transform
When the complex variable s is purely imaginary, i.e., s ¼ jΩ, Eq. (4.1) becomes
4.1 The Laplace Transform 173
ð1
X ðjΩÞ ¼ xðt ÞejΩt dt ð4:4Þ
1
Eq. (4.4) is the Fourier transform of x(t), that is,

X ðsÞs¼jΩ ¼ Ffxðt Þg ð4:5Þ
If s is not purely imaginary, Eq. (4.1) can be written as

ð1
X ðσ þ jΩÞ ¼ xðt Þ eðσþjΩÞt dt ð4:6Þ
1

ð1
X ðσ þ jΩÞ ¼ xðt Þeσt ejΩt dt ð4:7Þ
1
The right hand side of Eq. (4.7) is the Fourier transform of x(t)eσt. Thus, the
Laplace transform can be interpreted as the Fourier transform of x(t) after multipli-
cation by a real exponential signal.
4.1.5 Representation of Laplace Transform in the S-Plane
The Laplace transform is a ratio of polynomials in the complex variable, which can
be represented by
N ðsÞ
X ðsÞ ¼ ð4:8Þ
DðsÞ
where N(s) is the numerator polynomial and D(s) represents the denominator
polynomial. The Eq. (4.8) is referred to as rational. The roots of the numerator
polynomial are referred to as zeros of X(s) ¼ 0 because for those values of s, X(s)
becomes zero. The roots of the denominator polynomial are called the poles of X(s),
as for those values of s, X(s) ¼ 1. A rational Laplace transform can be specified by
marking the locations of poles and zeros by x and o in the s-plane, which is called as
pole-zero plot of the Laplace transform. For a signal, the Laplace transform con-
verges for a range of values of s. This range is referred to as the region of
convergence (ROC), which is indicated as shaded region in the pole-zero plot.
4.2 Properties of the Region of Convergence
Property 1 ROC of X(s) consists of strips parallel to the jΩ axis.

The ROC of X(s) contains the values of s ¼ σ + jΩ for which the Fourier transform of
x(t)eσt converges. Thus, the ROC of X(s) is on the real part of s not on the frequency
Ω. Hence, ROC of X(s) contains strips parallel to the jΩ axis
Property 2 ROC of a rational Laplace transform should not contain poles.
In the ROC, X(s) should be finite for all s since X(s) is infinite at a pole and
Eq. (4.1) does not converge at a pole. Hence, the ROC should not contain poles
Property 3 ROC is the entire s-plane for a finite duration x(t), if there is at
least one value of s for which the Laplace transform converges.
Proof A finite duration signal is zero outside a finite interval as shown in Figure 4.1.
Let us assume that x(t)eσt is absolutely integrable for some value of σ ¼ σ 1 such that
ð t2
jxðt Þjeσ1 t < 1 ð4:9Þ
t1
Then, the line ℜe(s) ¼ σ 1 is in the ROC. For ℜe(s) ¼ σ 2 also to be in the ROC, it
is required that
ð t2 ð t2
σ 2 t
jxðt Þje ¼ jxðt Þjeσ 1 t eðσ2 σ1 Þt < 1 ð4:10Þ
t1 t1
If σ 2 > σ 1 such that eðσ 2 σ 1 Þt is decaying, then the maximum value of eðσ 2 σ 1 Þt
becomes eðσ 2 σ1 Þt1 for nonzero x(t) over the interval.
Hence,
ð t2 ð t2
σ 2 t ðσ 2 σ 1 Þt 1
jxðt Þje <e jxðt Þjeσ 1 t ð4:11Þ
t1 t1
The RHS of Eq. (4.11) is bounded and hence the LHS. Thus, the ℜe(s) > σ 1 must
also be in the ROC. Similarly, if σ 2 < σ 1, it can be shown that xðt Þeσ 2 t is absolutely
integrable. Hence, the ROC is the entire s-plane.
Figure 4.1 Finite duration

signal
t
4.2 Properties of the Region of Convergence 175
Property 4: If ROC of a right-sided signal contains the line ℜe(s) ¼ σ 1, then

ℜe(s) > σ 1will also be in the ROC for all values of s.
Proof For a right-sided signal, x(t) ¼ 0 prior to some finite time t1 as shown in
Figure 4.2
If the Laplace transform converges for some value of σ ¼ σ 1, then
ð1
jxðt Þjeσ 1 t < 1 ð4:12Þ
1
If x(t) is right sided, then

ð1
jxðt Þjeσ 1 t < 1 ð4:13Þ
t1
For σ 2 > σ 1 , xðt Þ eσ 2 t is absolutely integrable as eσ 2 t decays faster than eσ1 t as
t ! 1. Thus, ℜe(s) > σ 1 will also be in the ROC for all values of s.
Property 5: If ROC of left-sided signal contains the line ℜe(s) ¼ σ1, then ℜe(s)
< σ1 will also be in the ROC for all values of s.
Proof For left-sided signal, x(t) ¼ 0 after some finite time t2 as shown in Figure 4.3.
This can be proved easily with the same argument and intuition for the property 4.
Figure 4.2 Right-sided x(t)

signal
Figure 4.3 Left-sided x(t)

signal
t
Property 6: If ROC of a two-sided signal contains the line ℜe(s) ¼ σ0, then
ROC will contain a strip, which includes the line.
Proof A two-sided signal is of infinite duration for both t > 0 and t < 0 as shown in
Figure 4.4(a)
Let us choose an arbitrary time t0 that divides the signal into as sum of right-sided
signal and left-sided signal as shown Figure 4.4(b) and (c). The Laplace transform of
x(t) converges for the values of s for which both the right-handed signal and left-
handed signal converge. It is known from property 4 that the ROC of Laplace
transform of right-handed signal Xr(s) consists of a half plane ℜe(s) > σ r for some
value σ r; and from property 5, it is known that Xr(s) consists of a half plane ℜe
(s) > σ l for some value σ l. Then the overlap of these two half planes is the ROC of the
two-sided signal x(t) as shown in Figure 4.4(d) with the assumption that σ r < σ l. If σ r
is not less than σ l, then there is no overlap. In this case, X(s) does not exist even Xr(s)
and Xl(s) individually exist.
4.3 The Inverse Laplace Transform
From Eq. (4.6), it is known that the Laplace transform X(σ þ jΩ) of a signal x(t) is
given by
ð1
X ðσ þ jΩÞ ¼ xðt Þ eσt ejΩt dt ð4:14Þ
1
Applying the inverse Fourier transform on the above relationship, we obtain

ð1
1
xðtÞeσt ¼ F 1 fXðσ þ jΩÞg ¼ ðXðσ þ jΩÞÞ ejΩt dΩ ð4:15Þ
2π 1
Multiplying both sides of Eq. (4.15) by eσt, it follows that

ð1
1
xðtÞ ¼ ðXðσ þ jΩÞÞ eðσþjΩÞt dΩ ð4:16Þ
2π 1
As s ¼ σ + jΩ and σ is a constant, ds ¼ j dΩ.

Substituting s ¼ σ + jΩ ds ¼ j dΩ in Eq. (4.16) changing the variable of
integration from s to Ω, we arrive at the following inverse Laplace transform
ð σþj1
1
xð t Þ ¼ X ðsÞ est ds ð4:17Þ
2πj σj1
(a) (b)
x(t) x(t)
t t
4.3 The Inverse Laplace Transform
(c) (d) Im
s-plane
Re
t
177
Figure 4.4 (a) Two-sided signal (b) Right-sided signal. (c) Left-sided signal. (d) ROC of the two-sided signal
4.4 Properties of the Laplace Transform
Linearity If x1(t) and x2(t) are two signals with Laplace transforms X1(s) and X2(s)
and ROCs R1 and R2, respectively, then the Laplace transform of a linear combina-
tion of x1(t) and x2(t) is given by
Lfa1 x1 ðtÞ þ a2 x2 ðtÞg ¼ a1 X 1 ðsÞ þ a2 X 2 ðsÞ ð4:18Þ
whose ROC is at least (R1 \ R2), a1 and a2 being arbitrary constants.

Proof ð1
Lfa1 x1 ðtÞ þ a2 x2 ðtÞg ¼ fa1 x1 ðtÞ þ a2 x2 ðtÞg est dt
1
ð1 ð1 ð4:19Þ
¼ a1 x1 ðtÞest dt þ a2 x2 ðtÞest dt
1 1
¼ a1 X 1 ðsÞ þ a2 X 2 ðsÞ ð4:20Þ
The result concerning the ROC follows directly from the theory of complex
variables concerning the convergence of a sum of two convergent series.
Time Shifting If x(t) is a signal with Laplace transform X(s) and ROC R, then for
any constant t0 0, the Laplace transform of x(t – t0) is given by
Lfxðt t 0 Þg ¼ est0 X ðsÞ
whose ROC is the same as that of X(s).

Proof ð1
Lfxðt t 0 Þg ¼ xðt t 0 Þ est dt ð4:22Þ
1
Substituting τ ¼ t – t0,
ð1
L fx ð t t 0 Þ g ¼ xðτÞ esðτþt0 Þ dτ
1
ð1
¼e st0
xðτÞ esτ dτ
1 ð4:23Þ
st0
¼e X ðsÞ
Shifting in the s-domain. If x(t) is a signal with Laplace transform X(s) and ROC R,
then the Laplace transform of the signal es0 t xðt Þ is given by
Lfes0 t xðt Þg ¼ X ðs s0 Þ ð4:24Þ
whose ROC is the R þ ℜe(s)

4.4 Properties of the Laplace Transform 179
Proof ð1
Lfes0 t xðt Þg ¼ es0 t xðt Þest dt
1
ð1
ð4:25Þ
¼ xðt Þeðss0 Þt dt
1
¼ X ðs s0 Þ
Time Scaling. If x(t) is a signal with Laplace transform X(s) and ROC R, then the
Laplace transform of the x(at) for any constant a, real or complex, is given by
1 s
Lfxðat Þg ¼ X ð4:26Þ
a a
whose ROC is the Ra
Proof ð1
Lfxðat Þg ¼ xðat Þ est dt ð4:27Þ
1
Letting τ ¼ at; dt ¼ dτ/a

Then,
Ð1 τ1
Lfxðat Þg ¼ 1 xðτÞ esa dτ
a
ð
1 1 sτ
¼ xðτÞe a dτ
a 1
1 s ð4:28Þ
¼ X
a a
Differentiation in the Time Domain. If x(t) is a signal with the Laplace transform X
(s) and ROC R, then

dx
L ¼ sX ðsÞ ð4:29Þ
dt
with ROC containing R.

Proof This property can be proved by differentiating both sides of the inverse
Laplace transform expression
ð σþj1
1
xð t Þ ¼ X ðsÞ est ds
2πj σj1
Then,
ð σþj1
dx 1
¼ sX ðsÞest ds ð4:30Þ
dt 2πj σj1
From the above expression, it can be stated that the inverse Laplace transform of
sX(s) is dx
dt .
Differentiation in the s-Domain. If x(t) is a signal with the Laplace transform X(s),
then
dX ðsÞ
¼ Lftxðt Þg ð4:31Þ
ds
Proof From the definition of the Laplace transform,

ð1
X ð s Þ ¼ L fx ð t Þ g ¼ xðt Þ est dt
1
Differentiating both sides of the above equation, we get

ð1
dX ðsÞ
¼ txðt Þ est dt
ds 1 ð4:32Þ
¼ Lftxðt Þg
with ROC ¼ R
Division by t
If x(t) is a signal with Laplace transform X(s), then
ð1
xðt Þ
L ¼ X ðuÞ du ð4:33Þ
t s
h i
xð t Þ
provided that lim t exists.
t!0
Proof Let x1 ðt Þ ¼ xðttÞ, then (t) ¼ t x1(t).

By using the differentiation in the s-domain property,
d
X ðsÞ ¼ L fx 1 ð t Þ g ð4:34Þ
ds
dLfx1 ðt Þg ¼ X ðsÞds ð4:35Þ
Integrating both sides of Eq. (4.35) yields

ð
Ð
dLfx1 ðt Þg ¼ X ðsÞ ds
ðs
L fx 1 ð t Þ g ¼ X ðuÞ du ð4:36Þ
1
ð1
L fx 1 ð t Þ g ¼ X ðuÞ du
s
Integration. If x(t) is a signal with Laplace transform X(s) and ROC R, then
ð t
1
L xðt Þ dt ¼ X ðsÞ ð4:37Þ
1 s
with ROC containing R \ ℜe(s) > 0.

Proof This property can be proved by integrating both sides of the inverse Laplace
transform expression
ð σþj1
1
xð t Þ ¼ X ðsÞest ds
2πj σj1
Then,
ðτ ð σþj1
1 1
xðτÞ dτ ¼ X ðsÞ est ds
1 2πj σj1 s
Ðτ
Consequently, the inverse Laplace transform of 1s X ðsÞ is 1 xðτÞ dτ:
Convolution in the Time Domain. If x1(t) and x2(t) are two signals with Laplace
transforms X1(s) and X2(s) with ROCs R1 and R2, respectively, then
Lfx1 ðt Þ ∗ x2 ðt Þg ¼ X 1 ðsÞX 2 ðsÞ ð4:38Þ
with ROC containing R1 \ R2

Proof ð 1 ð 1
Lfx1 ðt Þ ∗ x2 ðt Þg ¼ x1 ðτÞx2 ðt τÞ dτ est dt ð4:39Þ
1 1
Changing the order of integration, Eq. (4.39) can be rewritten as

ð1 ð 1
st
L fx 1 ð t Þ ∗ x 2 ð t Þ g ¼ x1 ð τ Þ x2 ðt τÞe dt dτ ð4:40Þ
1 1
Let t1 ¼ t – τ; dt1 ¼ dt;

ð1 ð 1
sτ st 1
L fx 1 ð t Þ ∗ x 2 ð t Þ g ¼ x1 ð τ Þ e x2 ðt 1 Þe dt 1 dτ
1 1
ð1 ð 1
sτ sτ ð4:41Þ
¼ e x1 ðτÞ X 2 ðsÞdτ ¼ e x1 ðτÞdτ X 2 ðsÞ
1 1
¼ X 1 ðsÞX 2 ðsÞ
The ROC includes R1 \ R2 and is large if pole-zero cancellation occurs.

Convolution in the Frequency Domain
If x1(t) and x2(t) are two signals with Laplace transforms X1(s) and X2(s), then
ð cþj1
1
Lfx1 ðt Þx2 ðt Þg ¼ X 1 ðpÞX 2 ðs pÞdp ð4:42Þ
2πj cj1
Proof Let x(t) ¼ x1(t)x2(t)

with Laplace transforms X1(s) and X2(s) and areas of convergence ℜe(s) > σ 1 and
ℜe(s) > σ 2, respectively, then
ð1
L fx ð t Þ g ¼ x1 ðt Þ x2 ðt Þ est dt ð4:43Þ
0
According to the inverse integral,

ð cþj1
1
x1 ð t Þ ¼ X 1 ðpÞ ept dp, c > σ 1 ð4:44Þ
2πj cj1
Substituting this relationship in Eq. (4.43), it follows that

ð1 ð cþj1
1
L fx ð t Þ g ¼ x2 ðt Þest X 1 ðpÞept dp dt ð4:45Þ
0 2πj cj1
Permuting the sequence of integration, we obtain

ð cþj1 ð1
1
X ðsÞ ¼ X 1 ðpÞdp x2 ðt ÞeðspÞt dt ð4:46Þ
2πj cj1 0
where
ð1
X 2 ðs p Þ ¼ x2 ðt ÞeðspÞt dt ð4:47Þ
0
This integral converges for ℜe(s p) > σ 2. By substituting Eq. (4.47) in

Eq. (4.46), yields the following proving the property
ð cþj1
1
X ðsÞ ¼ X 1 ðpÞX 2 ðs pÞdp ð4:48Þ
2πj cj1
The above properties of the Laplace transform are summarized in Table 4.1.
4.4.1 Laplace Transform Properties of Even and Odd

Functions
Even Property
If x(t) is an even function such that x(t) ¼ x(–t), then X(s) ¼ X(s).
Proof ð1
X ðsÞ ¼ xðt Þ est dt ð4:49Þ
1
Consider
ð1
X 1 ðsÞ ¼ xðt Þ est dt ð4:50Þ
1
Let t1 ¼ t; then,

ð1
X 1 ðsÞ ¼ xðt 1 Þ est1 dt 1
1 ð4:51Þ
¼ X ðsÞ
Since x(t) ¼ x(t) then L{x(t)} ¼ L{x(t)}

Thus, X(s) ¼ X(s).
Odd Property
If x(t) is an odd function such that x(t) ¼ x(t), then X(s) ¼ X(s).
Proof ð1
X ðsÞ ¼ xðt Þ est dt ð4:52Þ
1
Consider
ð1
X 1 ðsÞ ¼ xðt Þ est dt ð4:53Þ
1
Let t1 ¼ t; then

ð1 ð1
X 1 ðsÞ ¼ xðt 1 Þ e dt 1 ¼
st 1
xðt 1 Þ est1 dt 1
1 1 ð4:54Þ
¼ X ðsÞ
Since x(t) ¼ x(t) then L{x(t)} ¼ L{x(t)}

Thus, X(s) ¼ X(s).
4.4.2 Differentiation Property of the Unilateral Laplace

Transform
Most of the properties of the bilateral transform tabulated in Table 4.1 are the same
for the unilateral transform. In particular, the differential property of unilateral
Laplace transform is different as it requires that x(t) ¼ 0 for t < 0 and contains no
impulses and higher-order singularities.
If x(t) is a signal with unilateral Laplace transform X (s), then the unilateral
Laplace transform of dxdt can be found by using integrating by parts as
Table 4.1 Some properties of the Laplace transform

Property Signal Laplace transform ROC
Linearity a1x1(t) + a2x2(t) a1X1(s) + a2X2(s) At least R1 \ R2
Time shifting x(t – t0) est0 X ðsÞ Same as R
Shifting in the s-domain es0 t xðt Þ X(s – s0) Shifted version of R
1
s R
Time scaling x(at) aX a a
Differentiation in the time domain dx sX(s) At least R
dt
Differentiation in the s-domain tx(t) dX ðsÞ R
ds
ðt
Integration s X ðsÞ
1 R \ ℜe(s) > 0.
xðt Þdt
1
Convolution x1(t) * x2(t) X1(s)X2(s) R1 \ R2
ð1 ð1
dx st
e dt ¼ xðt Þest 10þ þ s xðt Þest dt
0 dt 0 ð4:55Þ
þ
¼ sX ðsÞ xð0 Þ
2
Applying this second time yields the unilateral Laplace transform of ddt2x as given
by
ð1
d 2 x st
e dt ¼ s2 X ðsÞ sxð0þ Þ x_ ð0þ Þ ð4:56Þ
0 dt 2
where x_ ð0þ Þ is the dx
dt evaluated at t ¼ 0 .
+
Similarly, applying this for third time yields the unilateral Laplace transform
ð1
d 3 x st
e dt ¼ s3 XðsÞ s2 xð0þ Þ s_x ð0þ Þ €xð0þ Þ ð4:57Þ
0 dt 3
where €xð0þ Þ is the ddt2x evaluated at t ¼ 0+.

2
n
Continuing this procedure for the nth time, the unilateral Laplace transform of ddtnx
is given by
ð1
d n x st
e dt ¼ sn X ðsÞ sn1 xð0þ Þ sn2 x_ ð0þ Þ sn3€xð0þ Þ: . . . ð4:58Þ
0 dt n
Example 4.1 Determine whether the following Laplace transforms correspond to

the even time function or odd time function and comment on the ROCs.
(a) X ðsÞ ¼ ðsþ2Ks
Þðs2Þ
(b) X ðsÞ ¼ Kððsþ2
sþj2Þðsj2Þ
Þðs2Þ
(c) Comment on the ROCs
Solution
(a) X ðsÞ ¼ ðsþ2Ks
Þðs2Þ ; X ðsÞ ¼ ðsþ2Þðs2Þ ¼ X ðsÞ;
Ks
Hence, the corresponding x(t) is an odd function.

(b) X ðsÞ ¼ Kððsþ2
sþj2Þðsj2Þ
Þðs2Þ ¼ X ðsÞ
Hence, the corresponding x(t) is an even function.
(c) The ROCs for (a) and (b) are shown in Figure 4.5(a) and (b), respectively. From
Figure 4.5(a) and (b), it can be stated that for the time function to be even or odd,
the ROC must be two sided.
Example 4.2 A real and even signal x(t) with its Laplace transform X(s) has four
poles with one pole located at 12 ejπ=4 , with no zeros in the finite s-plane and X
(0) ¼ 16. Find X(s).
Solution Since X(s) has four poles with no zeros in the finite s-plane, it is of the
form
K
X ðsÞ ¼
ð s p1 Þ ð s p2 Þ ð s p3 Þ ð s p 4 Þ
As x(t) is real, the poles of X(s) must occur as conjugate reciprocal pairs.
Hence, p2 ¼ p1 ∗ ; p4 ¼ p3 ∗ and X(s) becomes
K
X ðsÞ ¼
ð s p1 Þ ð s p1 ∗ Þ ð s p3 Þ ð s p 3 ∗ Þ
(a) (b)
Im Im
j2
s plane s plane
Re Re
-2 2
-2 2
-j2
K ðsþj2Þðsj2Þ
Figure 4.5 (a) ROC of X ðsÞ ¼ ðsþ2Ks
Þðs1Þ. (b) ROC of X ðsÞ ¼ ðsþ2Þðs2Þ
Since x(t) is even, the X(s) also must be even, and hence the poles must be
symmetric about the jΩ axis. Therefore, p3 ¼ p1 ∗ .
Thus,
K
XðsÞ ¼
ðs p1 Þðs p∗
1 Þðs þ p∗
1 Þðs þ p1 Þ
Assuming that the given pole location is that of p1, that is, p1 ¼ 12 ejπ=4 ,
We obtain
K
XðsÞ ¼
1 jπ=4 1 jπ=4 1 1
s e s e s þ ejπ=4 s þ ejπ=4
2 2 2 2

1 jπ=4 1 π π 1 1 1 1 1
e ¼ cos þ jsin ¼ pffiffiffi þ jpffiffiffi ¼ pffiffiffi þ j pffiffiffi
2 2 4 4 2 2 2 2 2 2 2
K
XðsÞ ¼
1 1 1 1
S
2
p ffiffi
ffi sþ S þ
2
p ffiffi
ffi sþ
2 4 2 4
when s ¼ 0, X ð0Þ ¼ 1=16

K
¼ 16, and therefore K ¼ 1.
Hence,
1
X ðsÞ ¼
s2 p1ffiffi2s þ 14 s2 þ p1ffiffi2s þ 14
4.4.3 Initial Value Theorem
For a signal x(t) with Laplace transform X(s) and x(t) ¼ 0 for t < 0, then
xð0þ Þ ¼ lims!1 sX ðsÞ ð4:59Þ
Proof To prove the theorem, let the following integral first be evaluated by using
integration by parts
ð1
Ð 1 dx st
0þ e dt ¼ xðt Þ est 10þ þ xðt Þs est dt
dt 0þ ð4:60Þ
¼ xð0þ Þ þ sX ðsÞ
As s tends to 1, the Eq. (4.60) can be expressed as

ð1
dx st
lim e dt ¼ lim ½sXðsÞ xð0þ Þ ð4:61Þ
s!1 0 þ dt s!1
4.5 Laplace Transforms of Elementary Functions 187
As the integration is independent of s, the calculation of the limit and

the integration can be permuted provided that the integral converges uniformly. If
L{x(t)} exists, then
dx st
lim e ¼0 ð4:62Þ
s!1 dt
is valid. Hence, we get
xð0þ Þ ¼ lims!1 sX ðsÞ ð4:63Þ
4.4.4 Final Value Theorem
For a signal x(t) with Laplace transform X(s) and x(t) ¼ 0 for t < 0, then
xð1Þ ¼ lims!0 sX ðsÞ ð4:64Þ
Proof To prove this, the following integration is to be evaluated

ð1
dx st
lim e dt ¼ sX ðsÞ xð0þ Þ ð4:65Þ
s!0 0 þ dt
Again one can permute the sequence of determining the limit and the integration
provided the integral converges. The result is
ð1
dx
dt ¼ lim ½sX ðsÞ xð0þ Þ, ð4:66Þ
0þ dt s!0
and after integration it follows that
xð1Þ xð0þ Þ ¼ lim ½sX ðsÞ xð0þ Þ

s!0
ð4:67Þ
¼ lim ½sX ðsÞ xð0þ Þ
s!0
Therefore,
xð1Þ ¼ lim sX ðsÞ ð4:68Þ

s!0
4.5 Laplace Transforms of Elementary Functions
Unit Impulse Function The unit impulse function is defined by

1 for t ¼ 0
δðt Þ ¼ ð4:69Þ
0 elsewhere
By definition, the Laplace transform of δ (t) can be written as

ð1
X ðsÞ ¼ Lfδðt Þg ¼ δðt Þest dt
0 ð4:70Þ
¼1
The ROC is the entire s-plane.

Unit Step Function The unit step function is defined by

1 for t 0
uð t Þ ¼ ð4:71Þ
0 elsewhere
The Laplace transform of u(t) by definition can be written as

ð1
X ðsÞ ¼ Lfuðt Þg ¼ uðt Þ est dt
ð01
1
¼ 1est dt ¼ est 1
0
ð4:72Þ
0 s
1
¼
s
Hence, the ROC for X(s) is ℜe(s) > 0.
Example 4.3 Find the Laplace transform of x(t) ¼ δ(t – t0).
Solution By using the time shifting property, we get
Lfδðt t 0 Þg ¼ est0 Lfδðt Þg ¼ est0
The ROC is the entire s-plane.

Example 4.4 Find the Laplace transforms of the following:
(i) x(t) ¼ eαtu(t) (ii) x(t) ¼ eαtu(t) (iii) x(t) ¼ eαtu(t)
(iv) x(t) ¼ eαtu(t) ð1
Solution (i) X ðsÞ ¼ Lfeαt uðt Þg ¼ eαt uðt Þest dt
0
Because u(t) ¼ 1 for t < 0 and u(t) ¼ 0 for t > 0,

ð 0
X ðsÞ ¼ eðsþαÞt dt
1
1
¼
sþα
The ROC for X(s) is ℜe(s) < α

ð1
αt
(ii) X ðsÞ ¼ Lfe uðt Þg ¼ eαt uðt Þest dt
0
Because u(t) ¼ 1 for t < 0 and u(t) ¼ 0 for t > 0,

ð 0
X ðsÞ ¼ eðsαÞt dt
1
1
¼
sα
The ROC for X (s) is ℜe(s) < α
(iii) Let x1(t) ¼ u(t), then
1
X 1 ðsÞ ¼ Lfuðt Þg ¼
s
By using the shifting in the s-domain property, we get
1
X ðsÞ ¼ Lfeαt uðt Þg ¼ X 1 ðs þ αÞ ¼
sþα
The ROC for X(s) is ℜe(s) > α
(iv) Let x1(t) ¼ u(t), then
1
s
1
X ðsÞ ¼ Lfeαt uðt Þg ¼ X 1 ðs αÞ ¼
sα
The ROC for X(s) is ℜe(s) > α.
Example 4.5 Find the Laplace transform of
t ðn1Þ
xð t Þ ¼ uð t Þ
ðn 1Þ!
Solution Let x1(t) ¼ u(t), then
1
s
and for n ¼ 2, x2(t) ¼ tu(t).
By using the differentiation in the s-Domain property, we get
dX 1 1
X 2 ðsÞ ¼ Lftuðt Þg ¼ ¼ 2
ds s
2
Similarly, for n ¼ 3, x3 ðt Þ ¼ ð31
t
Þ! uðt Þ.
Again, by using the differentiation in the s-domain property, we get

t2 dX 2 1
X 3 ðsÞ ¼ L uð t Þ ¼ ¼ 3
ðn 1Þ! ds s
n ðn1Þ o 1
In general, X ðsÞ ¼ L ðtn1Þ!uðt Þ ¼ n .
s
Example 4.6 Find the Laplace transform of
t ðn1Þ αt
xð t Þ ¼ e uð t Þ
ðn 1Þ!
ðn1Þ
Solution Let x1 ðt Þ ¼ ðtn1Þ! uðt Þ, then

t ðn1Þ 1
X 1 ðsÞ ¼ L uð t Þ ¼
ðn 1Þ! Sn

t ðn1Þ αt 1
X ðsÞ ¼ L e uðt Þ ¼ X 1 ðs þ αÞ ¼ for ℜeðsÞ > α
ðn 1Þ! ðs þ αÞn
Example 4.7 Find the Laplace transform of x(t) ¼ sin ωt u(t)

Solution
ejωt ejωt
Lf sin ωt uðt Þ ¼ L uð t Þ
2j
1 jωt
¼ L e uðt Þ L ejωt uðt Þ
2j
Using the shifting in the s-domain property, we get

1 jωt jωt 1 1 1
L e uð t Þ L e uð t Þ ¼
2j 2j s jω s þ jω
ω
¼ 2 for ℜeðsÞ > 0
s þ ω2
Therefore,
ω
Lf sin ωt uðt Þ ¼ for ℜeðsÞ > 0
s2 þ ω2
Example 4.8 Find the Laplace transform of x(t) ¼ cos ωt u(t).

Solution
ejωt þ ejωt
Lf cos ωt uðt Þ ¼ L uð t Þ
2
1
¼ L ejωt uðt Þ þL ejωt uðt Þ
2
Using the shifting in the s-domain property, we get

1 jωt 1 1 1
L e uðt Þ þ L ejωt uðt Þ ¼ þ
2 2 s jω s þ jω
s
¼ 2 for ℜeðsÞ > 0
s þ ω2
Therefore,
s
Lf cos ωt uðt Þ ¼ for ℜeðsÞ > 0
s 2 þ ω2
Example 4.9 Find the Laplace transform of x(t) ¼ eαtsin ωt u(t).

Solution Let x1(t) ¼ sin ωt u(t), then
ω
X 1 ðsÞ ¼ Lf sin ωt uðt Þg ¼
s 2 þ ω2
ω
X ðsÞ ¼ Lfeαt sin ωt uðt Þg ¼ X 1 ðs þ αÞ ¼ for ℜeðsÞ > α
ðs þ αÞ2 þ ω2
Example 4.10 Find the Laplace transform of x(t) ¼ eαtcos ωt u(t).

Solution Let x1(t) ¼ cos ωt u(t), then
s
X 1 ðsÞ ¼ Lf cos ωt uðt Þg ¼
s2 þ ω2
sþα
X ðsÞ ¼ Lfeαt cos ωt uðt Þg ¼ X 1 ðs þ αÞ ¼ for ℜeðsÞ > α
ðs þ αÞ2 þ ω2
sin t
Example 4.11 Find the Laplace transform of t
Solution
1
Lf sin t g ¼
s2 þ 1
ð1
sin t 1
L ¼ 2þ1
du
t s u
¼ tan 1 u1s
π
¼ tan 1 s
2
4t
e3t
Example 4.12 Find the Laplace transform of e t
Solution
1 1
L e4t ¼ ; L e3t ¼
s4 sþ3
4t ð1 ð1
e e 3t
1 1
L ¼ du du
t s u4 s uþ3
1
¼ ln ðu 4Þ1
s ln ðu þ 3Þ s
ðs 4Þ
¼ ln
ðs þ 3Þ
ð s þ 3Þ
¼ ln
ð s 4Þ
Example 4.13 Consider the signal x(t) ¼ etu(t) + 2e2tu(t).

(a) Does the Fourier transform of this signal converge?
(b) For which of the following values of a does the Fourier transform of x(t) eαt
converge?
(i) α ¼ 1
(ii) α ¼ 2.5
(c) Determine the Laplace transform X(s) of x(t).
Sketch the location of the poles and zeros of X(s) and the ROC.
Solution
(a) The Fourier transform of the signal does not converge as x(t) is not absolutely
integrable due to the rising exponentials.
(b) (i) For α ¼ 1, x(t) eαt ¼ u(t) þ 2etu(t).
Although the growth rate has been slowed, the Fourier transform still does not
converge.
(ii) For α ¼ 2.5, x(t) eαt ¼ e1.5tu(t) + 2e0.5tu(t), the Fourier transform
converges
(c) The Laplace transform of x(t) is

1 2 2s 3 2 s 32
X ðsÞ ¼ þ ¼ ¼
s 1 s 2 ðs 1Þðs 2Þ ðs 1Þðs 2Þ
and its pole-zero plot and ROC are as shown in Figure 4.6.
Figure 4.6 Pole-zero plot

and ROC
Im
Re
1 1.5 2
It is noted that if α > 2, s ¼ α þ jΩ is in the region of convergence, as it is shown

in part (b) (ii), the Fourier transform converges.
Example 4.14 The Laplace transform H(s) of the impulse response h(t) for an LTI
system is given by
1
H ðsÞ ¼ ℜeðsÞ > 2
ðs þ 2Þ
Determine the system output y(t) for all t if the input x(t) is given by x(t) ¼ e3t/2
+ 2et for all t.
Solution From the convolution integral,
ð1
1
Let x(t) ¼ eαt, then

ð1 ð1
αðtτÞ αt
yð t Þ ¼ hðτÞe dτ ¼ e hðτÞeατ dτ
1 1
ð1
hðτÞ eατ dτ can be recognized as H(s)|s ¼ α.
1
Hence, if x(t) ¼ eαt, then
yðt Þ ¼ eαt ½H ðsÞjs¼α
Using linearity and superposition, it can be recognized that if x(t) ¼ e3t/2 þ 2et,
then

yðt Þ ¼ e3t=2 H ðsÞs¼3=2 þ 2et H ðsÞjs¼1
So that
yðt Þ ¼ 2e3t=2 þ 2et for all t

Example 4.15 The output y(t) of a LTI system is

yðt Þ ¼ 2 3et þ e3t uðt Þ
for an input

xðt Þ ¼ 2 þ 4e3t uðt Þ
Determine the corresponding input for an output
y1 ðt Þ ¼ ð1 et tet Þuðt Þ
Solution For the input

x(t) ¼ (2 þ 4e3t)u(t), the Laplace transform is
2 4 6ð s þ 1Þ
X ðsÞ ¼ þ ¼
s s þ 3 s ð s þ 3Þ
The corresponding output has the Laplace transform
2 3 1 6
Y ðsÞ ¼ þ ¼
s s þ 1 s þ 3 sðs þ 1Þðs þ 3Þ
Y ðsÞ 1
Hence, H ðsÞ ¼ ¼ ℜeðsÞ > 0
X ðsÞ ðs þ 1Þ2
Now, the output
y1(t) ¼ (1 – et– tet)u(t) has the Laplace transform
1 1 1 1
Y 1 ðsÞ ¼ þ ¼ ℜeðsÞ > 0
s s þ 1 ð s þ 1Þ 2 s ð s þ 1Þ 2
Hence, the Laplace transform of the corresponding input is
Y 1 ðsÞ 1
X 1 ðsÞ ¼ ¼ ℜeðsÞ > 0
H ðsÞ s
The inverse Laplace transform of X1 (s) gives
x1 ðt Þ ¼ uðt Þ:
4.6 Computation of Inverse Laplace Transform Using

Partial Fraction Expansion
The inverse Laplace transform of a rational function X(s) can be easily computed by
using the partial fraction expansion.
4.6 Computation of Inverse Laplace Transform Using Partial Fraction Expansion 195
4.6.1 Partial Fraction Expansion of X(s) with Simple Poles
Consider a rational Laplace transform X(s) of the form
N ðsÞ
X ðsÞ ¼ ð4:73Þ
ðs p1 Þðs p2 Þ: . . . . . . ðs pn Þ
with the order of N(s) is less than the order of the denominator polynomial.
The poles p1, p2, . . . .., pn are distinct.
The rational Laplace transform X(s) can be expanded using partial fraction
expansion as
k1 k2 kn
X ðsÞ ¼ þ þ ð4:74Þ
ðs p1 Þ ðs p2 Þ ð s pn Þ
The coefficients k1, k2, . . . ., kn are called the residues of the partial fraction
expansion. The residues are computed as

ki ¼ ðs-pi Þ XðsÞs¼pi i ¼ 1, 2, . . . , n ð4:75Þ
With the known values of the coefficients k1, k1, . . . ., kn inverse transform of
each term can be determined depending on the location of each pole relative to the
ROC.
4.6.2 Partial Fraction Expansion of X(s) with Multiple Poles
Consider a rational Laplace transform X(s) with repeated poles of the form
N ðsÞ
X ðsÞ ¼ ð4:76Þ
ðs p1 Þr ðs p2 Þ: . . . . . . ðs pn Þ
with multiplicity r poles at s ¼ p1.

The X(s) with multiple poles can be expanded as
k11 k12 k1r k2 kn

YðsÞ ¼ þ þ þ þ ð4:77Þ
ðs p1 Þ ðs p1 Þ2 ð s p1 Þ r ð s p2 Þ ðs pn Þ
The coefficients k2, . . . ., kn can be computed using the residue formula used in
Section 4.6.1. The residues k11, k12, . . ., klr are computed as

k1r ¼ ðs p1 Þr XðsÞs¼pi ð4:78Þ
1 d
k1ðr1Þ ¼ ½ðs p1 Þr XðsÞs¼pi ð4:79Þ
1! ds
1 d2
k1ðr2Þ ¼ 2
½ðs p1 Þr XðsÞs¼pi ð4:80Þ
2! ds
and so on.
Example 4.16 Find the time function x(t) for each of the following Laplace trans-
form X(s)
sþ2
(a) X ðsÞ ¼ ℜeðsÞ > 3
s2 þ 7s þ 12
s þsþ1
2
(b) X ðsÞ ¼ 2 0 < ℜeðsÞ < 1
s ð s 1Þ
s2 s þ 1
(c) X ðsÞ ¼ 1 < ℜeðsÞ
ð s þ 1Þ 2
sþ1
(d) X ðsÞ ¼ 2 ℜeðsÞ < 3
s þ 5s þ 6
Solution (a) X ðsÞ ¼ s2 þ7sþ12
sþ2
Using partial fraction expansion,
sþ2 k1 k2
¼ þ
s2 þ 7s þ 12 s þ 3 s þ 4
k1 þ k2 ¼ 1
4k1 þ 3k2 ¼ 2
Solving for k1 and k2, k1 ¼ 1; k2¼2.

Thus, X ðsÞ ¼ s2 þ7sþ12
sþ2
¼ sþ3
1
þ sþ4
2
Using Table 4.2, we obtain

x(t) ¼ e3tu(t) þ 2e4tu(t)
s2 s þ 1 1 1 1
(b) X ðsÞ ¼ ¼ þ
s 2 ð s 1Þ s 1 sðs 1Þ s2 ðs 1Þ
1 k1 k2
¼ þ
sðs 1Þ s s1
Solving for k1 and k2, k1 ¼ 1; k2 ¼ 1.
1 1 1
¼ þ
s ð s 1Þ s s1
1 k1 k11 k 12
¼ þ þ 2
s 2 ð s 1Þ s 1 s s
Table 4.2 Elementary Signal Laplace transform ROC

functions and their Laplace
δ(t) 1 All s
transforms
u(t) 1
s
ℛe(s) > 0
u(t) 1
s
ℛe(s) > 0
δ(t t0) est0 All s
eαtu(t) 1
sþα
ℛe(s) > α
eatu(t) 1
sþα
ℛe(s) < α
t ðn1Þ 1 ℛe(s) > 0
uð t Þ sn
ðn 1Þ!
t ðn1Þ αt 1
ðsþαÞn
ℛe(s) > α
e uð t Þ
ðn 1Þ!
ω
sin ωt u(t) s2 þω2 ℜe(s) > 0
cos ωt u(t) s
s2 þω2 ℜe(s) > 0
αt ω
e sin ωt u(t) ðsþαÞ2 þω2
ℜe(s) > α
eαt cos ωt u(t) sþα
ðsþαÞ2 þω2
ℜe(s) > α
Solving for k1, k11, and k12
k1 ¼ 1; k11 ¼ 1; k12 ¼ 1:

1 1 1 1
¼ 2
s2 ðs 1Þ s 1 s s
Thus,
s2 s þ 1
X ðsÞ ¼
s 2 ð s 1Þ
1 1 1 1 1 1 1 1 1
¼ þ ¼ þ þ
s 1 sðs 1Þ s2 ðs 1Þ s 1 s s 1 s 1 s s2
1 1
¼
s 1 s2
xðt Þ ¼ et uðt Þ tuðt Þ
sþ1
2
(c) X ðsÞ ¼ sðsþ1Þ2
¼ 1 ðsþ1
3s
Þ2
3s k11 k 12
¼ þ
ð s þ 1Þ 2 s þ 1 ð s þ 1Þ 2
Solving for k11 and k12, we get
k11 ¼ 3; k12 ¼ 3
Hence,
s2 s þ 1 3 3
X ðsÞ ¼ ¼1 þ
ð s þ 1Þ 2 s þ 1 ð s þ 1Þ 2
xðt Þ ¼ δðt Þ 3et uðt Þ þ 3tet uðt Þ
(d) X ðsÞ ¼ s2 þ5sþ6

sþ1
sþ1 k1 k2
¼ þ
s2 þ 5s þ 6 s þ 3 s þ 2
k1 þ k2 ¼ 1
2k1 þ 3k 2 ¼ 1
Solving for k1 and k2, k1 ¼ 2; k2 ¼ 1.

Thus, X ðsÞ ¼ s2 þ5sþ6
sþ1
¼ sþ3
2
sþ2
1
xðt Þ ¼ 2e3t uðt Þ þ e2t uðt Þ
Example 4.17 Find the inverse Laplace transform of the following
s2 þ s þ 1
X ðsÞ ¼
s2 s þ 1
Solution
2s
X ðsÞ ¼ 1 þ
s2 s þ 1
2s
¼1þ pffiffi2
2
s 12 þ 23
1
s 1
¼ 1 þ 2 2
pffiffi2 þ pffiffi2
2 2
s 12 þ 23 s 12 þ 23

pffiffiffi pffiffiffi
3 2 3
xðt Þ ¼ δðt Þ þ 2et=2 cos t uðt Þ þ pffiffiffi et=2 sin t uð t Þ
2 3 2
Example 4.18 Determine x(t) for the following conditions if X(s) is given by
1
X ðsÞ ¼
ð s þ 2Þ ð s þ 3Þ
(a) x(t) is right sided

(b) x(t) is left sided
(c) x(t) is both sided
Solution Using partial fraction, X(s) can be written as
(a)
1 1 1
X ðsÞ ¼ ¼
ð s þ 2Þ ð s þ 3Þ ð s þ 2Þ ð s þ 3Þ
If x(t) is right sided,

1 1 1 1
xðt Þ ¼ L L ¼ e2t uðt Þ e3t uðt Þ
The ROC is to the right of the rightmost pole as shown in Figure 4.7(a).
(b) If x(t) is left sided,

1 1 1 1
xð t Þ ¼ L L ¼ e2t uðt Þ e3t uðt Þ
The ROC is to the left of the leftmost pole as shown in Figure 4.7(b)
(a) (b) Im Im
(c)
s plane s plane
s plane
Re Re 0 Re
0 -3 -2
-3 -2 -3 -2
Figure 4.7 (a) ROC of right-sided x(t) (b) ROC of left-sided x(t) (c) ROC of both-sided x(t)
(c) If x(t) is two sided,

( e2t uðt Þ for right sided
1 1
L ¼
ð s þ 2Þ e2t uðt Þ for left sided
(
1 e3t uðt Þ for right sided
L1 ¼
ð s þ 3Þ e3t uðt Þ for left sided
Hence, if x(t) is chosen as,
xðt Þ ¼ e2t uðt Þ e3t uðt Þ
The ROC is as shown in Figure 4.7(c)

Example 4.19 Determine x(t) first for the following and verify the initial and final
value theorems.
(i) XðsÞ ¼ sþ4
1
(ii) XðsÞ ¼ ðsþ3sþ5

Þðsþ4Þ
n o
Solution (i) xðtÞ ¼ L1 1
sþ4 ¼ e4t uðt Þ
xð0þ Þ ¼ lim e4t ¼ 1:

t!0
þ s 1 1 1
xð0 Þ ¼ lim sX ðsÞ ¼ lim ¼ lim ¼ ¼ ¼ 1:
s!1 s!1 s þ 4 s!1 4 4 1þ0
1þ 1þ
s 1
4t
xð1Þ ¼ lim e ¼ 0:
t!1
s 1 1 1
xð1Þ ¼ lim sX ðsÞ ¼ lim ¼ lim ¼ ¼ ¼ 0:
s!0 s!0 s þ 4 s!0 4 4 1þ1
1þ 1þ
0 0
n o n o n o
(ii) xðtÞ ¼ L1 sþ5
ðsþ3Þðsþ4Þ ¼ L1 2
sþ3 L1 1
sþ4 ¼ 2e3t uðt Þ e4t uðt Þ
4.7 Inverse Laplace Transform by Partial Fraction Expansion Using MATLAB 201

xð0þ Þ ¼ lim 2e3t e4t ¼ 1::
t!0

þ sþ5 2s s
xð0 Þ ¼ lim sX ðsÞ ¼ lim ¼ lim lim
s!1 s!1 ðs þ 3Þðs þ 4Þ s!1 s þ 3 s!1 s þ 4
2 1
¼ lim lim ¼ 2 1 ¼ 1:
s!1 3 s!1 4
1þ 1þ
s s

xð1Þ ¼ lim 2e3t e4t ¼ 0:
t!1

sþ5 2s s
xð1Þ ¼ lim sX ðsÞ ¼ lim ¼ lim lim
s!0 s!0 ð s þ 3Þ ð s þ 4Þ s!0 s þ 3 s!0 s þ 4
2 1
¼ lim lim ¼ 0 0 ¼ 0:
s!0 3 s!0 4
1þ 1þ
s s
4.7 Inverse Laplace Transform by Partial Fraction

Expansion Using MATLAB
The MATLAB command residue can be used to find the inverse transform using the
power series expansion. To find the partial fraction decomposition of Y(s), we must
first enter the numerator polynomial coefficients and the denominator polynomial
coefficients as vectors.
The following MATLAB statement determines the residue (r), poles (p), and
direct terms (k) of the partial fraction expansion of H(s).
½k; p; const ¼ residue ðN; DÞ;
where N is the vector of the numerator polynomial coefficients in decreasing order

and the vector D contains the denominator polynomial coefficients in decreasing
order.
Example 4.20 Find the inverse Laplace transform of the following using MATLAB
sþ1
X ðsÞ ¼
s2 þ 5s þ 6
Solution The following MATLAB statements are used to find the Laplace trans-
form of given X(s)
N ¼ ½ 1 1 ; % coefficients of the numerator polynomial in decreasing order
D ¼ ½ 1 5 6 ; % coefficients of the denominator polynomial in decreasing
order
[k, p, const] ¼ residue (N and D); % computes residues, poles, and constants
Execution of the above statements gives the following output
k¼
2.0000
1.0000
p¼
3.0000
2.0000
Thus, the partial fraction decomposition of X(s) is
2 1
X ðsÞ ¼
sþ3 sþ2
After getting the partial fraction expansion of X(s), the following MATLAB
statements are used to obtain x(t), that is, the inverse Laplace transform of X(s).
syms s t
X=2/(s+3)-1/(s+2);
ilaplace(X)
Execution of the above three MATLAB statements gives
xðt Þ ¼ 2e3t e2t
4.8 Analysis of Continuous-Time LTI Systems Using

the Laplace Transform
4.8.1 Transfer Function
It was stated in Chapter 2 that a continuous time LTI system can be completely
characterized by its impulse response h(t). The output signal y(t) of a LTI system and
the input signal x(t) are related by convolution as
y ð t Þ ¼ hð t Þ ∗ x ð t Þ ð4:81Þ
By using the convolution property, we get
Y ðsÞ ¼ H ðsÞX ðsÞ ð4:82Þ
indicating the Laplace transform of the output signal y(t) is the product of the
Laplace transforms of the impulse response h(t) and the input signal x(t). The
transform H(s) is called the transfer function or the system function and expressed as
4.8 Analysis of Continuous-Time LTI Systems Using the Laplace Transform 203
Figure 4.8 Sallen-Key low-pass filter circuit
Y ðsÞ
H ðsÞ ¼ ð4:83Þ
X ðsÞ
The roots of the denominator of the transfer function are called poles. The roots of
the numerator are called zeros. The places where the transfer function is infinite (the
poles) determine the region of convergence.
Example 4.21 Obtain the system function of the following Sallen-Key low-pass
filter circuit.
Solution For the circuit shown in Figure 4.8, the following relations can be
established:
V i V 1 ¼ r 1 I r1 ; V 1 V 2 ¼ r 2 I r2 ;
I c2 ¼ sc2 V 2 ; I c1 ¼ sc1 ðV 1 V 0 Þ;
I r 1 ¼ I r 2 þ I c1 ; I r 2 ¼ I c2 ; V 2 ¼ V 0
Rewriting the current node equations, we get
Vi V1 V1 V0
¼ þ sc1 ðV 1 V 0 Þ
r1 r2
Which can be rewritten as
V i r 2 ¼ ðr 1 þ r 2 þ r 1 r 2 sc1 ÞV 1 ðr 1 þ r 1 r 2 sc1 ÞV 0
V1 V0
¼ sc2 V 0 ;
r2
V 1 ¼ ð1 þ r 2 sc2 ÞV 0
Substituting the above Eq. for V 1 , the input-output relation can be written as
V i r 2 ¼ ½ðr 1 þ r 2 þ r 1 r 2 sc1 Þð1 þ r 2 sc2 Þ r 1 r 1 r 2 sc1 V 0

r1 r1
Vi ¼ 1 þ þ r 1 sc1 ð1 þ r 2 sc2 Þ r 1 sc1 V 0
r2 r2
Thus,
V 0 ðsÞ 1
¼
V i ðsÞ r1 r1
1 þ þ r 1 sc1 ð1 þ r 2 sc2 Þ r 1 sc1
r2 r2
1
¼
1 þ c2 ðr 1 þ r 2 Þs þ s2 c2 c1 r 2 r 1
Hence, the system function is given by
1
r 1 r 2 c1 c2
H ðsÞ ¼
r 1 þ r2 1
s2 þ sþ
r 1 r 2 c1 r 1 r 2 c1 c2
4.8.2 Stability and Causality
Stabile LTI System

A continuous-time LTI system is stable if and only if the impulse response is
absolutely integrable, that is,
ð1
j hðt Þ j dt < 1: ð4:84Þ
1
The Laplace transform of the impulse response is known as the system function,
which can be written as
ð1
H ðsÞ ¼ hðt Þest dt ð4:85Þ
1
A continuous-time LTI system is stable if and only if the transfer function has
ROC that includes the imaginary axis (the line in complex where the real part
is zero).
Causal LTI System
A continuous time LTI system is causal if its output y(t) depends only on the current
and past input x(t) but not the future input. Hence, h(t) ¼ 0 for t < 0.
For causal system, the system function can be written as
ð1
HðsÞ ¼ hðtÞest dt ð4:86Þ
0
Im
s plane
Re
Figure 4.9 ROC of a causal LTI system
If s ¼ σ + jΩ and h(t)eσt being absolutely integrable for convergence of

H(s) leads to the following convergence condition,
ð1
j hðt Þeσt j dt < 1 ð4:87Þ
0
Any large value of σ satisfies the above equation. Thus, the ROC is the region to
the right of a vertical line that passes through the point ℜe(s) ¼ σ as shown in
Figure 4.9.
In particular, if H(s) is rational, H ðsÞ ¼ NDððssÞÞ, then the system is causal if and only if
its ROC is the right-sided half plane to the right of the rightmost pole and the order of
numerator N(s) is no greater than that of the denominator D(s), so that the ROC is a
right-sided plane without any poles (even at s ! 1).
Stable and Causal LTI System
As the ROC of a causal system is to the right of the rightmost pole and for a stable
system, the rightmost pole should be in the left half of the s-plane and should include
the jΩ axis; all the poles of a system should lie in the left half of the s-plane (the real
parts of all poles are negative, ℜe(sp) < 0 for all sp) for a system to be causal and
stable as shown in Figure 4.10.
Stable and Causal Inverse LTI System
For a causal stable system, the poles must lie in the left half of the s-plane. But it is
known that the poles of the inverse system are zeros of the original system.
Therefore, the zeros of the original system should be in the left half of the s-plane.
Im
s plane
Re
Figure 4.10 ROC of a stable and causal LTI system
Example 4.22 Consider a LTI system with the system function
s1
H ðsÞ ¼
s2 s 6
Show the pole-zero locations of the system and ROCs for the following:
(a) the system is causal
(b) the system is stable, noncausal
(c) the system is neither causal nor stable
Solution
(a) The system function can be rewritten as
s1
H ðsÞ ¼
ð s þ 2Þ ð s 3Þ
Since the ROC of a causal system is to the right of the rightmost pole, the
pole-zero plot and ROC of the system are shown in Figure 4.11.
(b) For a stable system, the ROC should include the imaginary axis. The pole-zero
plot and the ROC are shown in Figure 4.12.
(c) Since the system is neither causal nor stable, the ROC should not include the
imaginary axis and not to the right of the rightmost pole. Hence, the pole-zero
plot and the ROC are shown in Figure 4.13.
Im
s-plane
Re
-2 1 3
Figure 4.11 Pole-zero plot and ROC of the causal system
Im
s-plane
Re
-2 1 3
Figure 4.12 Pole-zero plot and ROC of the noncausal, stable system
4.8.3 LTI Systems Characterized by Linear Constant

Coefficient Differential Equations
The system function for a system characterized by a linear constant coefficient

equation can be obtained by exploiting the properties of the Laplace transforms.
The Laplace transform transforms a differential equation into an algebraic equation
in the s-domain making it easy to find the time-domain solution of the differential
equation.
s-plane Im
Re
-2 1 3
Figure 4.13 Pole-zero plot and ROC of the system neither causal nor stable
Consider a general linear constant coefficient differential equation of the form
dn y dðn1Þ y d2 y dy
an n þ a n1 ðn1 Þ
þ : þ a 2 2
þ a1 þ a0 y
dt dt dt dt
ð4:88Þ
dm x dðm1Þ x d2 x dx
¼ bm m þ bm1 ðm1Þ þ : þ b2 2 þ b1 þ b0 x
dt dt dt dt
Taking the Laplace transform of both sides of equation repeated use of the
differentiation property and linearity property, we obtain
ðan sn þ an1 sðn1Þ þ : þ a2 s2 þ a1 s þ a0 ÞYðsÞ

ð4:89Þ
¼ ðbm sm þ bm1 sðm1Þ þ : þ b2 s2 þ b1 s þ b0 ÞXðsÞ
Thus, the system function is given by
YðsÞ bm sm þ bm1 sðm1Þ þ : þ b2 s2 þ b1 s þ b0

HðsÞ ¼ ¼ ð4:90Þ
XðsÞ an sn þ an1 sðn1Þ þ : þ a2 s2 þ a1 s þ a0
The system function is rational for a system characterized by a differential

equation with the roots of the numerator polynomial as zeros and the roots of the
denominator polynomial as poles.
Eq. (4.90) does not specify any ROC since a differential equation by itself does
not constrain any region of convergence. However, with the additional knowledge of
stability and causality, the ROC can be specified.
Example 4.23 Consider a continuous LTI system described by the following dif-
2
ferential equation ddt2y dy
dt 6y ¼ x.
(a) Determine the system function

(b) Determine h(t) for each of the following:
(i) the system is stable, noncausal
(ii) the system is causal
(iii) the system is neither causal nor stable
Solution Taking the Laplace transform of both sides of the given differential
equation, we obtain
s2 YðsÞ sYðsÞ 6YðsÞ ¼ XðsÞ
The above relation can be rewritten as

s2 s 6 YðsÞ ¼ XðsÞ
Now, the system function is given by
Y ðsÞ 1
H ðsÞ ¼ ¼
X ðsÞ s2 s 6
The pole-zero plot for the system function is shown in Figure 4.14.
(b) The partial fraction expansion of H(s) yields
1 1
H ðsÞ ¼
5ð s 3Þ 5ð s þ 2Þ
Im
Re
-2 3
Figure 4.14 Pole-zero plot of the system function

(i) For H(s) to be stable, noncausal, the ROC is 2 < ℜe(s) < 3.
Hence, hðt Þ ¼ 15 e3t uðt Þ 15 e2t uðt Þ
(ii) For the system to be causal, the ROC is ℜe(s) > 3.
Therefore, hðt Þ ¼ 15 e3t uðt Þ 15 e2t uðt Þ.
(iii) For the system to be neither causal nor stable, the ROC is ℜe(s) < 2.
Hence, hðt Þ ¼ 15 e3t uðt Þ þ 15 e2t uðt Þ
4.8.4 Solution of linear Differential Equations Using Laplace

Transform
The stepwise procedure to solve a linear differential equation is as follows:

Step 1: Take Laplace transform both sides of the equation.
Step 2: Simplify the algebraic equation obtained for Y(s) in the s-domain.
Step 3: Find the inverse transform of Y(s) to obtain y(t), the solution of the
differential equation.
Example 4.24 Find the solution of the following differential equation using Laplace
transform:
d2 y dy
2
5 þ 6y ¼ 0 yð0Þ ¼ 2, y_ ð0Þ ¼ 1:
dt dt
Solution
Step 1: Laplace transform both sides of given differential equation is
s2 2s 1 5ðsYðsÞ 2Þ þ 6YðsÞ ¼ 0
Step 2: Simplifying the expression for Y(s),
ðs2 5s þ 6ÞYðsÞ 2s 1 þ 10 ¼ 0
2s 9
Y ðsÞ ¼
ðs2 5s þ 6Þ
2s 9
YðsÞ ¼
ð s 3Þ ð s 2Þ
Step 3: Expanding Y(S) using partial fraction expansion,
k1 k2
YðsÞ ¼ þ
ð s 3Þ ð s 2Þ
k1 ð s 2Þ þ k2 ð s 3Þ
¼
ð s 3Þ ð s 2Þ
Comparing the numerator polynomial with the numerator polynomial of Y(s) of

step 2, we get
k1 þ k2 ¼ 2
2k1 3k2 ¼ 9
Solving for k1 and k2,
k1 ¼ 3, k2 ¼ 5:
Thus,
3 5
YðsÞ ¼ þ
ð s 3Þ ð s 2Þ
Inverse transform of Y(s) gives the solution of the differential equation as
yðt Þ ¼ 3e3t þ 5e2t
Example 4.25 Find the solution of the following differential equation using Laplace
transform
dy
þ y ¼ 2tet yð0Þ ¼ 2
dt
Solution
Step 1: Laplace transform both sides of given differential equation is
2
sYðsÞ ð2Þ þ YðsÞ ¼
ð s þ 1Þ 2
Step 2: Simplifying the expression for Y(s),

2
ðs þ 1ÞYðsÞ þ 2 ¼
ð s þ 1Þ 2
2
ðs þ 1ÞYðsÞ ¼ 2
ð s þ 1Þ 2
2 2
Y ðsÞ ¼
ð s þ 1Þ ð s þ 1Þ 2 ð s þ 1Þ
2 2
Y ðsÞ ¼
ð s þ 1Þ 3 ðs þ 1 Þ
2s2 4s
¼
ð s þ 1Þ 3
Step 3: Expanding Y(S) using partial fraction expansion,
k11 k12 k13

YðsÞ ¼ þ þ
ð s þ 1Þ ð s þ 1Þ 2
ð s þ 1Þ 3
k11 ðs2 þ 2s þ 1Þ þ k12 ðs þ 1Þ þ k13
¼
ðs þ 1Þ3
Comparing the numerator polynomial with the numerator polynomial of Y(s) of

step 2, we get
k11 þ k12 þ k13 ¼ 0

2k11 þ k12 ¼ 4
k11 ¼ 2
Solving for k11, k12, and k13,

k11 ¼ 2, k12 ¼ 0, and k13 ¼ 2.
Thus,
2 2
YðsÞ ¼ þ
ð s þ 1Þ ð s þ 1Þ 3
Inverse transform of Y(s) gives the solution of the differential equation as
yðt Þ ¼ 2et þ t 2 et

Figure 4.15 Series RLC

circuit
Example 4.26 Consider the following RLC circuit with R ¼ 5 ohms, L ¼ 1h, and
C ¼ 0.25 f.
(a) Determine the differential equation relating V in and V c .
(b) Obtain V c ðt Þ using Laplace transform for V in ðt Þ ¼ et uðt Þ with V c ð0Þ ¼ 1, V_ c
ð0Þ ¼ 2 (Figure 4.15).
Solution (a) By applying Kirchhoff’s voltage law, we can arrive at the following
differential equation:
ð
di 1
L þ Ri þ idt ¼ vin
dt C
It is known that i ¼ C dVdtc , hence the above differential equation can be rewritten
as
d2 vc dvc
LC þ RC þ vc ¼ vin
dt2 dt
For given values of R, L, and C, the differential equation becomes
d2 vc dvc
þ5 þ 4vc ¼ 4vin
dt2 dt
2
(b) For given vin, ddtv2c þ 5dvdtc þ 4vc ¼ 4et uðt Þ
Laplace transform both sides of given differential equation is
4
s2 Vc ðsÞ s 2 þ 5ðsVc ðsÞ 1Þ þ 4Vc ðsÞ ¼
sþ1
Simplifying the expression for vc(s)
2 4
s Vc ðsÞ s 2 þ 5ðsVc ðsÞ 1Þ þ 4vc ðsÞ ¼
sþ1
4
ðs2 þ 5s þ 4ÞVc ðsÞ s 2 5 ¼
sþ1
4
ðs2 þ 5s þ 4ÞVc ðsÞ ¼ þsþ7
sþ1
s2 þ 8s þ 11
Vc ðsÞ ¼
ðs2 þ 5s þ 4Þ ðs þ 1Þ
s2 þ 8s þ 11
Vc ðsÞ ¼
ð s þ 1Þ 2 ð s þ 4Þ
Expanding vc(s) using partial fraction expansion
k11 k12 k2
Vc ðsÞ ¼ þ þ
ð s þ 1Þ ð s þ 1Þ 2 ð s þ 4Þ
Solving for k11, k12, and k2, we obtain

k11 ¼ 14
9 , k12 ¼ 3, k2 ¼ 9.
4 5
Thus,
14 4 5
Vc ðsÞ ¼ þ
9ð s þ 1Þ 3ð s þ 1Þ 2 9ð s þ 4Þ
Inverse transform of Vc(s) gives V c ðtÞ as
14 t 4 t 5 4t
vc ðtÞ ¼ e þ te e
9 3 9
Example 4.27 Consider the following parallel RLC circuit with R¼1 ohm, L¼1 h.
(a) Determine the differential equation relating Is and IL.
(b) Obtain zero-state response for IL(t) using Laplace transform for Is(t) ¼ e3tu(t).
(c) Obtain zero-input response for IL(t) using Laplace transform with IL(0) ¼ 1
(Figure 4.16).
Figure 4.16 Parallel RLC

circuit
Solution (a) The Is and IL are related by the following differential equation
dIL
þ IL ¼ IS
dt
(b) For the given Is,
dIL
þ IL ¼ e3t uðtÞ
dt
Applying the unilateral Laplace transform to the above differential equation, we
obtain
1
sIL ðsÞ IL ð0Þ þ IL ðsÞ ¼
sþ3
Since IL(0) ¼ 0 for zero state, ðs þ 1ÞIL ðsÞ ¼ sþ3
1
1
IL ðsÞ ¼
ðs þ 1Þðs þ 3Þ
By using partial fraction expansion, IL(s) can be expanded as
1 1
IL ðsÞ ¼
2ð s þ 1Þ 2ð s þ 3Þ
The inverse unilateral Laplace transform gives
1 1
IL ðtÞ ¼ et uðt Þ e3t uðt Þ
2 2
(c) For the zero-input response, Is(t) ¼ 0 and given that IL(0) ¼ 1, we have to find the
solution of the following differential equation for zero-input response
dIL
þ IL ¼ 0 IL ð0Þ ¼ 1
dt
The unilateral Laplace transform of the above differential equation is
sIL ðsÞ 1 þ IL ðsÞ ¼ 0
Thus,
1
IL ðsÞ ¼
sþ1
The inverse transform of IL(s) is zero-input response given by
IL ðtÞ ¼ et uðt Þ
4.8.5 Solution of Linear Differential Equations Using

Laplace Transform and MATLAB
Example 4.28 Find the solution of linear differential equation considered in Exam-
ple 4.24 using MATLAB
Solution The following MATLAB statements are used to find the solution of the
differential equation considered in Example 4.24
syms s t Y
Y1 = s*Y - 2;%Laplace transform of with y(0)=2
Y2 = s*Y1 - 1; %Laplace transform of with =1
Sol = solve(Y2 - 5*Y1 + 6*Y, Y);%Y(s)
y = ilaplace(Sol,s,t);%inverse Laplace transform of Y(s)
Execution of the above MATLAB statements gives the solution of the differential
equation as
yðt Þ ¼ 3e3t þ 5e2t
Example 4.29 Find the solution of linear differential equation considered in Exam-
Solution The following MATLAB statements are used to find the solution of the
differential equation considered in Example 4.25
syms s t Y
f = 2*t*exp(-t);%input signal
F = laplace(f,t,s);% finds Laplace transform of input
Y1 = s*Y + 2;% Laplace transform of with y(0)=-2
Sol = solve(Y1 + Y-F, Y);%Y(s)
y = ilaplace(Sol,s,t);% inverse of Y(s)
Execution of the above MATLAB statements gives the solution of the differential
equation as
yðt Þ ¼ 2et þ t 2 et

4.8.6 System Function for Interconnections of LTI Systems
Series Combination of Two LTI Systems

Impulse response of the series combination of two LTI systems is
hðt Þ ¼ h1 ðt Þ ∗ h2 ðt Þ ð4:91Þ
and from convolution property of the Laplace transform, the associated system
function is (Figure 4.17)
H ðsÞ ¼ H 1 ðsÞH 2 ðsÞ ð4:92Þ
Parallel Combination of Two LTI Systems

Impulse response of the parallel combination of two LTI systems is
hð t Þ ¼ h1 ð t Þ þ h2 ð t Þ ð4:93Þ
and from linearity property of the Laplace transform, the associated system function
is (Figure 4.18)
H ðsÞ ¼ H 1 ðsÞ þ H 2 ðsÞ ð4:94Þ
Figure 4.17 Series combination of two LTI systems
Figure 4.18 Parallel combination of two LTI systems

4.9 Block-Diagram Representation of System Functions

in the S-Domain
Consider the system function
YðsÞ bm sn þ bm1 sðn1Þ þ : þ b2 s2 þ b1 s þ b0

HðsÞ ¼ ¼ ð4:95Þ
X ðsÞ an sn þ an1 sðn1Þ þ : þ a2 s2 þ a1 s þ a0
Let us define the following basic elements for addition, multiplication, differen-
tiation, and integration in the s-domain (Figure 4.19)
The block-diagram representation of the above system function can be obtained
as the interconnection of these basic elements similar to the block-diagram repre-
sentation of differential equations in the time domain carried out in Section 2.6 of
Chapter 2. The block-diagram representation of the system function is shown in
Figure 4.20.
Example 4.30 Is the system represented by the following block-diagram stable?
(Figure 4.21)
Solution Let the signal at the bottom node of the block diagram be denoted by E(s).
Then we have the following relations
Figure 4.19 Block-

diagram representation basic
elements (a) adder,
(b) multiplier,
(c) differentiator,
(d) integrator
(a)
(b)
Differentiation
(c)
Integration
(d)
4.9 Block-Diagram Representation of System Functions in the S-Domain 219
1/an bm
X(s) Y(s)
1/s
-an-1 bm-1
1/s
-a1 b1
1/s
-a0 b0
Figure 4.20 Block-diagram representation of the system function
Figure 4.21 Block-diagram representation of a second-order system function
X ðsÞ 2sEðsÞ EðsÞ ¼ s2 EðsÞ

s2 E ðsÞ sE ðsÞ 6E ðsÞ ¼ Y ðsÞ
Eliminating the auxiliary signal E(s), we get
Y ðsÞ s2 s 6
H ðsÞ ¼ ¼ 2
X ðsÞ s þ 2s þ 1
From the system function H(s) found in the previous part, we see that the poles of
the system are at s ¼ 1. Since the system is given to be causal, and the rightmost
pole of the system is left of the imaginary axis, the system is stable.
4.10 Solution of State-Space Equations Using Laplace

Transform
For convenience, the state-space equations from Chapter 2 are repeated here:
X_ ðt Þ ¼ A X ðt Þ þ b℧ðt Þ ð4:96Þ
yðt Þ ¼ cX ðt Þ ð4:97Þ
Taking Laplace transform both sides of Eq. (4.96), we obtain

SX ðSÞ X ð0Þ ¼ AX ðsÞ þ b℧ðSÞ ð4:98Þ
½SI AX ðsÞ ¼ X ð0Þ þ b℧ðSÞ ð4:99Þ
where I is the identity matrix.

From Eq. (4.99), we get
XðsÞ ¼ ½SI A1 ½X ð0Þ þ b℧ðSÞ ð4:100aÞ
X ðsÞ ¼ ½SI A1 X ð0Þ þ ½SI A1 b℧ðSÞ ð4:100bÞ
Taking inverse Laplace transform both sides of Eq. (4.100b) yields

h i h i
L1 ½X ðsÞ ¼ L1 ½SI A1 Xð0Þ þ L1 ½SI A1 b℧ðSÞ ð4:101Þ
h i
L1 ½SI A1 Xð0Þ ¼ eAt X ð0Þ ð4:102Þ
By using convolution theorem, we obtain

h i ðt
1 1
L ½SI A b℧ðSÞ ¼ eAðtτÞ b℧ðτÞ dτ ð4:103Þ
0
Thus,
ðt
1
L ½ X ð s Þ ¼ X ð t Þ ¼ e X ð 0Þ þ
At
eAðtτÞ b℧ðτÞ dτ ð4:104Þ
0
Example 4.31 Consider the electrical circuit given in Example 2.36. Find V c ðt Þ if
V s ðt Þ ¼ uðtÞ under an initially relaxed condition.
4.10 Solution of State-Space Equations Using Laplace Transform 221
Solution "" # " ## " #

s 0 1 1 sþ1 1
½sI A ¼ ¼
0 s 1 1 1 sþ1
" #
1 sþ1 1
½sI A1 ¼ 2
ðs þ 1Þ þ 1 1 s þ 1
" #
cost sint
eAt ¼ L1 ½½sI A1 ¼ et
sint cost
" #
Ðt 0
XðtÞ ¼ eAt Xð0Þ þ 0 eAðtτÞ V s ðτÞdτ
1
Since the circuit is initially relaxed, eAtX(0) ¼ 0.

Therefore,
ðt
AðtτÞ 0
X ðt Þ ¼ e V ðτÞdτ
0 1 s
Since V s ðt Þ ¼ uðt Þ
ðt " #" #
eðtτÞ cos ðt τÞ eðtτÞ sin ðt τÞ 0
X ðt Þ ¼ dτ
0 eðtτÞ sin ðt τÞ eðtτÞ cos ðt τÞ 1
ð t " ðtτÞ #
e sin ðt τÞ
¼ dτ
0 eðtτÞ cos ðt τÞ
Ðt
V c ðt Þ ¼ x2 ðt Þ ¼ 0 eðtτÞ cos ðt τÞ dτ
ðt ðt
eðtτÞ cos ðt τÞdτ ¼ eðtτÞ ðcost cos τ þ sint sin τÞ dτ
0 0
ðt ðt
ðtτÞ
¼ e cost cos τ dτ þ eðtτÞ sint sin τ dτ
0 0
ðt ðt
¼ et cost eτ cos τ dτ þ et sint eτ sin τ dτ
0 0
integration by parts gives

ðt ðt
et cost eτ cos τ dτ ¼ et cost eτ cos τj0t þ eτ sin τ dτ

0 0
Ðt
¼ et cost et cost 1 þ eτ sin τj0t 0 eτ cos τ dτ
This equation can be written as

ðt
2 et cost eτ cos τ dτ ¼ cos 2 t þ sint cost et cost
ðt 0
ð
ð cos 2 t þ sint cost et cost Þ t t
et cost eτ cos τ dτ ¼ e sint eτ sin τ dτ
0 2 0
ðt
t τ t τ
¼ e sint e sin τj0 e cos τ dτ
0
ðt
¼ et sint et sint eτ cos τj0 eτ sin τ dτ

t
which can be written as

Ðt
2 0 et sint eτ sin τ dτ ¼ sin 2 t sint cost þ et sint
Ðt ð sin 2 t sint cost þ et sint Þ
0 et sint eτ sin τ dτ ¼
2
Hence,
ð cos 2 t þ sint cost et cost Þ

V c ðt Þ ¼ x2 ðt Þ ¼
2
ð sin t sint cost þ et sint Þ
2
þ
2
1
¼ ð1 þ et sint et cost Þ, t > 0
2
4.11 Problems
1. Find the Laplace transforms of the following:

(i) x(t) ¼ e2tu(t) þ e3tu(t)
(ii) x(t) ¼ etu(t) þ e3tu(t)

sin 2 t 2 1
ð1 cos 2t Þ
2. Find the Laplace transform of t hint : sint t ¼2 t .
cos 4t cos 5t
3. Find the Laplace transform of t .
4. Show that the ROC for the Laplace transform of a noncausal signal is the region
to the left of a vertical line in the s-plane.
5. Find Laplace transform of a periodic signal with period T, that is, x(t+T) ¼ x(t).
6. By first determining x(t), verify the final value theorem for the following with
comment
1
XðsÞ ¼
s2 þ1
4.11 Problems 223
7. The transfer function of an LTI system is
1
HðsÞ ¼
sþα
Find the impulse response and region of convergence and the value of α for
the system to be causal and stable.
8. Consider a continuous LTI system described by the following differential
equation
d3 y d2 y dy
3
þ 6 2
þ 11 þ 6y ¼ x
dt dt dt
(a) Determine the system function

(b) Determine h(t) for each of the following:
(i) The system is causal
(ii) The system is stable
(iii) The system is neither causal nor stable
9. Find the solution of the following differential equation using Laplace transform
d2 y dy
2 þ 2y ¼ cos t yð0Þ ¼ 1, y_ ð0Þ ¼ 0:
dt 2 dt
10. Consider the following RC circuit with R¼2 ohms, C¼0.5 f.
(a) Determine the differential equation relating V i and V c .
(b) Obtain V c ðt Þ using Laplace transform for V i ðt Þ ¼ e3t uðt Þ with V c ð0Þ ¼ 1.
11. Consider the following RLC circuit.

Obtain y(t) using Laplace transform for x(t)¼u(t).
12. Determine the differential equation characterizing the system represented by the
following block diagram
Y(s)
-5 6
X(s) 1/s 1/s
-7 -12
13. Consider the following state-space representation of a system.

2 3Determine the
3
system output y(t) with the initial state condition X ½0 ¼ 4 3 5.
47
2 3 2 32 2 3 3
x_ 1 ðt Þ 0 1 0 x 1 ðt Þ
0
6 7 6 76 7 6 7 t
6 x_ 2 ðt Þ 7 ¼ 6 0 1 7 6 7 6 7
4 5 4 0 54 x2 ðt Þ 5 þ 4 0 5e
x_ 3 ðt Þ 1 3 3 x 3 ðt Þ 1
2 3
x1 ð t Þ
6 7
yðt Þ ¼ ½ 1 0 0 6 7
4 x2 ð t Þ 5
x3 ð t Þ
Further Reading 225
1. Write a MATLAB program for magnitude response of Sallen-Key low-pass filter.

Further Reading
1. Doetsch, G.: Introduction to the theory and applications of the Laplace transformation with a
table of Laplace transformations. Springer, New York (1974)
2. LePage, W.R.: Complex variables and the Laplace transforms for engineers. McGraw-Hill,
New York (1961)
3. Oppenheim, A.V., Willsky, A.S.: Signals and systems. Prentice-Hall, Englewood Cliffs (1983)
4. Hsu, H.: Signals and systems, 2nd edn. Schaum’s Outlines, Mc Graw Hill (2011)
5. Kailath, T.: Linear systems. Prentice-Hall, Englewood Cliffs (1980)
6. Zadeh, L., Desoer, C.: Linear system theory. McGraw-Hill, New York (1963)
Chapter 5
Analog Filters
Filtering is an important aspect of signal processing. It allows desired frequency

components of a signal to pass through the system without distortion and suppresses
the undesired frequency components. One of the most important steps in the design
of a filter is to obtain a realizable transfer function H(s), satisfying the given
frequency response specifications. In this chapter, the design of analog low-pass
filters is first described. Second, frequency transformations for transforming analog
low-pass filter into band-pass, band-stop, or high-pass analog filters are considered.
The design of analog filters is illustrated with numerical examples. Further, the
design of analog filters using MATLAB is demonstrated with a number of examples.
Also, the design of special filters by pole and zero placement is illustrated with
examples.
5.1 Ideal Analog Filters
An ideal filter passes a signal for one set of frequencies and completely rejects for the
rest of the frequencies.
Low-Pass Filter
The frequency response of an ideal analog low-pass filter HLP (Ω) that passes a
signal for Ω in the range –Ωc Ωc can be expressed by

1, jΩj Ωc
H LP ðΩÞ ¼ ð5:1Þ
0, jΩj > Ωc
The frequency Ωc is called the cutoff frequency.
The impulse response of the ideal low-pass filter corresponds to the inverse
Fourier transform of the frequency response shown in Figure 5.1.

228 5 Analog Filters
Stop band Pass band Stop band

Figure 5.1 Frequency response of ideal low-pass filter
Hence,
ð Ωc
1 sin Ωc t
hð t Þ ¼ ejΩc t dΩ ¼ ð5:2Þ
2π Ωc πt
sinc function can be defined as
sin πx
sincðxÞ ¼ ð5:3Þ
πx
therefore from sinc function we can express Eq. (5.2) as

sin Ωc t Ωc Ωc t
¼ sinc ð5:4Þ
πt π π
Thus,

Ωc Ωc t
hlp ðt Þ ¼ sinc ð5:5Þ
π π
The impulse response for Ωc ¼ 200 Hz is shown in Figure 5.2.
The filter bandwidth is proportional to Ωc and the width of the main lobe is
proportional to Ω1c . The impulse response becomes narrow with increase in the
bandwidth.
High-Pass Filter
The following system (Figure 5.3) is generally used to obtain high-pass filter from a
low-pass filter. The frequency response of an ideal analog high-pass filter HHP (Ω)
that passes a signal for |Ω| > Ωc can be expressed by

0, j Ωj Ωc
H HP ðΩÞ ¼ ð5:6Þ
1, j Ωj > Ωc
and is shown in Figure 5.4. The frequency Ωc is called the cutoff frequency
5.1 Ideal Analog Filters 229
Figure 5.2 Impulse

response for Ωc ¼ 200 Hz
Figure 5.3 System to obtain a high-pass filter from low-pass filter
Figure 5.4 Frequency response of ideal high-pass filter

From Figure 5.3, the frequency response of the ideal high-pass filter can also be
expressed as
H HP ðΩÞ ¼ 1 H LP ðΩÞ ð5:7Þ
Therefore, the impulse response of an ideal high-pass filter is given by the inverse
Fourier transform of Eq. (5.7).
Hence, the impulse response of the ideal high pass filter is given by

Ωc Ωc t
hhp ðt Þ ¼ δðt Þ sinc ð5:8Þ
π π
Band-Pass Filter
The frequency response of band-pass filter can be expressed by

1, Ωc1 jΩj Ωc2
H BP ðΩÞ ¼ ð5:9Þ
0, jΩj < Ωc1 and jΩj > Ωc2
From Figure 5.5, the frequency response of the ideal band-pass filter can be
expressed as
ð Ωc1 ð Ωc2
1 1
hBP ðt Þ ¼ e dΩ þ
jΩt
ejΩt dΩ ð5:10Þ
2π Ωc2 2π Ωc1

1 ejΩt Ωc1 1 ejΩt Ωc2
hBP ðtÞ ¼ þ
2π jt Ωc2 2π jt Ωc1
1 jΩc1 t 1 jΩc2 t
¼ e ejΩc2 t þ e ejΩc1 t
j2πt j2πt
½sin Ωc2 t sin Ωc1 t
¼
tπ
Thus, the impulse response of an ideal band-pass filter is
Figure 5.5 Frequency response of ideal band-pass filter

5.1 Ideal Analog Filters 231

Ωc2 Ωc2 t Ωc1 Ωc1 t
hBP ðt Þ ¼ sinc sinc ð5:11Þ
π π π π
Band-Stop Filter
The band-stop filter can be realized as a parallel combination of low-pass filter

with cutoff frequency Ωc1 and high-pass filter cutoff frequency Ωc2. The frequency
response of band-stop filter can be expressed by
(
1, jΩj Ωc1 and jΩj Ωc2
H BS ðΩÞ ¼ ð5:12Þ
0, Ωc1 < jΩj < Ωc2
From Figure 5.6, the frequency response of the ideal band-stop filter can also be
expressed as
H BS ðΩÞ ¼ H LP ðΩÞ þ H HP ðΩÞ ð5:13Þ
Therefore, the impulse response of an ideal band-stop filter is given by the inverse
Fourier transform of Eq. (5.13).
Hence, the impulse response of the ideal band-stop filter is given by

Ωc1 Ωc1 t Ωc2 Ωc2 t
hBS ðt Þ ¼ δðt Þ þ sinc sinc : ð5:14Þ
π π π π
Figure 5.6 System with

summation of high-pass
filter and low-pass filter
Figure 5.7 Frequency response of ideal band

5.2 Practical Analog Low-Pass Filter Design
A number of approximation techniques for the design of analog low-pass filters are
well established in the literature. The design of analog low-pass filter using
Butterworth, Chebyshev I, Chebyshev II (inverse Chebyshev), and elliptic approx-
imations is discussed in this section.
5.2.1 Filter Specifications
The specifications for an analog low-pass filter with tolerances are depicted in
Figure 5.8, where
Ωp - Passband edge frequency
Ωs - Stopband edge frequency
δp- Peak ripple value in the passband
δs - Peak ripple value in the stopband
Peak passband ripple in dB ¼ αp ¼ 20 log10(1 – δp) dB
Minimum stopband ripple in dB ¼ αs ¼ 20 log10 (δs) dB
Peak ripple value in passband δp ¼ 1 10αp =20
Peak ripple value in stopband δs ¼ 10αs =20
|H (jW)|
Transition
band
1+
1-
Pass band Stop band
Figure 5.8 Specifications of a low-pass analog filter

5.2 Practical Analog Low-Pass Filter Design 233
5.2.2 Butterworth Analog Low-Pass Filter
The magnitude-square response of an Nth-order analog low-pass Butterworth filter is

given by
1
jH a ðjΩÞj2 ¼ ð5:15Þ
1 þ ðΩ=Ωc Þ2N
Two parameters completely characterizing a Butterworth low-pass filter are Ωs

and N. These are determined from the specified band edges Ωp and Ωc, peak
passband ripple αp, and minimum stopband attenuation αs. The first (2N 1)
derivatives of |Ha( jΩ|2 at Ω ¼ 0 are equal to zero. Thus, the Butterworth low-pass
filter is said to have a maximally flat magnitude at Ω ¼ 0. The gain in dB is given by
10log10|Ha( jΩ|2. At Ω ¼ Ωc, the gain is 10log10(0.5) ¼ 3 dB; therefore, Ωc is
called the 3 dB cutoff frequency. The loss in dB in a Butterworth filter is given by

α ¼ 10log 1 þ ðΩ=Ωc Þ2N ð5:16Þ
For Ω ¼ Ωp, the passband attenuation is given by

2N
αp ¼ 10log 1 þ Ωp =Ωc ð5:17Þ
For Ω ¼ Ωs, the stopband attenuation is

αs ¼ 10log 1 þ ðΩs =Ωc Þ2N ð5:18Þ
Eqs.(5.17) and (5.18) can be rewritten as

2N
Ωp =Ωc ¼ 100:1αp 1 ð5:19Þ
ðΩs =Ωc Þ2N ¼ 100:1αs 1 ð5:20Þ
From Eqs.(5.19) and (5.20), we obtain

1=2N

100:1αs 1
Ωs =Ωp ¼ ð5:21Þ
100:1αp 1
0:1αs

1 10 1
log Ωs =Ωp ¼ log ð5:22Þ
2N 100:1αp 1
From Eq. (5.22), solving for N we get

100:1αs 1
log 100:1αp 1
N ð5:23Þ
2logðΩs =ΩpÞ
Since the order N must be an integer, the value obtained is rounded to the next
higher integer. This value of N is used in either Eq. (5.19) or Eq. (5.20) to determine
the 3 dB cutoff frequency Ωc. In practice, Ωc is determined by Eq. (5.20) that exactly
satisfies stopband specification at Ωc, while the passband specification is exceeded
with a safe margin at Ωp. We know that |H( jΩ)|2 may be evaluated by letting s ¼ jΩ
in H(s)H(s), which may be expressed as
1
H ðsÞH ðsÞ ¼
N ð5:24Þ
1 þ s2 =Ω2c
If Ωc ¼ 1, the magnitude response |HN( jΩ)| is called the normalized magnitude

response. Now, we have

N Y
2N
1 þ s2 ¼ ðs sk Þ ð5:25Þ
k¼1
where
(
ejð2k1Þπ=2N for n even
sk ¼ ð5:26Þ
ejðk1Þπ=N for n odd
Since |sk| ¼ 1, we can conclude that there are 2N poles placed on the unit circle in
the s-plane. The normalized transfer function can be formed as
1
H N ðsÞ ¼ ð5:27Þ
Q
N
ð s pl Þ
l¼1
where pl for l ¼ 1, 2,.., N are the left half s-plane poles. The complex poles occur in
conjugate pairs.
For example, in the case of N ¼ 2, from Eq. (5.26), we have

ð2k 1Þπ ð2k 1Þπ
sk ¼ cos þ j sin k ¼ 1, . . . ::, 2N
4 4
The poles in the left half of the s-plane are
1 j 1 j
s2 ¼ pffiffiffi þ pffiffiffi ; s3 ¼ pffiffiffi pffiffiffi
2 2 2 2
Hence,
1 j 1 j
p1 ¼ pffiffiffi þ pffiffiffi ; p2 ¼ pffiffiffi pffiffiffi
2 2 2 2
and
1
H N ðsÞ ¼ pffiffiffi
s2 þ 2s þ 1
In the case of N ¼ 3,

ðk 1Þπ ðk 1Þπ
sk ¼ cos þ j sin k ¼ 1, . . . ::, 2N
3 3
The left half of s-plane poles are

1 j 3 1 j 3
s3 ¼ þ ; s4 ¼ 1; s5 ¼
2 2 2 2
Hence
1 j 3 1 j 3
p1 ¼ þ ; p2 ¼ 1; p3 ¼ ;
2 2 2 2
and
1
H N ðsÞ ¼
ð s þ 1Þ ð s 2 þ s þ 1Þ
The following MATLAB Program 5.1 can be used to obtain the Butterworth
normalized transfer function for various values of N.
Program 5.1 Analog Butterworth Low-Pass Filter Normalized Transfer Function
N=input(‘enter order of the filter’);

[z,p,k] = buttap(N)% determines poles and zeros
disp(‘Poles are at’);disp(p);
[num,den]=zp2tf(z,p,k);
%Print coefficients in powers of s
disp(‘Numerator polynomial’);disp(num);
disp(‘Denominator polynomial’);disp(den);
sos=zp2sos(z,p,k);%determines coefficients of second order sections
The normalized Butterworth polynomials generated from the above program for
typical values of N are tabulated in Table 5.1.
The magnitude response of the normalized Butterworth low-pass filter for some
typical values of N is shown in Figure 5.9. From this figure, it can be seen that the
response monotonically decreases both in the passband and the stopband as Ω
Table 5.1 List of normalized Butterworth polynomials

N Denominator of HN(s)
1 Sþ1
pffiffiffi
2 s2 þ 2s þ 1
3 (s þ 1)(s2 + s þ 1)
4 (s2 þ 0.76537s þ 1)(s2 þ 1.8477s þ 1)
5 (s þ 1) (s2 þ 0.61803s þ 1)(s2 þ 1.61803s þ 1)

pffiffiffi
6 ðs2 þ 1:931855s þ 1Þ s2 þ 2s þ 1 ðs2 þ 0:51764s þ 1Þ
7 (s þ 1) (s2 þ 1.80194s þ 1)(s2 þ 1.247s þ 1)(s2 þ 0.445s þ 1)
Figure 5.9 Magnitude response of typical Butterworth low-pass filter
increases. As the filter order N increases, the magnitude responses both in the
passband and the stopband are improved with a corresponding decrease in the
transition width. Since the normalized transfer function corresponds to Ωc ¼ 1, the
transfer function of the low-pass filter corresponding to the actual Ωc can be obtained
by replacing s by (s/Ωc) in the normalized transfer function.
Example 5.1 Design a Butterworth analog low-pass filter with 1 dB passband ripple,
passband edge frequency Ωp ¼ 2000π rad/sec, stopband edge frequency
Ωs ¼ 10,000π rad/sec, and a minimum stopband ripple of 40 dB.
Solution Since αs ¼ 40 dB, αp ¼ 1 dB, Ωp ¼ 2000π, and Ωs ¼ 10,000π,

100:1αs 1 104 1
log ¼ log ¼ 4:5868:
100:1αp 1 100:1 1
Hence from (5.23),

104 1
log 100:1 1 4:5868
N ¼ ¼ 3:2811
2logð5=1Þ 1:3979
Since the order must be an integer, we choose N ¼ 4.

The normalized low-pass Butterworth filter for N ¼ 4 can be formulated as
1
H N ðsÞ ¼
ðs2 þ 0:76537s þ 1Þðs2 þ 1:8477s þ 1Þ
From Eq. (5.20), we have
Ωs 10000π
Ωc ¼
1=2N ¼
1=8 ¼ 9935
10 1
4
104 1
The transfer function for Ωc ¼ 9935 can be obtained by replacing s by (s/Ωc) ¼

(s/9935)in HN (s).
1 1
H a ðsÞ ¼
s
2 s
s 2
9935 þ 0:76537 þ 1 9935 þ 1:8477 þ1
s
9935 9935
9:7425 1015
¼
2

s þ 7:604 103 s þ 9:8704225 107 s2 þ 1:8357 104 s þ 9:8704225 107
5.2.3 Chebyshev Analog Low-Pass Filter
Type 1 Chebyshev Low-Pass Filter

The magnitude-square response of an Nth-order analog low-pass Type 1 Chebyshev
filter is given by
1
jH ðΩÞj2 ¼
ð5:28Þ
1 þ ε2 T 2N Ω=Ωp
where TN(Ω) is the Chebyshev polynomial of order N

(
cos ðN cos 1 ΩÞ, j Ωj 1
T N ð ΩÞ ¼
ð5:29Þ
cosh Ncosh1 Ω , j Ωj > 1
The loss in dB in a Type 1 Chebyshev filter is given by

α ¼ 10log 1 þ ε2 T 2N Ω=Ωp ð5:30Þ
For Ω ¼ Ωp, TN(Ω) ¼ 1, and the passband attenuation is given by

αp ¼ 10log 1 þ ε2 ð5:31Þ
From Eq. (5.31), ε can be obtained as

pffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffi
ε¼ 100:1αp 1 ð5:32Þ
For Ω ¼ Ωs the stopband attenuation is
αs ¼ 10logð1 þ22 T 2n ðΩs =Ωp ÞÞ ð5:33Þ
Since (Ωs/Ωp) > 1, the above equation can be written as

αs ¼ 10log 1 þ22 cosh2 Ncosh1 Ωs =Ωp ð5:34Þ
Substituting Eq. (5.32) for ε in the above equation and solving for N, we get
qffiffiffiffiffiffiffiffiffiffiffiffiffiffi
cosh1 100:1αs 1
100:1αp 1
N 1

ð5:35Þ
cosh Ωs =Ωp
We choose N to be the lowest integer satisfying (5.35). In determining N using the

above equation, it is convenient to evaluate cosh1(x) by applying the identity
1
pffiffiffiffiffiffiffiffiffiffiffiffiffi
cosh ðxÞ ¼ ln x þ x2 1 .
The poles of the normalized Type 1 Chebyshev filter transfer function lie on an
ellipse in the s-plane and are given by

1 1 1 ð2k 1Þπ
xk ¼ sinh sinh sin for k ¼ 1, 2, ::::, N ð5:36Þ
N 2 2N

1 1 ð2k 1Þπ
yk ¼ cosh sinh1 cos for k ¼ 1, 2, :::, N ð5:37Þ
N 2 2N
Also, the normalized transfer function is given by
H0
H N ðsÞ ¼ ð5:38Þ
Π k ðs p k Þ
where

1 1 1 ð2k 1Þπ 1 1 1 ð2k 1Þπ
pk ¼ sinh sinh sin þ j cosh sinh cos
N 2 2N N 2 2N
ð5:39aÞ
and
1 1
H0 ¼ ð5:39bÞ
2N1 ε
As an illustration, consider the case of N ¼ 2 with a passband ripple of 1 dB. From
Eq. (5.32), we have
1 1
¼ pffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffi ¼ 1:965227
ε 10 0:1αp
1
Hence

11
sinh ¼ sinh1 ð1:965227Þ ¼ 1:428
ε
Therefore, from (5.39a), the poles of the normalized Chebyshev transfer function
are given by

pk ¼ sinhð0:714Þ sin þ j coshð0:714Þ cos , k ¼ 1, 2
4 4
Hence
p1 ¼ 0:54887 þ j0:89513, p2 ¼ 0:54887 j0:89513
Also, from (5.39b), we have
1
H 0 ¼ ð1:965227Þ ¼ 0:98261
2
Thus for N ¼ 2, with a passband ripple of 1 dB, the normalized Chebyshev
transfer function is
0:98261 0:98261
H N ðsÞ ¼ ¼ 2
ðs p1 Þðs p2 Þ ðs þ 1:098s þ 1:103Þ
Similarly for N ¼ 3, for a passband ripple of 1 dB, we have

pk ¼ sinhð1:428=3Þ sin þ j coshð1:428=3Þ cos , k ¼ 1, 2, 3
6 6
Thus,
p1 ¼ 0:24709 þ j0:96600; p2 ¼ 0:49417; p3 ¼ 0:24709 j0:966:
Also, from (5.39b),
1
H 0 ¼ ð1:965227Þ ¼ 0:49131
4
Hence, the normalized transfer function of Type 1 Chebyshev low-pass filter for
N¼3 is given by
0:49131 0:49131
H N ðsÞ ¼ ¼
ðs p1 Þðs p2 Þðs p3 Þ ðs3 þ 0:988s2 þ 1:238s þ 0:49131Þ
The following MATLAB Program 5.2 can be used to form the Type1 Chebyshev
normalized transfer function for a given order and passband ripple.
Program 5.2 Analog Type 1 Chebyshev Low-Pass Filter Normalized Transfer
Function

Rp=input(‘enter passband ripple in dB’);
[z,p,k] = cheb1ap(N,Rp)% determines poles and zeros
The normalized Type 1 Chebyshev polynomials generated from the above pro-
gram for typical values of N and passband ripple of 1 dB are tabulated in Table 5.2.
The typical magnitude responses of a Type 1 Chebyshev low-pass filter for N ¼ 3,
5, and 8 with 1 dB passband ripple are shown in Figure 5.10. From this figure, it is
seen that Type 1 Chebyshev low-pass filter exhibits equiripple in the passband with a
monotonic decrease in the stopband.
Example 5.2 Design a Type 1 Chebyshev analog low-pass filter for the specifica-
tions given in Example 5.1.
Table 5.2 List of normalized Type 1 Chebyshev transfer functions for passband ripple ¼ 1 dB
N Denominator of HN(s) H0
1 S þ 1.9652 1.9652
2 s2 þ 1.0977s þ 1.1025 0.98261
3 s3 þ 0.98834s2 þ 1.2384s þ 0.49131 0.49131
4 s4 þ 0.95281s3 þ 1.4539s2 þ 0.74262s þ 0.27563 0.24565
5 s5 þ 0.93682s4 þ 1.6888s3 þ 0.9744s2 þ 0.58053s þ 0.12283 0.12283
Figure 5.10 Magnitude response of typical Type 1 Chebyshev low-pass filter with 1 dB passband
ripple
Solution Since αs ¼ 40 dB, αp ¼ 1 dB, Ωp ¼ 2000π, and Ωs ¼ 10,000π,

sffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffi sffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffi
100:1αs 1
1 1 104 1
cosh ¼ cosh ¼ cosh1 ð196:52Þ
10 0:1αp
1 100:1 1

cosh1 Ωs =Ωp ¼ cosh1 ð5Þ ¼ 2:2924
sffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffi
1 104 1
cosh
100:1 1
N ¼ 2:6059
cosh1 ð5Þ
Since the order of the filter must be an integer, we choose the next higher integer
value 3 for N. The normalized Type 1 Chebyshev low-pass filter for N ¼ 3 with a
passband ripple of 1 dB is given from Table 5.2 as
0:49131
H N ðsÞ ¼
s3 þ 0:988s2 þ 1:238s þ 0:49131
The transfer function for Ωp ¼ 2000π is obtained by substituting s ¼ (s/Ωp) ¼
(s/2000π) in HN(s):
0:49131
H a ðsÞ ¼
s 3 s 2 s
þ 0:988 þ 1:238 þ 0:49131
2000π 2000π 2000π
1:2187 1011
¼ 3
s þ 6:2099 10 s þ 4:889 107 s þ 1:2187 1011
3 2
Type 2 Chebyshev Filter

The squared-magnitude response of Type 2 Chebyshev low-pass filter, which is also
known as the inverse Chebyshev filter, is given by
1
jH ðΩÞj2 ¼ ð5:40Þ
T 2N ðΩs =Ωp Þ
1 þ ε2 T 2N ðΩs =ΩÞ
The order N can be determined using Eq. (5.35). The Type 2 Chebyshev filter has
both poles and zeros, and the zeros are on the jΩ axis. The normalized Type
2 Chebyshev low-pass filter, or the normalized inverse Chebyshev filter (normalized
to Ωs ¼ 1), may be formed as
Πk ðs zk Þ
H N ðsÞ ¼ H 0 , k ¼ 1, 2, ::, N ð5:41Þ
Πk ðs pk Þ
where
1
zk ¼ j ð2k1Þπ
for k ¼ 1, 2, ::, N ð5:42aÞ
cos N
σk ωk
pk ¼ þj 2 for k ¼ 1, 2, ::, N ð5:42bÞ
σ 2k þ Ω2k σ k þ Ω2k

1 1 ð2k 1Þπ
σ k ¼ sinh sinh1 sin for k ¼ 1, 2, ::, N ð5:42cÞ
N δs 2N

1 1 1 ð2k 1Þπ
Ωk ¼ cosh sinh cos for k ¼ 1, 2, ::, N ð5:42dÞ
N δs 2N
1
δs ¼ pffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffi ð5:42eÞ
10 0:1αs
1
Πk ðpk Þ
H0 ¼ ð5:42fÞ
Πk ðzk Þ
For example, if we consider N ¼ 3 with a stopband ripple of 40 dB, then from

(5.42e),
1 pffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffi pffiffiffiffiffiffiffiffiffiffiffiffiffiffiffi
¼ 100:1αs 1 ¼ 104 1 ¼ 99:995
δs
Hence,

1
sinh1 ¼ 5:28829
δs
Using (5.42c) and (5.42d), we have

ð2k 1Þπ
σ k ¼ sinhð5:28829=3Þ sin for k ¼ 1, 2, 3
6
ð2k 1Þπ
Ωk ¼ coshð5:28829=3Þ cos for k ¼ 1, 2, 3
6
Hence
σ 1 ¼ 1:41927, σ 2 ¼ 2:83854, σ 3 ¼ 1:41927

Ω1 ¼ 2:60387, Ω2 ¼ 2:83854, Ω3 ¼ 2:60387
Thus, from (5.42b), the poles are
p1 ¼ 0:16115 þ j0:29593, p2 ¼ 0:3523, p3 ¼ 0:16115 þ j0:29593
Also, using (5.42a), the zeros are given by

z1 ¼ j 2= 3 , z2 ¼ j 2= 3
Finally, from (5.42f),
H 0 ¼ 0:03
Therefore, the normalized Type 2 Chebyshev low-pass filter for N ¼ 3 with a

stopband ripple of 40 dB is given by
0:03ðs z1 Þðs z2 Þ 0:03ðs2 þ 1:3333Þ

H N ðsÞ ¼ ¼ 3
ðs p1 Þðs p2 Þðs p3 Þ ðs þ 0:6746s2 þ 0:22709s þ 0:04Þ
The following MATLAB Program 5.3 can be used to form the Type2 Chebyshev
normalized transfer function for a given order and stopband ripple.
Program 5.3 Analog Type 2 Chebyshev Low-Pass Filter Normalized Transfer
Function

Rs=input(‘enter stopband attenuation in dB’);
[z,p,k] = cheb2ap(N,Rs);% determines poles and zeros
The normalized Type 2 Chebyshev transfer functions generated from the above
program for typical values of N with a stopband ripple of 40 dB are tabulated in
Table 5.3.
Table 5.3 List of normalized Type 2 Chebyshev transfer functions for stopband ripple ¼ 40 dB
Order N HN (s)
1 0:01
s þ 0:01
2 0:01s2 þ 0:02
s þ 0:199s þ 0:02
2
3 0:03s2 þ 0:04
s3 þ 0:6746s2 þ 0:2271s þ 0:04
4 0:01s4 þ 0:08s2 þ 0:08
s þ 1:35s3 þ 0:9139s2 þ 0:3653s þ 0:08
4
5 0:05s4 þ 0:2s2 þ 0:16

s þ 2:1492s þ 2:3083s3 þ 1:5501s2 þ 0:6573s þ 0:16
5 4
6 0:01s6 þ 0:18s4 þ 0:48s2 þ 0:32

s6 þ 3:0166s5 þ 4:5519s4 þ 4:3819s3 þ 2:8798s2 þ 1:2393s þ 0:32
Figure 5.11 Magnitude response of typical Type 2 Chebyshev low-pass filter with 20 dB stopband
ripple
The typical magnitude response of a Type 2 Chebyshev low-pass filter for N ¼ 4

and 7 with 20 dB stopband ripple is shown in Figure 5.11. From this figure, it is seen
that Type 2 Chebyshev low-pass filter exhibits monotonicity in the passband and
equiripple in the stopband.
Example 5.3 Design a Type 2 Chebyshev low-pass filter for the specifications given
in Example 5.1.
Solution The order N is chosen as 3, as in Example 5.2, since the equation for order
finding is the same for both Type 1 and Type 2 Chebyshev filters. The normalized
Type 2 Chebyshev low-pass filter for N ¼ 3 with a stopband ripple of 40 dB has

already been found earlier and is given by
0:03ðs2 þ 1:3333Þ
H N ðsÞ ¼
ðs3 þ 0:6746s2 þ 0:2271s þ 0:04Þ
For Ωs ¼ 10,000π, the corresponding transfer function can be obtained by

substituting s ¼ (s/Ωs) ¼ (s/10000π) in the above expression for HN(s). Thus, the
required filter transfer function is

s 2
0:03 10000π þ 0:04
H a ðsÞ ¼

s
3 2
s
10000π þ 0:6746 10000π
s
þ 0:22709 þ 0:04
10000π
9:4252 102 s2 þ 1:2403 1012
¼ 3
s þ 2:1193 104 s2 þ 2:2413 108 s þ 1:2403 1012
5.2.4 Elliptic Analog Low-Pass Filter
The square-magnitude response of an elliptic low-pass filter is given by
1
jH a ðjΩÞj2 ¼
ð5:43Þ
1þ ε2 U N Ω=Ωp
where UN(x) is the Jacobian elliptic function of order N and ε is a parameter related to
the passband ripple. In an elliptic filter, a constant k, called the selectivity factor,
representing the sharpness of the transition region is defined as
Ωp
k¼ ð5:44Þ
Ωs
A large value of k represents a wide transition band, while a small value indicates
a narrow transition band.
For a given set of Ωp, Ωs, αp, and αs, the filter order can be estimated using the
formula

100:1αs 1
log 16 100:1αp
1
Nffi ð5:45Þ
log10 ð1=ρÞ
where ρ can be computed using

pffiffiffiffi
1 k0
ρ0 ¼ pffiffiffiffi ð5:46Þ
2 1 þ k0
pffiffiffiffiffiffiffiffiffiffiffiffiffi
k0 ¼ 1 k2 ð5:47Þ
ρ ¼ ρ0 þ 2ðρ0 Þ5 þ 15ðρ0 Þ9 þ 150ðρ0 Þ13 ð5:48Þ
The following MATLAB Program 5.4 can be used to form the elliptic normalized
transfer function for given filter order and passband ripple and stopband attenuation.
The normalized passband edge frequency is set to 1.
Program 5.4 Analog Elliptic Low-Pass Filter Normalized Transfer Function

Rp=input(‘enter passband ripple in dB’);
Rs=input(‘enter stopband attenuation in dB’);
[z,p,k] = ellipap(N,Rp,Rs)% determines poles and zeros
[num,den] =zp2tf(z,p,k);
The normalized elliptic transfer functions generated from the above program for
typical values of N and stopband ripple of 40 dB are tabulated in Table 5.4.
The magnitude response of a typical elliptic low-pass filter is shown in
Figure 5.12, from which it can be seen that it exhibits equiripple in both the passband
and the stopband.
Example 5.4 Design an elliptic analog low-pass filter for the specifications given in
the Example 5.1.
Table 5.4 List of normalized elliptic transfer functions for passband ripple ¼ 1 dB and stopband
ripple ¼ 40 dB
Order N HN(s)
1 1:9652
s þ 1:9652
2 0:01s2 þ 0:9876
s2 þ 1:0915s þ 1:1081
3 0:0692s2 þ 0:5265
s þ 0:9782s2 þ 1:2434s þ 0:5265
3
4 0:01s4 þ 0:1502s2 þ 0:3220

s þ 0:9391s3 þ 1:5137s2 þ 0:8037s þ 0:3612
4
5 0:0470s4 þ 0:2201s2 þ 0:2299

s5 þ 0:9234s4 þ 1:8471s3 þ 1:1292s2 þ 0:7881s þ 0:2299
6 0:01s6 þ 0:1172s4 þ 0:28s2 þ 0:186
s þ 0:9154s þ 2:2378s4 þ 1:4799s3 þ 1:4316s2 þ 0:5652s þ 0:2087
6 5
Figure 5.12 Magnitude response of typical elliptic low-pass filter with 1 dB passband ripple and
30 dB stopband ripple
Solution
Ωp 2000π
k¼ ¼ ¼ 0:2
Ωs 10000π
and
pffiffiffiffiffiffiffiffiffiffiffiffiffi pffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffi
k0 ¼ 1 k2 ¼ 1 0:04 ¼ 0:979796:
Substituting these values in Eq. (5.46) and Eq. (5.47), we get
ρ0 ¼ 0:00255135,
ρ ¼ 0:0025513525
and hence

104 1
log 16 10 0:1
1
N¼
log10 ð0:0025513525
1
Þ ¼ 2:2331:
Choose N ¼ 3. Then, for N ¼ 3, a passband ripple of 1 dB, and a stopband ripple

of 40dB, the normalized elliptic transfer function is as given in Table 5.4. For
Ωp ¼ 2000π, the corresponding transfer function can be obtained by substituting
s ¼ (s/Ωp) ¼ (s/2000π) in the expression for HN(s). Thus, the required filter transfer
function is

s 2
0:0692 2000π þ 0:5265
H a ðsÞ ¼
s 3 s 2 s
þ 0:97825 þ 1:2434 þ 0:5265
2000π 2000π 2000π
4:348 102 s2 þ 1:306 1011
¼ 3
s þ 6:1465 103 s2 þ 4:9087 107 s þ 1:306 1011
5.2.5 Bessel Filter
Bessel filter is a class of all-pole filters that provide linear phase response in the
passband and characterized by the transfer function
1
H a ðsÞ ¼ ð5:49Þ
a0 þ a1 s þ a2 s2 þ þ aN1 sN1 þ aN sN
where the coefficients an are given by
ð2N nÞ!
an ¼ ð5:50Þ
2 n!ðN nÞ!
Nn
The magnitude responses of a third-order Bessel filter and Butterworth filter are
shown in Figure 5.13 and the phase responses of the same filters with the same order
are shown in Figure 5.14. From these figures, it is seen that the magnitude response
Figure 5.13 Magnitude response of a third-order Bessel filter and Butterworth filter
Figure 5.14 Phase response of a third-order Bessel filter and Butterworth filter
of the Bessel filter is poorer than that of the Butterworth filter, whereas the phase
response of the Bessel filter is more linear in the passband than that of the
Butterworth filter.
5.2.6 Comparison of Various Types of Analog Filters
The magnitude response and phase response of the normalized Butterworth,

Chebyshev Type 1, Chebyshev Type 2, and elliptic filters of the same order are
compared with the following specifications:
filter order ¼ 8, maximum passband ripple ¼ 1 dB and minimum stopband ripple

¼ 35 dB:
The following MATLAB program is used to generate the magnitude and phase
responses for these specifications.
Program 5.5 Magnitude and Phase Responses of Analog Filters of Order 8 with a
Passband Ripple of 1 dB and a Stopband Ripple of 35 dB
clear all;clc;
[z,p,k]=buttap(8);
[num1,den1]=zp2tf(z,p,k);[z,p,k]=cheb1ap(8,1);
[num2,den2]=zp2tf(z,p,k);[z,p,k]=cheb2ap(8,35);
[num3,den3]=zp2tf(z,p,k); [z,p,k]=ellipap(8,1,35);
[num4,den4]=zp2tf(z,p,k);
omega=[0:0.01:5];
h1=freqs(num1,den1,omega);h2=freqs(num2,den2,omega);
h3=freqs(num3,den3,omega);h4=freqs(num4,den4,omega);
ph1=angle(h1);ph1=unwrap(ph1);
figure(1),plot(omega,20*log10(abs(h1)),‘-’);hold on
plot(omega,20*log10(abs(h2)),‘--’);hold on
plot(omega,20*log10(abs(h3)),‘: ’);hold on
plot(omega,20*log10(abs(h4)),‘-.’);
xlabel(‘Normalized frequency’);ylabel(‘Gain,dB’);axis([0 5 -80 5]);
legend(‘Butterworth’,‘Chebyshev Type 1’,‘Chebyshev Type 2’,‘Ellip-
tic’);hold off
figure(2),plot(omega,ph1,‘-’);hold on
plot(omega,ph2,‘--’);hold on
plot(omega,ph3,‘: ’);hold on
plot(omega,ph4,‘-.’)
xlabel(‘Normalized frequency’);ylabel(‘Phase,radians’);axis([0 5 -8 0]);
legend(‘Butterworth’,‘Chebyshev Type 1’,‘Chebyshev Type 2’,‘Elliptic’);
The magnitude and phase responses for the above specifications are shown in
Figure 5.15. The magnitude response of Butterworth filter decreases monotonically
both in passband and stopband with wider transition band. The magnitude response
of the Chebyshev Type 1 exhibits ripples in the passband, whereas the Chebyshev
Type 2 has approximately the same magnitude response to that of the Butterworth
filter. The transition band of both the Type 1 and Type 2 Chebyshev filters is the
same, but less than that of the Butterworth filter. The elliptic filter exhibits an
equiripple magnitude response both in the passband and the stopband with a
transition width smaller than that of the Chebyshev Type 1 and Type 2 filters. But
the phase response of the elliptic filter is more nonlinear in the passband than that of
the phase response of the Butterworth and Chebyshev filters. If linear phase in the
passband is the stringent requirement, then the Bessel filter is preferred, but with a
poor magnitude response.
Another way of comparing the various filters is in terms of the order of the filter
required to satisfy the same specifications. Consider a low-pass filter that meets the
passband edge frequency of 450 Hz, stopband edge frequency of 550 Hz, passband
ripple of 1 dB, and stopband ripple of 35 dB. The orders of the Butterworth,
Chebyshev Type 1, Chebyshev Type2, and elliptic filters are computed for the
above specifications and listed in Table 5.5. From this table, we can see that elliptic
filter can meet the specifications with the lowest filter order.
Figure 5.15 A comparison of various types of analog low-pass filters: (a) magnitude response and
(b) phase response
Table 5.5 Comparison of Filter Order

orders of various types of
Butterworth 24
filters
Chebyshev Type 1 9
Chebyshev Type 2 9
Elliptic 5
5.2.7 Design of Analog High-Pass, Band-Pass,

and Band-Stop Filters
The analog high-pass, band-pass, and band-stop filters can be designed using analog
frequency transformations. In this design process, first, the analog prototype low-
pass filter specifications are derived from the desired specifications of the analog
filter using suitable analog-to-analog transformation. Next, by using the specifica-
tions so obtained, a prototype low-pass filter is designed. Finally, the transfer
function of the desired analog filter is determined from the transfer function of the
prototype analog low-pass transfer function using the appropriate analog-to-analog
frequency transformation. The low-pass to low-pass, low-pass to high-pass,
low-pass to band-pass, and low-pass to band-stop analog transformations are
considered next.
Low Pass to Low Pass
Let Ωp ¼ 1 and Ω b p be the passband edge frequencies of the normalized prototype

low-pass filter and the desired low-pass filter, as shown in Figure 5.16. The trans-
formation from the prototype low pass to the required low pass must convert Ω b ¼0
b
to Ω ¼ 0 and Ω ¼ 1 to Ω ¼ 1. The transformation such as s ¼ kbs or Ω
¼ k
Ωb achieves the above transformation for any positive value of k. If k is chosen to

be 1=Ω b p gets transformed to Ωp ¼ 1, and Ω
b p , then Ω b s to Ωs ¼ Ωb s =Ωb p . Since
Ωp ¼ 1 is the passband edge frequency for the normalized Type I Chebyshev and
elliptic low-pass filters, we have the design equations for these filters as
Ωp ¼ 1, Ωs ¼ Ω b s =Ω
b p: ð5:51aÞ

Also, the transfer function H LP bs for these filters is related to the corresponding
normalized low-pass transfer function HN (s) by

H LP bs ¼ H N ðsÞc bp ð5:51bÞ
s¼b
s=Ω
However, in the case of a Butterworth filter, since

Ω
¼ 1 corresponds to the cutoff
frequency of the filter, the transfer function H LP bs for the Butterworth filter is
related to the normalized low-pass Butterworth transfer function HN (s) by
Figure 5.16 Low-pass to

low-pass frequency
transformation.
(a) Prototype Low-pass filter
frequency response.
(b) Low-pass filter
frequency response
(a)
(b)

H LP bs ¼ H N ðsÞc bc ð5:51cÞ
s¼b
s=Ω
where Ωb c is the cutoff frequency of the desired Butterworth filter and is given by

Eq. (5.19). For similar reasons, the transfer function H LP bs for the Type
2 Chebyshev filter is related to the normalized transfer function HN (s) by

H LP bs ¼ H N ðsÞc bs ð5:51dÞ
s¼b
s=Ω
Low Pass to High Pass (Figure 5.17)
Let the passband edge frequencies of the prototype low-pass and the desired high-
pass filters be Ωp ¼ 1 and Ω b p , as shown in Figure 5.17. The transformation from
prototype low pass to the desired high pass must transform Ω b ¼ 0 to Ω ¼ 1 and
b ¼ 1 to Ω ¼ 0. The transformation such as s ¼ k=bs or Ω ¼ k=Ω
Ω b achieves the
(a)
(b)
Figure 5.17 Low-pass to high-pass frequency transformation. (a) Prototype low-pass filter fre-
quency response. (b) High-pass filter frequency response
above transformation for any positive value of k. By transforming Ω b p to Ωp ¼ 1,

b
the constant k can be determined as k ¼ Ω p .Thus, design equations are
Ωp ¼ 1, Ωs ¼ Ω b p =Ω
b s, ð5:52aÞ

and the desired transfer function H HP bs is related to the low-pass transfer function
HN (s) by

H HP bs ¼ H N ðsÞj b ð5:52bÞ
s¼Ω p=b
s
(a)
(b)
Figure 5.18 Low-pass to band-pass frequency transformation. (a) Prototype low-pass filter fre-
quency response. (b) Band-pass filter frequency response
The above equations (5.52a) and (5.52b) hold for all filters except for Butterworth
and Type 2 Chebyshev filter. For Butterworth

H LP bs ¼ H N ðsÞc b ð5:53aÞ
s¼bs=Ω c

H HP bs ¼ H N ðsÞ b ð5:53bÞ
s¼Ω p=b
s
For Type 2 Chebyshev filter, the design equations are

b s =Ω
Ωp ¼ Ω b p , Ωs ¼ 1 ð5:53cÞ
and

H HP bs ¼ H N ðsÞc b ð5:53dÞ
s¼Ω s =b
s
Example 5.5 Design a Butterworth analog high-pass filter for the following
specifications:
Passband edge frequency: 30.777 Hz
Stopband edge frequency: 10 Hz
Passband ripple: 1 dB
Stopband ripple: 20 dB
Solution For the prototype analog low-pass filter, we have
b p =Ω
Ωp ¼ 1, Ωs ¼ Ω b s ¼ 3:0777, αp ¼ 1 dB, αs ¼ 20 dB
Substituting these values in Eq. (5.23), the order of the filter is given by

102 1
log 100:1 1
N
¼ 2:6447
2 log 3:077
1
Hence, we choose N ¼ 3. From Table 5.1, the third-order normalized Butterworth

low-pass filter transfer function is given by
1
H N ðsÞ ¼
Substituting the values of Ωs and N in Eq. (5.20), we obtain

3:0777 6
¼ 102 1
Ωc
Solving for Ωc, we get Ωc ¼ 1.4309.

The analog transfer function of the low-pass filter is obtained from the above
transfer function by substituting s ¼ Ωsc ¼ 1:4309
s
; hence,
2:93
H LP ðsÞ ¼
s3 þ 2:8619s2 þ 4:0952s þ 2:93
From the above transfer function, the analog transfer function of the high-pass
_
Ωp 3:0777
filter can be obtained by substituting s ¼ ¼
s s
s3
H HP ðsÞ ¼
s3 þ 4:3017s2 þ 9:2521s þ 9:9499
Example 5.6 Design a Butterworth analog high-pass filter for the specifications of
Example 5.5 using MATLAB
Solution The following MATLAB code fragments can be used to design HHP (s)
[N,Wn]=buttord(1,3.0777,1,20,‘s’);
[B,A]=butter(N,Wn,‘s’);
[num, den]=1p2hp(B,A,3.0777);
The transfer function HLP(s) of the analog low-pass filter can be obtained by
displaying numerator and denominator coefficient vectors B and A and is given by
2:93
H LP ðsÞ ¼
s3 þ 2:8619s2 þ 4:0952s þ 2:93
The transfer function HHP (s) of the analog high-pass filter can be obtained
by displaying numerator and denominator coefficient vectors num and den and is
given by
s3
H HP ðsÞ ¼
s3 þ 4:3017s2 þ 9:2521s þ 9:499
Low Pass to Band Pass

The prototype low-pass and the desired band-pass filters are shown in Figure 5.18. In
b p1 is the lower passband edge frequency, Ω
this figure, Ω b p2 the upper passband edge
frequency, Ω b s2 the upper stopband edge
b s1 the lower stopband edge frequency, and Ω
frequency of the desired band-pass filter. Let us denote by Bp the bandwidth of the
passband and by Ω b mp the geometric mean between the passband edge frequencies of
the band-pass filter, i.e.,
b p2 Ω
Bp ¼ Ω b p1 ð5:54aÞ
qffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffi
b mp ¼ Ω
Ω b p1 Ω b p2 ð5:54bÞ
Now, consider the transformation

2
bs þ Ωb2
mp
S¼ ð5:55Þ
Bpbs
b ¼ 0, Ω
As a consequence of this transformation, it is seen that Ω b p1 , Ω
b mp , Ω
b p2 ,
and 1 transform to the frequencies Ω ¼ 1, 1, 0, þ1, and 1, respectively, for
the normalized low-pass filter. Also, the transformation (5.55) transforms the fre-
b s1 and Ω
quencies Ω b s2 to Ω0 and Ω00 , respectively, where
s s
Ωb2 Ωb p1 Ωb p2
Ω0s ¼
s1 ¼ A1 ðsayÞ ð5:56Þ
b b
Ω p2 Ω p1 Ωb s1
and
Ωb2 Ωb p1 Ωb p2
Ω00s ¼
s2 ¼ A2 ðsayÞ ð5:57Þ
b p2 Ω
Ω b p1 Ωb s2
In order to satisfy the stopband requirements and to have symmetry of the

stopband edges in the low-pass filter, we choose Ωs to be themin{|A1|, |A2|}. Thus,
the spectral transformation (5.55) leads to the following design equations for the
normalized low-pass filter (except in the case of the Type 2 Chebyshev filter)
Ωp ¼ 1, Ωs ¼ minfjA1 j; jA2 jg ð5:58aÞ
where A1 and A2 are given

by (5.56) and (5.57), respectively, and the desired high-
pass transfer function H BP bs can be obtained from the normalized low-pass transfer
function HN (s) using (5.55). In the case of the Type 2 Chebyshev filter, the equation
corresponding to (5.58a) is
Ωp ¼ maxf1=jA1 j; 1=jA2 jg, Ωs ¼ 1 ð5:58bÞ
Example 5.7 Design a Butterworth IIR digital band-pass filter for the following
specifications:
Lower passband edge frequency: 41.4 Hz
Upper passband edge frequency: 50.95 Hz
Lower stopband edge frequency: 7.87 Hz
Upper stopband edge frequency: 100 Hz
Solution We have
ð0:0787Þ2 þ ð0, 414Þð0:5095Þ

A1 ¼ ¼ 27:25
0:0787ð0:5095 0, 414Þ
1 þ ð0, 414Þð0:5095Þ
A2 ¼ ¼ 8:26
1ð0:5095 0, 414Þ
For the prototype analog low-pass filter, Ωp ¼ 1, Ωs ¼ min {|A1|, |A2|} ¼ 8.26;
αp ¼ 2 dB αs ¼ 10 dB
h i
101 1
log10 100:2 :1
N¼ ¼ 0:5203
2log10 ð8:26Þ
Let us choose N ¼ 1
The transfer function of the first-order normalized Butterworth low-pass filter is
given by
1
H N ðsÞ ¼
sþ1
Substituting the values of Ωs and N in Eq. (5.20), we obtain

8:26 2
¼ 101 1
Ωc
Solving for Ωc, we get Ωc ¼ 2.7533

The analog transfer function of the low-pass filter can be obtained from the above
transfer function by substituting s ¼ Ωsc ¼ 2:7533
s
2:7533
H LP ðsÞ ¼
s þ 2:7533
To arrive at the analog transfer function of the band-pass filter, variable s in the
above normalized transfer function is to be replaced by

2
s þΩ b p2
b p1 Ω s2 þ 2109:3
S¼
¼
b p2 Ω
Ω b p1 s 9:55s
26:2943 s
H BP ðsÞ ¼ 2
s þ 26:2943s þ 2109:3
Example 5.8 Design a band-pass Butterworth filter for the specifications of Exam-
Solution The following MATLAB code fragments can be used to design HBS (s):
pffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffi
Bandwidth ¼ bw ¼ 50.95–41.4 ¼ 9.55; Ωo ¼ ð50:95Þð41:4Þ ¼ 45:9271.
[N,Wn]=buttord(1,8.26,2,10,‘s’);
[num, den]=1p2bp(B,A, 45.9271,9.55);% [num, den]=1p2bp(B,A, Ωo, bw);
The transfer function HLP(s) of the analog low-pass filter can be obtained by
displaying numerator and denominator coefficient vectors B and A. It is given by
2:7533
H LP ðsÞ ¼
s þ 2:7533
The transfer function HBP (s) of the analog band-pass filter can be obtained by
displaying numerator and denominator coefficient vectors num and den. It is given
by
26:2943 s
H BP ðsÞ ¼
s2 þ 26:2943s þ 2109:3
Low Pass to Band Stop

The prototype low-pass and the desired band-stop filters are shown in Figure 5.19. In
b p1 is the lower passband edge frequency, Ω
this figure, Ω b p2 the upper passband edge
b
frequency, Ω s1 the lower stopband edge frequency, and Ω b s2 the upper stopband
edge frequency of the desired band-stop filter. Let us now consider the
transformation
kbs
S¼
ð5:59Þ
bs 2 þ Ωb2
ms
where Ωb ms is the geometric mean between the stopband edge frequencies of the
band-stop filter, i.e.,
qffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffi
b ms ¼
Ω Ωb s1 Ω b s2 ð5:60Þ
As a consequence of this transformation, it is seen that Ω b ¼ 0 and 1

b
transformed to the frequency Ω ¼ 0 for the normalized low-pass filter. Now, we
b s1 to the stopband edge frequency Ωs
transform the lower stopband edge frequency Ω
of the normalized low-pass filter; hence,
k k
Ωs ¼ ¼ ð5:61aÞ
b s2 Ω
Ω b s1 Bs

where Bs ¼ Ω b s2 Ωb s1 is the bandwidth of the stopband. Also, the upper stopband
edge frequency Ωb s2 is transformed to
k k
¼ ¼ Ωs ð5:61bÞ
b b
Ω s2 Ω s1 Bs
Hence, the constant k is given by

k ¼ Bs Ωs ¼ Ωb s2 Ω
b s1 Ωs ð5:61cÞ
As a consequence, the passband edge frequencies b p1 and Ω

Ω b p2 are
transformed to

Ωb s2 Ωb s1 Ωb p1 1
Ω0p ¼ Ωs ¼ Ωs ð5:62aÞ
b s1 Ω
Ω b s2 Ωb2 A 1
p1
and

Ωb s2 Ωb s1 Ωb p2 1
Ω00p ¼ Ωs ¼ Ωs ð5:62bÞ
b s1 Ω
Ω b s2 Ωb 2 A2
p2
In order to satisfy the passband requirement as well as to satisfy the symmetry

requirement of the passband
edge of the normalized low-pass filter, we have to

choose the higher of Ω0p and Ω00p as Ωp. Since for the normalized filter (except for
the case of Type 2 Chebyshev filter), Ωp ¼ 1, we have to choose Ωs to be the lower of
{|A1|, |A2|}. Hence, the design equations for the normalized low-pass filter (except for
the Type 2 Chebyshev) (Figure 5.19) are
Ωp ¼ 1, Ωs ¼ minfjA1 j; jA2 jg ð5:63aÞ
where
Ωb s1 Ω
b s2 Ωb2 Ωb s1 Ω
b s2 Ωb2
p1 p2
A1 ¼
, A2 ¼
ð5:63bÞ
b s2 Ω
Ω b s1 Ωb p1 b s2 Ω
Ω b s1 Ωb p2
and the transfer function of the required band-stop filter is

H BS bs ¼ H N ðsÞc
^ ^
ð5:63cÞ
Ω s2 Ω s1 Ωs ^s
s¼ ^ Ω ^
^s 2 þΩ s1 s2
For the Type 2 Chebyshev filter, Eq. (5.63a) would be replaced by
Ωp ¼ maxf1=jA1 j; 1=jA2 jg, Ωs ¼ 1 ð5:63dÞ
Example 5.9 Design an analog band-stop Butterworth filter with the following
specifications:
Lower passband edge frequency: 22.35 Hz
Upper passband edge frequency: 447.37 Hz
Lower stopband edge frequency: 72.65 Hz
Upper stopband edge frequency: 137.64 Hz
Figure 5.19 Low-pass to

band-stop frequency
transformation.
(a) Prototype low-pass filter
frequency response.
(b) Band-stop filter
frequency response
(a)
(b)
Solution From Eq. (5.63b), we have
Ωb s1 Ωb s2 Ωb2 Ωb s1 Ωb s2 Ωb2
p1 p2
A1 ¼
¼ 6:5403, A2 ¼
¼ 6:5397
b s2 Ω
Ω b s1 Ωb p1 b s2 Ω
Ω b s1 Ωb p2
Now using (5.63a), we get the specifications for the normalized analog low-pass
filter to be
Ωp ¼ 1, Ωs ¼ minfjA1 j; jA2 jg, αp ¼ 3 dB, αs ¼ 15 dB


101:5 1
log 100:3 1
N ¼ 0:9125
2 logð6:5397Þ
We choose N ¼ 1. The transfer function of the first-order normalized Butterworth

low-pass filter is
1
H N ðsÞ ¼
ð s þ 1Þ
2
Substituting the values of Ωs and N in Eq. (5.20), we obtain 6:5397
Ωc ¼ 101:5 1:
Solving for Ωc, we get Ωc ¼ 1.1818. The analog transfer function of the low-pass
filter is obtained from HN(s) by substituting s ¼ Ωsc ¼ 1:1818
s
1:1818
H LP ðsÞ ¼
s þ 1:1818
To arrive at the analog transfer function of the band-stop filter, we use, in the
above expression, the low-pass to band-stop transformation given by (5.63c),
namely,

b s2 Ω
Ω b s1 Ωs s ð64:99Þð6:5397Þs 425s
S¼ ¼ ¼ 2
s2 þ Ω b s2
b s1 Ω s2 þ 10000 s þ 10000
to obtain
s2 þ 10000
H BS ðsÞ ¼
s2 þ 360s þ 10000
Example 5.10 Design a band-stop Butterworth filter for the specifications of Exam-
Solution The following MATLAB code fragments can be used to design HBS(s):
pffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffi
Bandwidth ¼ bw ¼ 447.37 – 22.35; Ωo ¼ ð137:64Þð72:65Þ ¼ 100.
[N,Wn]=buttord(1,6.5397,3,15,‘s’);
[num, den]=1p2bs(B,A,100,425.02);% [num, den]=1p2bs(B,A, Ωo, bw);
The transfer function HLP (s) of the analog low-pass filter can be obtained by
displaying numerator and denominator coefficient vectors B and A and is given by
1:1818
H LP ðsÞ ¼
s þ 1:1818
The transfer function HBS(s) of the analog band-stop filter can be obtained
by displaying numerator and denominator coefficient vectors num and den and is
given by
s2 þ 10000
H BS ðsÞ ¼
s2 þ 360s þ 10000
5.3 Effect of Poles and Zeros on Frequency Response
Frequency response of a system can be obtained by evaluating H(s) for all values of
s ¼ jΩ.
5.3.1 Effect of Two Complex System Poles on the Frequency

Response
Consider the following system function with complex poles
1
H ðsÞ ¼

ð5:64Þ
s ðα þ jΩÞ s ðα jΩÞ
with placement of poles as shown in Figure 5.20(a). The magnitude response of

the system with pole locations shown in Figure 5.20(a) is given by
1
jH ðjΩÞj ¼ ð5:65Þ
dd0
and is shown in Figure 5.20(b), and its phase response is shown in Figure 5.20(c)
5.3.2 Effect of Two Complex System Zeros on the Frequency

Response
Consider the following system function with complex zeros

HðsÞ ¼ s ðα þ jΩÞÞðs ðα jΩÞÞ ð5:66Þ
with placement of zeros as shown in Figure 5.21(a). The magnitude response of the
system with zeros locations shown in Figure 5.21(a) is given by
jH ðjΩÞj ¼ rr 0 ð5:67Þ
and is shown in Figure 5.21(b), and its phase response is shown in Figure 5.21(c)
5.4 Design of Specialized Analog Filters by Pole-Zero Placement 265
(a)
(b) (c)
Figure 5.20 (a) Pole locations of H(s). (b) Magnitude response. (c) Pole locations of H(s)
5.4 Design of Specialized Analog Filters by Pole-Zero

Placement
There are certain specialized filters often used in signal processing applications in
addition to the filters designed in the previous sections. These specialized filters can
be directly designed based on placement of poles and zeros.
(a)
(b) (c)
Figure 5.21 (a) Zero locations of H(s). (b) Magnitude response. (c) Phase response
5.4.1 Notch Filter
The notch filter removes a single frequency f0, called the notch frequency. The notch
filter can be realized with two zeros placed at jΩ0,
where Ω0 ¼ ð2πf 0 Þ
As such a filter does not have unity gain at zero frequency. The notch will not be
sharp. By placing two poles close to the two zeros on the semicircle as shown in
Figure 5.22(a), the notch can be made sharp with unity gain at zero frequency as
shown in Figure 5.22(b).
5.5 Problems 267
Figure 5.22 (a) Placing two poles close the two zeros on the semicircle. (b) Magnitude response
of (a)
Example 5.11 Design a second-order notch filter to suppress 50 Hz hum in an audio

signal
Solution Choose Ω0 ¼ 100π. Place zeros at s ¼ jΩ0 and poles at –Ω0 cos θ jΩ0
sin θ.
Then, the transfer function of the second-order notch filter is given by
ðs jΩ0 Þðs þ jΩ0 Þ

H ðsÞ ¼
ðs þ Ω0 cos θ þ jΩ0 sin θÞðs þ Ω0 cos θ jΩ0 sin θÞ
s 2 þ Ω0 2 s2 þ 98775:5102
¼ ¼ 2
s þ ð2Ω0 cos θÞs þ Ω0
2 2 s þ ð628:57 cos θÞs þ 98775:5102
5.5 Problems
1. Test the impulse response of an ideal low-pass filter for the following properties:
(i) Real valued
(ii) Even
(iii) Causal
2. Consider the first-order RC circuit shown in the figure below
(i) Determine H(Ω), the transfer function from vs to vc. Sketch the magnitude
and phase of H(Ω).
(ii) What is the cutoff frequency for H(Ω)?
(iii) Consider the following system:
(a) Draw the corresponding RC circuit and determine H(Ω), the transfer
function from v to vs. Sketch the magnitude and phase of H(Ω).
(b) What is the corresponding cutoff frequency?
3. Design a continuous time low-pass filter with the following transfer function
α
HðΩÞ ¼
α þ jΩ
with the following specifications
Find the range of values of α that meets the specifications.

4. Consider the following system
Further Reading 269
If H(Ω) is an ideal band-pass filter, determine for what values of α, it will act as
an ideal band-stop filter.
5. Design an elliptic analog high-pass filter for the specifications of Example 5.5
Further Reading
1. Raut, R., Swamy, M.N.S.: Modern Analog Filter Analysis and Design: a Practical Approach.
Springer, WILEY- VCH Verlag & Co. KGaA, Weinheim, Germany (2010)
2. Antoniou, A.: Digital Filters: Analysis and Design. McGraw Hill Book Co., New York (1979)
3. Parks, T.W., Burrus, C.S.: Digital Filter Design. Wiley, New York (1987)
4. Temes, G.C., Mitra, S.K. (eds.): Modern Filter Theory and Design. Wiley, New York (1973)
5. Vlach, J.: Computerized Approximation and Synthesis of Linear Networks. Wiley, New York
(1969)
6. Mitra, S.K.: Digital Signal Processing. McGraw-Hill, New York (2006)
7. Chen, C.T.: Digital Signal Processing, Spectral Computation and Filter Design. Oxford Univer-
sity Press, NewYork/Oxford, UK (2001)
Chapter 6
Discrete-Time Signals and Systems
Discrete-time signals are obtained by the sampling of continuous-time signals.

Digital signal processing deals basically with discrete-time signals, which are
processed by discrete-time systems. The characterization of discrete-time signals
as well as discrete-time systems in time domain is required to understand the theory
of digital signal processing. In this chapter, time-domain sampling and the funda-
mental concepts of discrete-time signals as well as discrete-time systems are con-
sidered. First, the sampling process of analog signals is described. Next, the basic
sequences of discrete-time systems and their classification are emphasized. The
input-output characterization of linear time-invariant (LTI) systems by means of
convolution sum is described. Further, sampling of discrete-time signals is intro-
duced. Finally, the state-space representation of discrete-time LTI systems is
described.
6.1 The Sampling Process of Analog Signals
6.1.1 Impulse-Train Sampling
The acquisition of an analog signal at discrete-time intervals is called sampling. The

sampling process mathematically can be treated as a multiplication of a continuous-
time signal x(t) by a periodic impulse train p(t) of unit amplitude with period T. For
example, consider an analog signal xa(t) as shown in Figure 6.1(a), and a periodic
pulse train p(t) of unit amplitude with period T as in Figure 6.1 (b) is referred to as the
sampling function, the period T as the sampling period, and the fundamental
frequency ωT ¼ (2π/T ) as the sampling frequency in radians. Then, the sampled
version xp(t) is shown in Figure 6.1 (c).

272 6 Discrete-Time Signals and Systems
xa (t ) p(t)
....
t
t
0 T 2T
0
(b)
(a)
xp(t)
....
0 T 2T
t
(c)
Figure 6.1 (a) Continuous-time signal, (b) pulse train, (c) sampled version of (b)
In the time domain, we have
xp ðt Þ ¼ xa ðt Þpðt Þ ð6:1Þ
where
X1
pð t Þ ¼ n¼1
δðt nT Þ ð6:1aÞ
xp(t) is the impulse train with the amplitudes of the impulses equal to the samples of
xa(t) at intervals T, 2T, 3T, . . . .
Therefore, the sampled version of signal xp(t) mathematically can be represented as
X1
xp ð t Þ ¼ x ðnT Þδðt
n¼1 a
nT Þ ð6:2Þ
6.1.2 Sampling with a Zero-Order Hold
In Section 6.1.1, the sampling process establishes a fact that the band-limited signal
can be uniquely represented by its samples. In a practical setting, it is difficult to
generate and transmit narrow large amplitude pulses that approximate impulses.
6.1 The Sampling Process of Analog Signals 273
Lowpass filter Sample and Quantizer Encoder

hold
x (t) 2b • Logic x (n)

• circuit
Analog •
input Digital
FT 1 output
code
Figure 6.2 A block diagram representation of an analog-to-digital conversion process.
Hence, it is more convenient to implement the sampling process using a zero-order

hold. It samples analog signal at a given sampling instant and holds the sample value
until the succeeding sampling instant. A block diagram representation of the analog-
to-digital conversion (ADC) process is shown in Figure 6.2. The amplitude of each
signal sample is quantized into one of the 2b levels, where b is the number of bits
used to represent a sample in the ADC. The discrete amplitude levels are encoded
into distinct binary word of length b bits.
A sequence of samples x(n) is obtained from an analog signal xa(t) according to
the relation
xðnÞ ¼ xa ðnT Þ 1 < n < 1: ð6:3Þ
In Eq. (6.2), T is the sampling period, and its reciprocal, FT ¼ 1/T is called the
sampling frequency, in samples per second. The sampling frequency FT is also
referred to as the Nyquist frequency.
Sampling Theorem
The sampling theorem states that an analog signal must be sampled at a rate at least
twice as large as highest frequency of the analog signal to be sampled. This means
that
F T 2f max ð6:4Þ
where fmax is maximum frequency component of the analog signal. The frequency
2fmax is called the Nyquist rate.
For example, to sample a speech signal containing up to 3 kHz frequencies, the
required minimum sampling rate is 6 kHz, that is, 6000 sample per second. To
sample an audio signal having frequencies up to 22 kHz, the required minimum
sampling rate is 44 kHz, that is, 44000 samples per second.
A signal whose energy is concentrated in a frequency band range fL < |f| < fH is
often referred to as a band-pass signal. The sampling process of such signals is
generally referred to as band-pass sampling. In the band-pass sampling process, to
prevent aliasing effect, the band-pass continuous-time signal can be sampled at
sampling rate greater than twice the highest frequency ( fH):
F T 2f H ð6:5Þ
The bandwidth of the band-pass signal is defined as
Δf ¼ f H f L ð6:6Þ
Consider that the highest frequency contained in the signal is an integer multiple
of the bandwidth that is given as
f H ¼ cðΔf Þ ð6:7Þ
The sampling frequency is to be selected to satisfy the condition as
fH
F T ¼ 2ðΔf Þ ¼ ð6:8Þ
c
6.1.3 Quantization and Coding
Quantization and coding are two primary steps involve in the process of A/D
conversion. Quantization is a nonlinear and non-invertible process that rounds the
given amplitude x(n) ¼ x(nT) to an amplitude xk that is taken from the finite set of
values at time t ¼ nT. Mathematically, the output of the quantizer is defined as
xq ðnÞ ¼ Q½xðnÞ ¼ b
xk ð6:9Þ
The procedure of the quantization process is depicted as
x 1 x1 x 2 x 2 x 3 x3 x 4 x4 x 5 x 5 ………………
The possible outputs of the quantizer (i.e., the quantization levels) are indicated
by b
x1 b x2 bx3 b x4 b x L where L stands for number of intervals into which the
signal amplitude is divided. For uniform quantization,
b
x kþ1 bxk ¼ Δ k ¼ 1, 2, , L:
xkþ1 xk ¼ Δ for finite xk , xkþ1 : ð6:10Þ
where Δ is the quantizer step size.

The coding process in an ADC assigns a unique binary number to each quanti-
zation level. For L levels, at least L different binary numbers are needed. With word
length of n bits, 2n distinct binary numbers can be represented. Then, the step size or
the resolution of the A/D converter is given by
A
Δ¼ ð6:11Þ
2n
where A is the range of the quantizer.
6.1 The Sampling Process of Analog Signals 275
(a) (b)
(c)
Figure 6.3 (a) Quantizer, (b) mathematical model, (c) power spectral density of quantization
noise
Quantization Error
Consider an n bit ADC sampling analog signal x(t) at sampling frequency of FTas
shown in Figure 6.3(a). The mathematical model of the quantizer is shown in
Figure 6.3(b). The power spectral density of the quantization noise with an assump-
tion of uniform probability distribution is shown in Figure 6.3(c).
If the quantization error is uniformly distributed in the range (‐Δ/2, Δ/2) as
shown in Figure 6.3(b), the mean value of the error is zero, and the variance (the
quantization noise power) σ 2e is given by
ð Δ=2
Δ2
Pqn ¼ σ 2e ¼ qe 2 ðnÞPðeÞde ¼ ð6:12Þ
Δ=2 12
The quantization noise power can be expressed by
quantization step2 A2 1 A2
σ 2e ¼ ¼ 2n ¼ 22n ð6:13Þ
12 12 2 12
The effect of the additive quantization noise on the desired signal can be
quantified by evaluating the signal-to-quantization noise (power) ratio (SQNR)
that is defined as
Px
SQNR ¼ 10log10 ð6:14Þ
Pqn
h i
where Px ¼ σ 2x ¼ E x2 ðnÞ is the signal power and Pqn ¼ σ 2e ¼ E e2q ðnÞ is the
quantization noise power.
6.2 Classification of Discrete-Time Signals
6.2.1 Symmetric and Anti-symmetric Signals
A real valued signal x(n) is said to be symmetric if it satisfies the condition
xðnÞ ¼ xðnÞ ð6:15aÞ
Example of a symmetric sequence is shown in Figure 6.4

On the other hand, a signal x(n) is called anti-symmetric if it follows the condition
xðnÞ ¼ xðnÞ ð6:15bÞ
An example of anti-symmetric sequence is shown in Figure 6.5.
6.2.2 Finite and Infinite Length Sequences
A signal is said to be of finite length or duration if it is defined only for a finite time
interval:
· · ·
··· · · · · · · ···
-1 0 1
· · · ·
· ·
Figure 6.4 An example of symmetric sequence
6.2 Classification of Discrete-Time Signals 277
· · ·
··· · · · · ·
1
-1 0
· · · · · ···
· ·
Figure 6.5 An example of anti-symmetric sequence
1 < N 1 n N 2 < 1 ð6:16Þ
The length of the sequence is N ¼ N2 N1 þ 1. Thus, a finite sequence of length

N has N samples. A discrete-time sequence consisting of N samples is called a N-
point sequence. Any finite sequence can be viewed as an infinite length sequence by
adding zero-valued samples outside the range (N1, N2). Also, an infinite length
sequence can be truncated to produce a finite length sequence.
6.2.3 Right-Sided and Left-Sided Sequences
A right-sided sequence is an infinite sequence x(n) for which x(n) ¼ 0 for n < N1,
where N1 is a positive or negative integer. If N1 0, the right-sided sequence is said
to be causal. Similarly, if x(n) ¼ 0 for n > N2, where N2 is a positive or negative
integer, then the sequence is called a left-sided sequence. Also, if N2 0, then the
sequence is said to be anti-causal.
6.2.4 Periodic and Aperiodic Signals
A sequence x(n) ¼ x(n + N ) for all n is periodic with a period N, where N is a positive
integer. The smallest value of N for which x(n) ¼ x(n + N ) is referred as the
fundamental period. A sequence is called aperiodic, if it is not periodic. An example
of a periodic sequence is shown in Figure 6.6.
Proposition 6.1 A discrete-time sinusoidal sequence x(n) ¼ A sin (ω0n + θ) is
periodic if and only if ω2π0 is a rational number.
The rational number is defined as the ratio of two integers. For the given periodic
signal x(n) ¼ A sin (ω0n þ θ), its fundamental period N is obtained from the
following relationship
· · ·
· · ·
··· ···
· · ·
· · · · · · · · ·
-5 -4 -3 -2 -1 0 1 2 -3 4 5 6 7 8 -9 10 11 12
Figure 6.6 An example of a periodic sequence
ω0 m
¼
2π N
2π
N¼ m
ω0
The fundamental period of a discrete-time sinusoidal sequence satisfying the
proposition 6.1 is calculated by setting m equal to a small integer that results in an
integer value for N.
The fundamental period of a discrete-time complex exponential sequence can
also be calculated satisfying the proposition 6.1.
Example 6.1 Determine if the discrete-time sequences are periodic:

(i) xðnÞ ¼ cos πn 4
(ii) x(n) ¼ sin2n

(iii) xðnÞ ¼ sin πn 4 þ cos 2n:
(iv) xðnÞ ¼ e 8 Þ
jð5πn þθ
Solution
(i) The value of ω0 in x(n) is π4. Since ω2π0 ¼ 18 is a rational number, it is periodic
discrete-time sequence. The fundamental period of x(n) is given by N ¼ ω2π0 m:

For m ¼ 1, N ¼ 2π π4 ¼ 8. Hence, xðnÞ ¼ cos πn 4 is periodic with fundamental
period N ¼ 8,
(ii) x(n) ¼ sin 2n is aperiodic because ω0N ¼ 2N ¼ 2πm is not satisfied for any
integer
πnvalue of m in making N to be an integer.
(iii) sin 4 is periodic and cos 2n is aperiodic. Since the sum of periodic and

aperiodic signals is aperiodic, the signal xðnÞ ¼ sin πn4 þ cos 2n is aperiodic.
ω0
(iv) The value of ω0 in x(n) is 8 : Since 2π ¼ 16 is a rational number, it is periodic
5π 5
discrete-time sequence. The fundamental period of x(n) is given by N ¼ ω2π0 m:

For m ¼ 5, N ¼ 8 2π 5 ¼ 16: Hence, xðnÞ ¼ ejð 8 þθÞ is periodic with funda-
5πn
5π
mental period N ¼ 16.
6.2 Classification of Discrete-Time Signals 279
6.2.5 Energy and Power Signals
The total energy of a signal x(n), real or complex, is defined as
X
1
E¼ jxðnÞj2 ð6:17Þ
n¼1
By definition, the average power of an aperiodic signal x(n) is given by
1 XN
P ¼ Lt jxðnÞj2 ð6:18aÞ
N!1 2N þ 1
n¼N
The signal is referred to as an energy signal if the total energy of the signal
satisfies the condition 0 < E < 1. It is clear that for a finite energy signal, the average
power P is zero. Hence, an energy signal has zero average power. On the other hand,
if E is infinite, then P may be finite or infinite. If P is finite and nonzero, then the
signal is called a power signal. Thus, a power signal is an infinite energy signal with
finite average power.
The average power of a periodic sequence x(n) with a period I is given by
1XI1
P¼ jxðnÞj2 ð6:18bÞ
I n¼0
Hence, periodic signals are power signals.

Example 6.2 Determine whether the sequence x(n) ¼ anu(n) is an energy signal or a
power signal or neither for the following cases:
ðaÞjaj < 1, ðbÞjaj ¼ 1, ðcÞjaj > 1:
Solution For x(n) ¼ anu(n), E is given by

X1 X1 2
E¼ xðnÞ2 ¼ an
1 0
1 X1 XN
P ¼ limN!1 xðnÞ2 ¼ limN!1 1 a2n
2N þ 1 1 2N þ 1 0
(a) For |a| < 1,
X1 X1 1
E¼ xðnÞ2 ¼ jan j2 ¼ is finite
1 0 1 j aj 2
1 XN 1 jaj2ðNþ1Þ
P ¼ limN!1 a2n ¼ limN!1 1 ¼0
2N þ 1 n¼0 2N þ 1 1 jaj2
The energy E is finite and the average power P is zero. Hence, the signal x(n) ¼ an
u(n) is an energy signal for |a| < 1.
(b) For |a| ¼ 1,
P
E¼ 1 n 2
0 ja j ! 1
1 X N 2n N þ1 1
P ¼ limN!1 a ¼ limn!1 ¼
2N þ 1 n¼0 2N þ 1 2
The energy E is infinite, and the average power P is finite. Hence, the signal x
(n) ¼ anu(n) is a power signal for |a| ¼ 1.
(c) For |a| > 1,
X1
E¼ 0
jan j2 ! 1
1 XN jaj2ðNþ1Þ 1
P ¼ limN!1 a2n ¼ limN!1 1 !1
2N þ 1 n¼0 2N þ 1 jaj2 1
The energy E is infinite and also the average power P is infinite. Hence, the signal
x(n) ¼ anu(n) is neither an energy signal nor a power signal for |a| > 1.
Example 6.3 Determine whether the following sequences
(i) x(n) ¼ e–nu(n),
(ii) x(n) ¼ enu(n),
(iii) x(n) ¼ nu(n), and
(iv) x(n) ¼ cosπn u(n)
are energy or power signals or neither energy nor power signals.
Solution
(i) x(n) ¼ enu(n). Hence, E and P are given by
X1 X1 1
E¼ j x ð n Þj 2
¼ e2n ¼ is finite
1 0 1 e2
1 XN
P ¼ limN!1 j x ð nÞ j 2
2N þ 1 n¼0
1 X N 2n
¼ limN!1 e
2N þ 1 n¼0
1 1 e2ðNþ1Þ
¼ limN!1 ¼0
2N þ 1 1 e2
The energy E is finite and the average power P is zero. Hence, the signal x(n) ¼
enu(n) is an energy signal.
6.3 Discrete-Time Systems 281
(ii) x(n) ¼ e+nu(n). Therefore, E and P are given by

P1 P1
E¼ 1 j x ð nÞ j 2 ¼ 0 e2n ! 1
1 X N
1 X M
1 e2ðNþ1Þ 1
P ¼ lim jxðnÞj2 ¼ lim e2n ¼ lim !1
N!1 2N þ 1 N!1 2N þ 1 n!1 2N þ 1 e2 1
n¼0 n¼0
x(n) ¼ enu(n) is neither an energy signal nor a power signal.
(iii) x(n) ¼ nu(n). Hence, E and P are given by
P1 P1
E¼ 1 jxðnÞj2 ¼ 0 n2 ! 1
1 X1
P ¼ limN!1 jxðnÞj2
2N þ 1 1
1 XN 2 NðN þ 1Þð2N þ 1Þ
¼ limN!1 n ¼ limN!1 !1
2N þ 1 n¼0 6ð2N þ 1Þ
x(n) ¼ nu(n) is neither an energy signal nor a power signal.
(iv) x(n) ¼ cos πn u(n). Sincecosπn ¼ (1)n, E and P are given by
P1 P1 P1
E¼ 1 jxðnÞj2 ¼ 0 j cos πnj2 ¼ 0 ð1Þ2n ! 1
1 X1
P ¼ limN!1 jxðnÞj2
2N þ 1 1
1 XN Nþ1 1
¼ limN!1 ð1Þ2n ¼ limN!1 ¼
2N þ 1 n¼0 2N þ 1 2
The energy E is not finite and the average power P is finite. Hence, the signal
x(n) ¼ cos πnu(n) is a power signal.
6.3 Discrete-Time Systems
A discrete-time system is defined mathematically as a transformation that maps an

input sequence x(n) into an output sequence y(n). This can be denoted as
yðnÞ ¼ ℜ½xðnÞ ð6:19Þ
where ℜ is an operator.
6.3.1 Classification of Discrete-Time Systems
Linear Systems
A system is said to be linear if and only if it satisfies the following conditions:
ℜ½axðnÞ ¼ aℜ½xðnÞ ð6:20Þ

ℜ½x1 ðnÞ þ x2 ðnÞ ¼ ℜ½x1 ðnÞ þ ℜ½x2 ðnÞ ¼ y1 ðnÞ þ y2 ðnÞ ð6:21Þ
where a is an arbitrary constant and y1(n) and y2(n) are the responses of the system
when x1(n) and x2(n) are the respective inputs. Equations (6.20) and (6.21) represent
the homogeneity and additivity properties, respectively.
The above two conditions can be combined into one representing the principle of
superposition as
ℜ½ax1 ðnÞ þ bx2 ðnÞ ¼ aℜ½x1 ðnÞ þ bℜ½x2 ðnÞ ð6:22Þ
where a and b are arbitrary constants.

Example 6.4 Check for linearity of the following systems described by the follow-
ing input-output relationships:
Xn
(i) yðnÞ ¼ k¼1
xðk Þ
(ii) y(n) ¼ x (n)
2
(iii) y(n) ¼ x(n n0), where n0 is an integer constant

Solution (i) The outputs y1(n) and y2(n) for inputs x1(n) and x2(n) are, respectively,
given by
X
n
y1 ðnÞ ¼ x1 ðkÞ
k¼1
Xn
y2 ðnÞ ¼ x2 ðkÞ
k¼1
The output y(n) due to an input x(n) ¼ ax1(n) þ bx2(n) is then given by
X
n X
n X
n
y ð nÞ ¼ ax1 ðk Þ þ bx2 ðkÞ ¼ a x1 ð k Þ þ b x2 ð k Þ
k¼1 k¼1 k¼1
¼ ay1 ðnÞ þ by2 ðnÞ

Xn
Hence the system described by yðnÞ ¼ k¼1
xðkÞ is a linear system.
(ii) The outputs y1(n) and y2(n) for inputs x1(n) and x2(n) are given by
y1 ðnÞ ¼ x21 ðnÞ

y2 ðnÞ ¼ x22 ðnÞ
yðnÞ ¼ ðax1 ðnÞ þ bx2 ðnÞÞ2 ¼ a2 x21 ðnÞ þ 2abx1 ðnÞx2 ðnÞ þ b2 x22 ðnÞ
ay1 ðnÞ þ by2 ðnÞ ¼ ax21 ðnÞ þ bx22 ðnÞ 6¼ yðnÞ
Therefore, the system y(n) ¼ x2(n) is not linear.

(iii) The outputs y1(n) and y2(n) for inputs x1(n) and x2(n), respectively, are given by
y 1 ð nÞ ¼ x 1 ð n n0 Þ
y 2 ð nÞ ¼ x 2 ð n n0 Þ
yðnÞ ¼ ax1 ðn n0 Þ þ bx2 ðn n0 Þ ¼ ay1 ðnÞ þ by2 ðnÞ
Hence, the system y(n) ¼ x(n n0) is linear.

Time-Invariant Systems
A time-invariant system (shift invariant system) is one in which the internal param-
eters do not vary with time. If y1(n) is output to an input x1(n), then the system is said
to be time invariant if, for all n0, the input sequence x1(n) ¼ x(n n0) produces the
output sequence y1(n) ¼ y(n n0), i.e.,
ℜ½xðn n0 Þ ¼ yðn n0 Þ
where n0 is a positive or negative integer.

Example
P1 6.5 Check for time-invariance of the system defined by y ð nÞ ¼
k¼1 xðk Þ
Solution From given Eq., the output y(n) of the system delayed by n0 can be written
as
X0
nn
y ð n n0 Þ ¼ xð k Þ
k¼1
For example, for an input x1(n) ¼ x(n n0), the output y1(n) can be written as
X
n
y 1 ð nÞ ¼ x ð k n0 Þ
k¼1
Substitution of the change of variables k1 ¼ k n0 in the above summation yields

X0
nn
y 1 ð nÞ ¼ x ð k 1 Þ ¼ y ð n n0 Þ
k1 ¼1

Example 6.6 Check for time-invariance of the down-sampling system with a factor
of 2, defined by the relation
yðnÞ ¼ xð2nÞ 1 < n < 1
Solution For an inputx1(n) ¼ x(n n0), the output y1(n) of the compressor system
can be written as
y1 ðnÞ ¼ xð2n n0 Þ
From given equation,

y ð n n0 Þ ¼ x ð 2ð n n0 Þ Þ
Comparing the above two equations, it can be observed that y1(n) 6¼ y(n n0).
Thus, the down-sampling system is not time invariant.
Causal System
A system is said to be causal if its output at time instant n depends only on the
present and past input values, but not on the future input values.
For example, a system defined by
yðnÞ ¼ xðn þ 2Þ xðn þ 1Þ
is not causal, as the output at time instant n depends on future values of the input.
But, the system defined by
y ð nÞ ¼ x ð nÞ x ð n 1Þ
is causal, since its output at time instant n depends only on the present and past
values of the input.
Stable System
A system is said to be stable if and only if every bounded-input sequence produces a
bounded-output sequence. The input x(n) is bounded if there exists a fixed positive
finite value βx such that
jxðnÞj βx < 1 for all n ð6:23Þ
Similarly, the output y(n) is bounded if there exists a fixed positive finite value βy
such that
jyðnÞj βy < 1 for all n ð6:24Þ
and this type of stability is called bounded-input bounded-output (BIBO) stability.

Example 6.7 Check for stability of the system described by the following input-
output relation
y ð nÞ ¼ x 2 ð nÞ
Solution Assume that the input x(n) is bounded such that |x(n)| βx < 1 for all n
Then, jyðnÞj ¼ jxðnÞj2 β2x < 1
Hence, y(n) is bounded and the system is stable.

Example 6.8 Check for stability, causality, linearity, and time-invariance of the
system described by ℜ[x(n)] ¼ (1)nx(n)
This transformation outputs the current value of x(n) multiplied by either 1.
It is stable, since it does not change the magnitude of x(n) and hence satisfies the
conditions for bounded-input bounded-output stability.
It is causal, because each output depends only on the current value of x(n).
Let y1 ðnÞ ¼ ℜ½x1 ðnÞ ¼ ð1Þn x1 ðnÞ

y2 ðnÞ ¼ ℜ½x2 ðnÞ ¼ ð1Þn x2 ðnÞ
Then, ℜ½ax1 ðnÞ þ bx2 ðnÞ ¼ ð1Þn ax1 ðnÞ þ ð1Þn bx2 ðnÞ ¼ ay1 ðnÞ þ by2 ðnÞ
Hence, it is linear.
yðnÞ ¼ ℜ½xðnÞ ¼ ð1Þn xðnÞ ℜ½xðn 1Þ ¼ ð1Þn xðn 1Þ

ℜ½xðn 1Þ 6¼ yðn 1Þ
Therefore, it is not time invariant.

Example 6.9 Check for stability, causality, linearity, and time-invariance of the
system described by ℜ[x(n)] ¼ x(n2)
Solution Stable, since if x(n) is bounded, x(n2) is also bounded.
It is not causal, since, for example, if n ¼ 4, then the output y(n) depends upon the
future input because y(4) ¼ ℜ[x(4)] ¼ x(16)
y1 ¼ ℜ½x1 ðnÞ ¼ x1 ðn2 Þ; y2 ðnÞ ¼ ℜ½x2 ðnÞ ¼ x2 ðn2 Þ;

ℜ½ax1 ðnÞ þ bx2 ðnÞ ¼ ax1 ðn2 Þ þ bx2 ðn2 Þ
¼ ay1 ðnÞ þ by2 ðnÞ
Therefore, it is linear.
yðnÞ ¼ ℜ½xðnÞ ¼ xðn2 Þ
ℜ½xðn 1Þ 6¼ yðn 1Þ
Hence, it is not time invariant.

6.3.2 Impulse and Step Responses
Let the input signal x(n) be transformed by the system to generate the output signal
y(n). This transformation operation is given by
y ð nÞ ¼ ℜ½ x ð nÞ ð6:25Þ
If the input to the system is a unit sample sequence (i.e., impulse input δ(n)), then
the system output is called as impulse response and denoted by h(n). If the input to
the system is a unit step sequence u(n), then the system output is called its step
response. In the next section, we show that a linear time-invariant discrete-time
system is characterized by its impulse response or step response.
6.4 Linear Time-Invariant Discrete-Time Systems
Linear time-invariant systems have significant signal processing applications, and

hence it is of interest to study the properties of such systems.
6.4.1 Input-Output Relationship
An arbitrary sequence x(n) can be expressed as a weighted linear combination of unit

sample sequences given by
X1
x ð nÞ ¼ k¼1
xðkÞδðn k Þ ð6:26Þ
Now, the discrete-time system response y(n) is given by

hX1 i
y ð nÞ ¼ ℜ½ x ð nÞ ¼ ℜ k¼1
xðkÞδðn k Þ ð6:27Þ
From the principle of superposition, the above equation can be written as

X1
yðnÞ ¼ k¼1
xðkÞℜ½δðn kÞ ð6:28Þ
Let the response of the system due to input δ(n k) be hk(n), that is,
hk ðnÞ ¼ ℜ½δðn kÞ
Then, the system response y(n) for an arbitrary input x(n) is given by
X1
yðnÞ ¼ k¼1
xðkÞhk ðnÞ
6.4 Linear Time-Invariant Discrete-Time Systems 287
Since δ(n k) is a time-shifted version of δ(n), the response hk(n) is the time-
shifted version of the impulse response h(n), since the operator is time invariant.
Hence,hk(n) ¼ h(n ‐ k). Thus,
X1
y ð nÞ ¼ k¼1
xðk Þhðn kÞ ð6:29Þ
The above equation for y(n) is commonly called the convolution sum and
represented by
yðnÞ ¼ xðnÞ∗hðnÞ ð6:29aÞ
where the symbol * stands for convolution. The discrete-time convolution operates
on the two sequences x(n) and h(n) to produce the third sequence y(n).
Example 6.10 Determine discrete convolution of the following sequences for large
value of n:
1n
hðnÞ ¼ 5 uð nÞ
xðnÞ ¼ ð1Þn uðnÞ
yðnÞ ¼ xðnÞ∗hðnÞ
P
Solution ¼ 1 k¼1 xðk Þhðn k Þ
X 1k
1 Xn k
1
¼ uðk Þð1Þnk uðn kÞ ¼ ð1Þn ð1Þk
k¼1
5 k¼0
5
1nþ1
X 1 5
n
1 k
¼ ð1Þn ¼ ð1Þn
5 1
k¼0 1
5
1nþ1
1 5
¼ ð1Þn
1
1þ
5
1nþ1
For large n, 5 tends to zero and hence,
1
yðnÞ ¼ ð1Þn
1:2
Example 6.11 Determine discrete convolution of the following two finite duration
sequences:
n
1
hð nÞ ¼ uð nÞ
3
n
1
x ð nÞ ¼ uð nÞ
5
Solution The impulse response h(n) ¼ 0 for n < 0; hence the given system is causal;
and x(n) ¼ 0 for n < 0, therefore the sequence x(n) is causal sequence:
P n 1k 1nk 1n P n 3k
yðnÞ ¼ xðnÞ∗hðnÞ ¼ k¼0 5 3 ¼ 3 k¼0 5
n 1 ð3=5Þnþ1
¼ 13
1 ð3=5Þ
6.4.2 Computation of Linear Convolution
Matrix Method
If the input x(n) is of length N1 and the impulse sequence h(n) is of length N2, then
the convolution sequence is of length N1 + N2 1. Thus, the linear convolution
given by Eq. (6.29) can be written in matrix form as
2 3 2 3
yð0Þ xð0Þ 0 0 0
6 7 6 7
6 yð1Þ 7 6 xð1Þ xð0Þ 0 0 7
6 7 6 7
6 7 6 7
6 yð2Þ 7 6 xð2Þ xð1Þ xð0Þ 0 7
6 7 6 7
6 7 6 7
6 yð3Þ 7 6 ⋮ xð2Þ xð1Þ ⋮ 7
6 7 6 7
6 7 6 7
6 ⋮ 7 ¼ 6 xðN 1 1Þ ⋮ xð2Þ ⋮ 7
6 7 6 7
6 7 6 7
6 yðN 1 1Þ 7 6 0 xðN 1 1Þ ⋮ ⋮ 7
6 7 6 7
6 7 6 7
6 yðN Þ 7 6 0 0 xðN 1Þ ⋮ 7
6 1 7 6 1 7
6 7 6 7
6 ⋮ 7 6 ⋮ ⋮ ⋮ ⋮ ⋮ 7
4 5 4 5
yðN 1 þ N 2 2Þ 0 0 0 xð0Þ
2 3
hð0Þ
6 7
6 hð1Þ 7
6 7
6 7
6 hð2Þ 7
6 7
6 7
6 hð3Þ 7
6 7
6 7
66 ⋮ 7
7
6 7
6 hðN 2 1Þ 7
6 7
6 7
6 0 7
6 7
6 7
6 ⋮ 7
4 5
0
ð6:30Þ
The following example illustrates the above procedure for computation of linear
convolution.
Example 6.12 Find the convolution of the sequences x(n) ¼ {6, 3} and
h(n) ¼ {3, 6, 3}.
Solution Using Eq. (6.19), the linear convolution of x(n) and h(n) is given by
2 3 2 32 3 2 3
yð0Þ 6 0 0 0 3 18
6 7 6 76 7 6 7
6 yð1Þ 7 6 3 6 0 0 7 6 7 6 7
6 7 6 76 6 7 6 45 7
6 7¼6 76 7¼6 7
6 yð2Þ 7 6 0 3 6 0 7 6 7 6 7
4 5 4 54 3 5 4 0 5
yð3Þ 0 0 3 6 0 9
Thus,
yðnÞ ¼ xðnÞ∗hðnÞ ¼ f18; 45; 0; 9g
Graphical Method for Computation of Linear Convolution

Evaluation of sum at any sample n consists of the following four important
operations:
(i) Time reversing or reflecting of the sequence h(k) about k ¼ 0 sample to give
h(–k).
(ii) Shifting the sequence h(–k) to the right by n samples to obtain h(n – k).
(iii) Forming the product x(k)h(n – k) sample by sample for the desired value of n.
(iv) Summing the product over the index k in y(n) for the desired value of n.
The length of the convolution sum sequence y(n) is given by n ¼ N1 + N2 1,
where N1 is length of the sequence x(n) and N2 is length of the sequence h(n).
Example 6.13 Compute the convolution of the sequences of Example 6.12 using the
graphical method.
Solution The sequences x(n) and h(n) are shown in Figures 6.7.
6 h (n ) 6
x ( n)
3
0
n n
0 1 1 2
-3
-3
Figure 6.7 Sequences x(n) and h(n)

Figure 6.8 Convolution of

sequences x(n) and h(n)
Figure 6.9 Sequence generated by the convolution
6.4.3 Computation of Convolution Sum Using MATLAB
The MATLAB function conv(a,b) can be used to compute convolution sum of two
sequences a and b as illustrated in the following example.
Example 6.14 Compute convolution sum of the sequences x(n) ¼ {2,1,0,0} and
h(n) ¼ {1,2,1}, using MATLAB.
Program 6.1. Illustration of convolution
a=[ 2 -1 0 0 ];% first sequence

b=[-1 2 1];% second sequence
c=conv(a,b);% convolution of first sequence and second sequence
len=length(c)-1;
n=0:1:len;
stem(n,c)
xlabel(‘Time index n’); ylabel(‘Amplitude’);
axis([0 5 -3 5])
6.4.4 Some Properties of the Convolution Sum
Starting with the convolution sum given by (6.30), namely, y(n) ¼ x(n) * h(n), we
can establish the following properties:
1) The convolution sum obeys the commutative law
xðnÞ∗hðnÞ ¼ hðnÞ∗xðnÞ ð6:31aÞ
2) The convolution sum obeys the associative law
ðxðnÞ∗h1 ðnÞÞ∗h2 ðnÞ ¼ xðnÞ∗ðh1 ðnÞ∗h2 ðnÞÞ ð6:31bÞ
3) The convolution sum obeys the distributive law
xðnÞ∗ðh1 ðnÞ þ h2 ðnÞÞ ¼ xðnÞ∗h1 ðnÞ þ xðnÞ∗h2 ðnÞ ð6:31cÞ
Let us now interpret the above relations physically.

1) The commutative law shows that the output is the same if we interchange the
roles of the input and the impulse response. This is illustrated in Figure 6.10.
2) To interpret the associative law, we consider a cascade of two systems whose
impulse responses are h1(n) and h2(n). Then y1(n) ¼ x(n) * h(n) if x(n) is the input
to the system with the impulse response h1(n). If y1(n) is now fed as the input to the
system with impulse response h2(n), then the overall system output is given by
yðnÞ ¼ y1 ðnÞ∗h2 ðnÞ ¼ ½xðnÞ∗h1 ðnÞ∗h2 ðnÞ

¼ xðnÞ∗½h1 ðnÞ∗h2 ðnÞ,
by associative law
¼ xðnÞ∗hðnÞ ð6:32Þ
This equivalence is shown in Figure 6.11. Hence, if two systems with impulse
responses h1(n) and h2(n) are cascaded, then the overall system response is given
by
Figure 6.10 Interpretation

of the commutative law
y1 ( n) y (n )
x( n)
x( n) y(n)
h1 (n) h2( n) ≡ h(n) = h1(n)* h2 (n)
Figure 6.11 Interpretation of the associative law

y1 ( n)
h1( n)
y(n) x( n) y(n)
x( n)
≡ h1(n) + h2 (n)
h2 ( n)
y2 ( n)
Figure 6.12 Interpretation of distributive law
Figure 6.13 Input-output x ( n) y (n )

relations for Example 6.15
h (n)
y(- n)
y1 ( n)
h (n)
x (n) y2 ( n)
h1 (n)
hðnÞ ¼ h1 ðnÞ∗h2 ðnÞ ð6:33Þ
This can be generalized to a number of LTI systems in cascade.

3) We now consider the distributive law given by (6.31c). This can be easily
interpreted as two LTI systems in parallel and that the overall system impulse
response h(n) of the two systems in parallel is given by
h ð n Þ ¼ h1 ð nÞ þ h 2 ð nÞ ð6:34Þ
This is illustrated in Figure 6.12.

Example 6.15 Consider the system shown in Figure 6.13 with h(n) being real. If
y2(n) ¼ y1(–n), find the overall impulse response h1(n) that relates y2(n) to x(n).
Solution
From Figure 6.13, we have the following relations:

y1(n) ¼ y(n) ∗ h(n)
y2 ðnÞ ¼ y1 ðnÞ ¼ yðnÞ∗hðnÞ

¼ ðxðnÞ∗hðnÞÞ∗hðnÞ
¼ xðnÞ∗ðhðnÞ∗hðnÞÞ ¼ xðnÞ∗h1 ðnÞ
Hence, the overall impulse response ¼ h1(n) ¼ h(n) ∗ h(–n)

x ( n) y ( n)
h1 (n ) h2 (n ) h2 (n )
Figure 6.14 Interconnection of three causal LTI systems
Example 6.16 Consider the cascade interconnection of three causal LTI systems as
shown in Figure 6.14. The impulse response h2(n) is given by
h2 ð nÞ ¼ uð nÞ uð n 2Þ
and the overall impulse response h(n) ¼ {1,5,10,11,8,4,1}. Determine the impulse
response h1(n).
Solution Let the overall impulse response of the cascaded system be h(n). Hence,
hðnÞ ¼ h1 ðnÞ∗h2 ðnÞ∗h2 ðnÞ
Since the convolution is associative in nature,
hðnÞ ¼ h1 ðnÞ∗ðh2 ðnÞ∗h2 ðnÞÞ
Let h3(n) ¼ h2(n) * h2(n).

Since h2(n) is nonzero for n ¼ 0 and 1 only, h3(n) can be written as
X
1
h3 ð nÞ ¼ h2 ðkÞh2 ðn kÞ
k¼0
X1
Therefore, h3 ð0Þ ¼ k¼0
h2 ðkÞh2 ðkÞ ¼ 1:1 þ 1:0 ¼ 1
X
1
h3 ð 1Þ ¼ h2 ðkÞh2 ð1 k Þ ¼ 1:1 þ 1:1 ¼ 2
k¼0
X
1
h3 ð 2Þ ¼ h2 ðkÞh2 ð2 k Þ ¼ 1:0 þ 1:1 ¼ 1
k¼0
Thus, we obtain
h3 ðnÞ ¼ f1; 2; 1g
Now, h(n) is nonzero in the interval 0 to 6 and h3(n) is nonzero in the interval
0 to 2:
hðnÞ ¼ h1 ðnÞ∗h3 ðnÞ
Hence, h1(n) will be nonzero in the interval 0 to 4. Then, we have

X
4
hðnÞ ¼ h1 ðnÞ∗h3 ðnÞ ¼ h1 ðkÞh3 ðn k Þ
k¼0
Let h1(n) ¼ {a1, a2, a3, a4, a5}.

Therefore, we have
X
4
hð 0Þ ¼ h1 ðkÞh3 ðk Þ ¼ a1 1 ¼ 1
k¼0
) a1 ¼ 1:
X
4
hð 1Þ ¼ h1 ðkÞh3 ð1 k Þ ¼ a1 1 þ a2 2 ¼ 5
k¼0
) a2 ¼ 3
X
4
hð 2Þ ¼ h1 ðkÞh3 ð2 k Þ ¼ a1 1 þ a2 2 þ a3 1 ¼ 101
k¼0
) a3 ¼ 3:
X
4
k¼0
) a4 ¼ 2:
X
4
k¼0
) a5 ¼ 1:
Thus,
h1 ðnÞ ¼ f1; 3; 3; 2; 1g
6.4.5 Stability and Causality of LTI Systems in Terms

of the Impulse Response
The output of a LTI system can be expressed as

X 1 X
1

j y ð nÞ j ¼ hðkÞxðn k Þ jhðkÞjjxðn kÞj
k¼1 k¼1
For bounded input x(n)
j x ð nÞ j β x < 1
we have
X
1
jyðnÞj βx jhðk Þj ð6:35Þ
k¼1
X1
It is seen from (6.35) that y(n) is bounded if and only if k¼1
jhðk Þj is
bounded. Hence, the necessary and sufficient condition for stability is that
X1
S¼ k¼1
jhðkÞj < 1: ð6:36Þ
The output y(n0) of a LTI causal system can be expressed as
X
1
y ð n0 Þ ¼ hðkÞxðn0 kÞ
k¼1
¼ hð1Þxðn0 þ 1Þ þ . . . . . . : þ hð2Þxðn0 þ 2Þ þ hð1Þxðn0 þ 1Þ

þhð0Þxðn0 Þ þ hð1Þxðn0 1Þ þ hð2Þxðn0 2Þ þ . . . ::
For a causal system, the output at n ¼ n0 should not depend on the future inputs.
Hence, in the above equation, h(k) ¼ 0 for k < 0.
Thus, it is clear that for causality of a LTI system, its impulse response sequence
hðnÞ ¼ 0 for n < 0: ð6:37Þ
Example 6.17 Check for the stability of the systems with the following impulse
responses:
(i) Ideal delay, h(n) ¼ δ(n nd); (ii) forward difference, h(n) ¼ δ(n þ 1) δ(n).
(iii) Backward difference, h(n) ¼ δ(n) δ(n 1); (iv) h(n) ¼ u(n).
(v) h(n) ¼ anu(n), where |a| < 1, and (vi) h(n) ¼ anu(n), where |a| 1.
Solution Given impulse responses of the systems, stability of each system can be
tested by computing the sum
X1
S¼ k¼1
jhðk Þj
In case of (i), (ii), and (iii), it is clear that S < 1. As such, the systems
corresponding to (i), (ii), and (iii) are stable.
For the impulse response given in (iv), the system is unstable since
X
1
S¼ uðnÞ ¼ 1:
n¼0
This is an exampleX of an infinite-duration impulse response (IIR) system.

1
In case of (v), S ¼ n¼0
jajn . For |a| < 1, S < 1, and hence the system is stable.
This is an example of a stable IIR system.
Finally, in case of (vi), |a| 1, and the sum is infinite, making the system
unstable.
6.5 Characterization of Discrete-Time Systems 297
Example 6.18 Check

n the followingsystems
n for causality:
n
(i) hðnÞ ¼ 34 uðnÞ, (ii) hðnÞ ¼ 12 uðn þ 2Þ þ 34 uðnÞ,
1n 3jnj
(iii) hðnÞ ¼ 2 uðn 1Þ, (iv) hðnÞ ¼ 4 , and (v) h(n) ¼ u(n þ 1) u(n)
Solution
(i) h(n) ¼ 0 for n < 0; hence the system is causal.
(ii) h(n) 6¼ 0 for n < 0; hence the system is not causal.
(iii) h(n) 6¼ 0 for n < 0; thus, the system is not causal.
jnj
(iv) hðnÞ ¼ 34 ; hence h(n) 6¼ 0 for n < 0; so, the system is not causal.
(v) h(n) ¼ u(n þ 1) – u(n), and h(n) 6¼ 0 for n < 0; so, the system is not causal.
Example 6.19
n Check the following systems for stability:
(i) hðnÞ ¼ 13 uðn 1Þ, (ii) h(n) ¼ u(n þ 2) u(n 5), (iii) h(n) ¼ 5nu(n 3),
1jnj
4 uðnÞ, and (v) hðnÞ ¼ 2
(iv) hðnÞ ¼ sin nπ cos πn
4
Solution
X1
(i) The system is stable, since S ¼ k¼1
jhðk Þj < 1:
(ii) h(n) ¼ u(n þ 2) – u(n – 5). The system is stable, since S is finite.
X X3 X1 n
1
(iii) h(n) ¼ 5nu(–n – 3). Hence, jhðnÞj ¼ 5n ¼ < 1: Therefore,
n n¼1 n¼3
5
the system is
nπstable.

(iv) hðnÞ ¼ sin 4 uðnÞ
Summing |h(n)| over all positive n, we see that S tends to infinity. Hence, the
system is not stable.
jnj
(v) hðnÞ ¼ 12 cos πn 4
jnj X1
|h(n)| is upper bounded by 12 . Thus, S ¼ k¼1
jhðk Þj < 1: Hence the
system is stable.
6.5 Characterization of Discrete-Time Systems
Discrete-time systems are characterized in terms of difference equations. An impor-

tant class of LTI discrete-time systems is one that is characterized by a linear
difference equation with constant coefficients. Such a difference equation may be
of two types, namely, non-recursive and recursive.
6.5.1 Non-Recursive Difference Equation
A non-recursive LTI discrete-time system is one that can be characterized by a linear

constant coefficient difference equation of the form
X
1
y ð nÞ ¼ bm x ð n m Þ ð6:38Þ
m¼1
where bm’s represent constants. By assuming causality, the above equation can be
written as
X
1
y ð nÞ ¼ bm xðn mÞ ð6:39Þ
m¼0
In addition, if x(n) ¼ 0 for n < 0 and bm ¼ 0 for m > N, then Eq. (6.39) becomes
X
N
y ð nÞ ¼ bm xðn mÞ ð6:40Þ
m¼0
Thus an LTI, causal, non-recursive system can be characterized by an Nth-order

linear non-recursive difference equation. The Nth-order non-recursive difference
equation has a finite impulse response (FIR). Therefore, an FIR filter is characterized
by a non-recursive difference equation.
6.5.2 Recursive Difference Equation
The response of a discrete-time system depends on the present and previous values
of the input as well as the previous values of the output. Hence a linear time-invariant
causal, recursive discrete-time system can be represented by the following Nth-order
linear recursive difference equation:
X
N X
N
y ð nÞ ¼ bm xðn mÞ am yðn mÞ ð6:41Þ
m¼0 m¼1
where am and bm are constants. An Nth-order recursive difference equation has an

infinite impulse response. Hence, an infinite impulse response (IIR) filter is charac-
terized by a recursive difference equation.
Example 6.20 An initially relaxed LTI system was tested with an input signal
x(n) ¼ u(n) and found to have a response as shown in Table 6.1.
(i) Obtain the impulse response of the system.
(ii) Deduce the difference equation of the system.
Table 6.1 Response of an LTI system for an input x(n) ¼ u(n)

n 1 2 3 4 5 . . .. . . . 100 . . .. . . .
y(n) 1 2 4 6 10 . . .. . . . 10 . . .. . . .
Solution
(i) From Table 6.1, it can be observed that the response y(n) for an input x(n) ¼ u(n)
is given by
yðnÞ ¼ f1; 2; 4; 6; 10; 10; 10; . . . . . . ::g
Similarly, for an input x(n) ¼ u(n1), the response y(n-1) is given by
yðn 1Þ ¼ f0; 1; 2; 4; 6; 10; 10; 10; . . . . . . ::g
For an input x(n) ¼ u(n)u(n1), the response of an LTI system is the impulse
response h(n) given by
hðnÞ ¼ yðnÞ yðn 1Þ ¼ f1; 1; 2; 2; 4g
(ii) The difference equation is given by
X
4
y ð nÞ ¼ hðmÞxðn mÞ
m¼0
Hence, the difference equation of the system can be written as
yðnÞ ¼ xðnÞ þ 1xðn 1Þ þ 2xðn 2Þ þ 2xðn 3Þ þ 4xðn 4Þ
A general linear constant coefficient difference equation can be expressed as

XN XM
y ð nÞ ¼ a yð n k Þ þ
k¼1 k k¼0
bk x ð n k Þ ð6:42Þ
The solution of the difference equation is the output response y(n). It is the sum of
two components which can be computed independently as
yðnÞ ¼ yc ðnÞ þ yp ðnÞ ð6:43aÞ
where yc(n) is called the complementary solution and yp(n) is called the particular
solution.
The complementary solution yc(n) is obtained by setting x(n) ¼ 0 in Eq. (6.42).

Thus yc(n) is the solution of the following homogeneous difference equation:
X
N
ak y ð n k Þ ¼ 0 ð6:43bÞ
k¼0
where a0 ¼ 1. To solve the above homogeneous difference equation, let us assume

that
y c ð nÞ ¼ λ n ð6:43cÞ
where the subscript c indicates the solution to the homogeneous difference equation.
Substituting yc(n) in Eq. (6.43b), the following equation can be obtained:
PN
k¼0 ak λnk ¼ 0
ð6:44Þ
¼ λnN λN þ a1 λN1 þ . . . :: þ aN1 λ þ aN ¼ 0
which takes the form:
λN þ a1 λN1 þ . . . :: þ aN1 λ þ aN ¼ 0 ð6:45Þ
The above equation is called the characteristic equation, which consists of N roots
represented by λ1, λ2, . . . . . . , λN. If the N roots are distinct, then the complementary
solution can be expressed as
yc ðnÞ ¼ α1 λ1n þ α2 λ2n þ . . . :: þ αN λNn ð6:46Þ
where α1, α2, . . . . . . , αN are constants which can be obtained from the specified
initial conditions of the discrete-time system. For multiple roots, the complementary
solution yc(n) assumes a different form. In the case when the root λ1 of the
characteristic equation is repeated m times, but λ2, . . . . . . , λN are distinct, then the
complementary solution yc(n) assumes the form

λ1n α1 þ α2 n þ . . . :: þ αm nm1 þ β2 λ2n þ . . . þ βNM λNM
n
ð6:47Þ
In case the characteristic equation consists of complex roots λ1, λ2 ¼ a jb, then
the complementary solution results in yc(n) ¼ (a2 + b2)n/2 (C1 cos nq + C2 sin nq),
where q ¼ tan–1b/a and C1 and C2 are constants.
We now look at the particular solution yp (n) of Eq. (6.42) The particular solution
yp(n) is any solution that satisfies the difference equation for the specific input signal
x(n), for n 0, i.e.,
XN XM
y ð nÞ þ a yð n k Þ ¼
k¼1 k k¼0
bk x ð n k Þ ð6:48Þ
The procedure to find the particular solution yp(n) assumes that yp(n) depends on
the form of x(n). Thus, if x(n) is a constant, then yp(n) is implicitly a constant.
Similarly, if x(n) is a sinusoidal sequence, then yp(n) is implicitly a sinusoidal
sequence and so on.
In order to find out the overall solution, the complementary and particular
solutions must be added. Hence,
yðnÞ ¼ yc ðnÞ þ yp ðnÞ ð6:49Þ
Example 6.21 Determine impulse response for the case of x(n) ¼ δ(n) of a discrete-
time system characterized by the following difference equation:
yðnÞ þ 2yðn 1Þ 3yðn 2Þ ¼ xðnÞ ð6:50Þ
Solution First, we determine the complementary solution by setting x(n) ¼ 0 and

y(n) ¼ λn in Eq. (6.50), which gives us

λn þ 2λn1 3λn2 ¼ λn2 λ2 þ 2λ 3
¼ λn2 ðλ 1Þðλ þ 3Þ ¼ 0
Hence, the zeros of the characteristic polynomial λ2 þ 2λ 3 are λ1 ¼ 3 and

λ2 ¼ 1.
Therefore, the complementary solution is of the form
yc ðnÞ ¼ α1 ð3Þn þ α2 ð1Þn ð6:51Þ
For impulse x(n) ¼ δ(n), x(n) ¼ 0 for n > 0 and x(0) ¼ 1. Substituting these
relations in Eq. (6.50) and assuming that y(1) ¼ 0 and y(2) ¼ 0, we get
yð0Þ þ 2yð1Þ 3yð2Þ ¼ xð0Þ ¼ 1,
i.e., y(0) ¼ 1. Similarly y(1) þ 2y(0) – 3y(–1) ¼ x(1) ¼ 0 yields y(1) ¼ –2.
Thus, from Eq. (6.51), we get
α1 þ α2 ¼ 1 and
3α1 þ α2 ¼ 2
Solving these two equations, we obtain α1 ¼ 3/4 and α2 ¼ 1/4.

Since x(n) ¼ 0 for n > 0, there is no particular solution. Hence, the impulse
response is given by
hðnÞ ¼ yc ðnÞ ¼ 0:75ð3Þn þ 0:25ð1Þn ð6:52Þ

Example 6.22 A discrete-time system is characterized by the following difference

equation:
yðnÞ þ 5yðn 1Þ þ 6yðn 2Þ ¼ xðnÞ ð6:53Þ
Determine the step response of the system, i.e., x(n) ¼ u(n).

Solution For the given difference equation, total solution is given by
yðnÞ ¼ yc ðnÞ þ yp ðnÞ
First, we determine the complementary solution by setting x(n) ¼ 0 and y(n) ¼ λn

in Eq. (6.53), which give us

λn þ 5λn1 þ 6λn2 ¼ λn2 λ2 þ 5λ þ 6 ¼ 0
Hence, the zeros of the characteristic polynomial λ2 + 5λ þ 6 are λ1 ¼ –3 and

λ2 ¼ –2.
Therefore, the complementary solution is of the form
yc(n) ¼ α1(3)n + α2(2)n
The particular solution for the step input is of the form
yp(n) ¼ K
For n > 2, substituting yp(n) ¼ K and x(n) ¼ 1 in Eq. (6.53), we get
K þ 5K þ 6K ¼ 1; K ¼ 12 1
, and yp ðnÞ ¼ 12
1
.
Therefore, the solution for given difference equation is
1
yðnÞ ¼ α1 ð3Þn þ α2 ð2Þn þ ð6:54Þ
12
For n ¼ 0, Eq. (6.53) becomes
y(0) þ 5y(–1) þ 6y(–2) ¼ x(0)
Assuming y(–1) ¼ y(–2) ¼ 0, from the above equation, we get y(0) ¼ x(0) ¼ 1
and for n ¼ 1, y(1) þ 5y(0) þ 6y(–1) ¼ x(1) ¼ 1, i.e., y(1) ¼ –4.
Then, we get from Eq. (6.54)
α1 þ α2 þ 12
1
¼1
1
3α1 2α2 þ ¼ 4
12
16
Solving these equations, we arrive at α1 ¼ 27
12 and α2 ¼ 12 .
Then, the step response is given by
27 16 1
y ð nÞ ¼ ð3Þn ð2Þn þ ð6:55Þ
12 12 12
Example 6.23 A discrete-time system is characterized by the following difference

equation:
yðnÞ 2yðn 1Þ þ yðn 2Þ ¼ xðnÞ xðn 1Þ ð6:56Þ
Determine the response y(n), n 0 when the system input is x(n) ¼ (–1)nu(n) and
the initial conditions are y(1) ¼ 1 and y(2) ¼ 1.
Solution For the given difference equation, the total solution is given by
yðnÞ ¼ yc ðnÞ þ yp ðnÞ
First, determine the complementary solution by setting x(n) ¼ 0 and y(n) ¼ λn in

Eq. (6.56); this gives

λn 2λn1 þ λn2 ¼ λn2 λ2 2λ þ 1 ¼ 0
Hence, the zeros of the characteristic polynomial λ2 2λ þ 1 are λ1 ¼ λ2 ¼ 1.

It has repeated roots; thus, the complementary solution is of the form
yc(n) ¼ 1n(α1 + nα2).
The particular solution for the step input is of the form
yp(n) ¼ K(1)nu(n).
Substituting x(n) ¼ (–1)nu(n) and yp(n) ¼ K(-1)nu(n) in Eq. (6.56), we get
K(–1)nu(n) – 2K(–1)n–1u(n – 1) þ K(–1)n–2u(n – 2) ¼ (–1)nu(n) – (–1)n–1u(n – 1)
For n ¼ 2, the above equation becomes K + 2K + K ¼ 2; K ¼ 12.
Therefore, the particular solution is given by
1
yp ðnÞ ¼ ð1Þn uðnÞ
2
Then, the total solution for given difference equation is
1
yðnÞ ¼ 1n ðα1 þ nα2 Þ þ ð1Þn uðnÞ: ð6:57Þ
2
For n ¼ 0, Eq. (6.56) becomes
yð0Þ 2yð1Þ þ yð2Þ ¼ 1
Using the initial conditions y(–1) ¼ 1 and y(–2) ¼ –1, we get y(0) ¼ 4.
Then, for n ¼ 1, from Eq. (6.56), we get y(1) ¼ 5. Thus, we get from Eq. (6.57)
α1 þ ð1=2Þ ¼ 4
α1 þ α2 ð1=2Þ ¼ 5
Solving these two equations, we arrive at α1 ¼ (7/2) and α2 ¼ 2. Thus, the

response of the system for the given input is

7 1
y ð nÞ ¼ 1n
þ 2n þ ð1Þn uðnÞ ð6:58Þ
2 2
6.5.4 Computation of Impulse and Step Responses Using

MATLAB
The impulse and step responses of LTI discrete-time systems can be computed using
MATLAB function:
y ¼ filter ðb; a; xÞ
where b and a are the coefficient vectors of difference equation describing the
system, x is the input data vector, and y is the vector generated assuming zero initial
conditions. The following example illustrates the computation of the impulse and
step responses of an LTI system.
Example 6.24 Determine the impulse and step responses of a discrete-time system
described by the following difference equation:
yðnÞ 2yðn 1Þ ¼ xðnÞ þ 0:1xðn 1Þ 0:06xðn 2Þ
Solution Program 6.2 is used to compute and plot the impulse and step responses,
which are shown in Figure 6.15a and b, respectively.
Program 6.2: Illustration of Impulse and Step Response Computation
clear;clc;
flag=input(‘enter 1 for impulse response, and 2 for step response’);
len=input(‘enter desired response length=‘);
b=[l -2];%b coefficients of the difference equation
a=[l 0.l -0.06]; %a coefficients of the difference equation
if flag==l;
x=[l,zeros(l,len-l)];
end
if flag==2;
x=[ones(1,len)];
end
y=filter(b,a,x);
n=0:1:len-1;
stem(n,y)
xlabel(‘Time index n’); ylabel(‘Amplitude’);
6.6 Sampling of Discrete-Time Signals 305
Figure 6.15 (a) Impulse response and (b) step response for Example 6.24
6.6 Sampling of Discrete-Time Signals
It is often necessary to change the sampling rate of a discrete-time signal, i.e., to

obtain a new discrete-time representation of the underlying continuous-time signal
of the form x’(n) ¼ xa(nT’). One approach to obtain the sequence x’(n) from x(n) is to
reconstruct xa(t) from x(n) and then resample xa(t) with period T’ to obtain x’(n).
x ( n) xd (n) = x(nM )
M
Sampling period T Sampling period T ' = MT
Figure 6.16 Block diagram representation of a down sampler
x ( n) xe (n) = x(n / L)
L
Sampling period T Sampling period T ' = TL
Figure 6.17 Block diagram representation of an up-sampler
Often, however, this is not a desirable approach, because of the non-ideal analog
reconstruction filter, DAC, and ADC that would be used in a practical implementation.
Thus, it is of interest to consider methods that involve only discrete-time operation.
6.6.1 Discrete-Time Down Sampler
The block diagram representation of a down sampler, also known as a sampling rate
compressor, is depicted in Figure 6.16.
The down-sampling operation is implemented by defining a new sequence xd(n)
in which every Mth sample of the input sequence is kept and (M1) in-between
samples are removed to obtain the output sequence, i.e., xd(n) is identical to the
sequence obtained from xa(t) with a sampling period T’ ¼ MT
xd ðnÞ ¼ xðnM Þ ð6:59Þ
For example, if x(n) ¼ {2,6,3,0,1,2,–5,2,4,7,–1,1,–2,. . .},

then xd(n) ¼ {2, 1, 4, –2, . . .} for M ¼ 4, i.e., M1 ¼ 3 samples are left in
between the samples of x(n) to get xd(n).
6.6.2 Discrete-Time Up-Sampler
The block diagram representation of an up-sampler, also called a sampling rate

expander or simply an interpolator, is shown in Figure 6.17.
The output of an up-sampler is given by
X
1 n
x e ð nÞ ¼ xðk Þδðn kLÞ ¼ x n ¼ 0, L, 2L, . . . . . . ::

k¼1
L ð6:60Þ
¼0 otherwise
6.7 State-Space Representation of Discrete-Time LTI Systems 307
Eq. (6.60) implies that the output of an up-sampler can be obtained by inserting
(L – 1) equidistant zero-valued samples between two consecutive samples of the
input sequence x(n), i.e., xe(n) is identical to the sequence obtained from xa(t) with
a sampling period T’ ¼ T/L. For example, if xe(n) ¼ {2,1,4,–2, . . . .}, then
xe(n) ¼ {2,0,0,0,1,0,0,0,4,0,0,0,–2,0,0,0, . . .} for L ¼ 4, i.e., L1 ¼ 3 zero-
valued samples are inserted in between the samples of x(n) to get xe(n).
6.7 State-Space Representation of Discrete-Time

LTI Systems

Single-Output Discrete-Time LTI Systems
Consider a single-input single-output discrete-time LTI system described by the

following Nth-order difference equation:
yðnÞ þ a1 yðn 1Þ þ a2 yðn 2Þ þ . . . þ aN yðn N Þ ¼ ℧ðnÞ ð6:61Þ
where y(n) is the system output and ℧(n) is the system input.
Define the following useful set of state variables:
x1 ðnÞ ¼ yðn N Þ, x2 ðnÞ ¼ yðn N þ 1Þ, x3 ðnÞ ¼ yðn N þ 2Þ, . . . ,

ð6:62Þ
x N ð nÞ ¼ y ð n 1Þ
Then from Eqs. (6.61) and (6.62), we get
x 1 ð n þ 1Þ ¼ x 2 ð nÞ
x 2 ð n þ 1Þ ¼ x 3 ð nÞ
⋮
xN1 ðn þ 1Þ ¼ xN ðnÞ
xN ðn þ 1Þ ¼ aN x1 ðnÞ aN1 x2 ðnÞ a1 xN ðnÞ þ ℧ðnÞ ð6:63aÞ
and
y ð nÞ ¼ x N ð n þ 1Þ ð6:63bÞ
Eqs. (6.63a) and (6.63b) can be written in matrix form as

2 3 2 32 3 2 3
x1 ð n þ 1 Þ 0 1 0 0 x1 ðnÞ 0
6 7 6 76 7 6 7
6 x2 ð n þ 1 Þ 7 6 0 0 76 7 6 7
6 7 6 0 1 76 x2 ðnÞ 7 6 0 7
6 7 6 76 7 6 7
6 ⋮ 7¼6 ⋮ ⋮ ⋮ ⋱ ⋮ 76 7 6 7
6 7 6 76 ⋮ 7 þ 6 0 7℧ðnÞ
6 7 6 76 7 6 7
6 xN1 ðn þ 1Þ 7 6 0 1 76 7 6 7
4 5 4 0 0 54 xN1 ðnÞ 5 4 ⋮ 5
xN ð n þ 1 Þ aN aN1 aN2 a1 xN ðnÞ 1
ð6:64aÞ
2 3 2 3
x 1 ð nÞ 0
6 7 6 7
6 x 2 ð nÞ 7 6 0 7
6 7 6 7
6 7 6 7
yðnÞ ¼ ½aN aN1 aN2 a1 6 ⋮ 7 þ 6 0 7℧ðnÞ ð6:64bÞ
6 7 6 7
6 x ðnÞ 7 6 ⋮ 7
4 N1 5 4 5
x N ð nÞ 1
Define a Nx1 dimensional vector called state vector as

2 3
x 1 ð nÞ
6 7
6 x 2 ð nÞ 7
6 7
6 7
X ð nÞ ¼ 6 ⋮ 7 ð6:65Þ
6 7
6 x ð nÞ 7
4 N1 5
x N ð nÞ
More compactly Eqs. (6.64a) and (6.64b) can be written as
X ðn þ 1Þ ¼ AX ðnÞ þ b℧ðnÞ ð6:66aÞ
yðnÞ ¼ cX ðnÞ þ d℧ðnÞ ð6:66bÞ
where 2 3 2 3
0 1 0 0 0
6 7 6 7
6 0 0 1 0 7 6 0 7
6 7 6 7
6 7 6 7
A¼6 ⋮ ⋮ ⋮ ⋱ ⋮ 7; b ¼ 6 0 7;
6 7 6 7
6 0 1 7 6⋮7
4 0 0 5 4 5
aN aN1 aN2 a1 1
c ¼ ½aN aN1 aN2 a1 ; d ¼ 1:
Equation (6.66a) and (6.66b) is called N-dimensional state-space representation

or state equations of the discrete-time system.
6.7 State-Space Representation of Discrete-Time LTI Systems 309
Example 6.25 Obtain the state-space representation of a discrete-time system

described by the following differential equation:
yðn 3Þ þ 2yðn 2Þ þ 3yðn 1Þ þ 4yðnÞ ¼ ℧ðnÞ
Solution The order of the differential equation is three. We have to choose three
state variables:
Let x1(n) ¼ y(n 3), x2(n) ¼ y(n 2), x3(n) ¼ y(n 1)
Then
x 1 ð n þ 1Þ ¼ x 2 ð nÞ
x 2 ð n þ 1Þ ¼ x 3 ð nÞ
1 1 3 1
x3 ðn þ 1Þ ¼ x1 ðnÞ x2 ðnÞ x3 ðnÞ þ ℧ðnÞ
4 2 4 4
y ð nÞ ¼ x 3 ð n þ 1Þ
The state-space representation in matrix form is given by

2 3 2 32 2 3 3
x 1 ð n þ 1Þ 0 1 0 0
x 1 ð nÞ
6 7 6 76 7 6 7
6 7 6 0 1 776 7 607
6 x 2 ð n þ 1Þ 7 ¼ 6
6
0
7 6 x 2 ð nÞ 7 þ 6 7 ℧ð n Þ
4 5 4 5 4 5 6 4
7
1 1 3 15
x 3 ð n þ 1Þ x 3 ð nÞ
4 2 4 4
2 3
x 1 ð nÞ
7 1
1 1 3 6 6 7
y ð nÞ ¼ 6 x 2 ð nÞ 7 þ ℧ð n Þ
4 2 4 4 5 4
x 3 ð nÞ
6.7.2 State-Space Representation of Multi-input Multi-output

Discrete-Time LTI Systems
The state-space representation of discrete-time system with m inputs and 1 output

and N state variables can be expressed as
X ðn þ 1Þ ¼ AX ðnÞ þ B℧ðnÞ ð6:67aÞ

yðnÞ ¼ CX ðnÞ þ D℧ðnÞ ð6:67bÞ
where
2 3 2 3
a11 a12 a1N b11 b12 b1m
6 7 6 7
6 a21 a22 a2N 7 6 b21 b22 b2m 7
A¼6
6⋮
7 B¼6 7
4 ⋮ ⋱ ⋮ 7 5
6⋮
4 ⋮ ⋱ ⋮ 7 5
aN1 aN2 aNN NN bN1 bN2 bNm Nm
2 3 2 3
c11 c12 c1N d11 d 12 d1m
6 7 6 7
6 c21 c22 c2N 7 6 d21 d 22 d2m 7
C¼6
6⋮
7 D¼6 7
4 ⋮ ⋱ ⋮7 5
6⋮
4 ⋮ ⋱ ⋮7 5
cl1 cl2 clN lN d l1 d l2 dlm lm
6.8 Problems
1. Determine if the following discrete-time signals are periodic:

(i) xðnÞ ¼ sin πn6 þ3
π
3πn
(ii) xðnÞ ¼ cos 10 þ =0

(iii) xðnÞ ¼ cos n2 þ θ

(iv) xðnÞ ¼ ej πn
4 þ= 0
(v) x(n) ¼ 6(1) n
63πn
(vi) xðnÞ ¼ sin 3πn 8 cos 64
3πn
(vii) xðnÞ ¼ sin 8 þ cos 63πn
64
(viii) xðnÞ ¼ ej 7πn
4 þe
j 3πn
4
2. Determine if the following discrete-time signals are energy or power signals or

neither. Calculate the energy and power of the signals in each case:
3πn
(ix) xðnÞ ¼ cos πn 2 þ sin 4
(x) x(n) ¼ (1)n
8 n
> ð 3Þ 0 n 10
<
(xi) xðnÞ ¼ 2 11 n 15
>
:
8 0 otherwise

< cos πn 10 n 0
(xii) xðnÞ ¼ 15
:
0 otherwise

(xiii) xðnÞ ¼ ej πn
2 þ π
8
3. Determine if the following discrete-time signals are even, odd, or neither even
nor odd:

(i) xðnÞ ¼ sin ð4nÞ þ cos 2πn
3
6.8 Problems 311
2πn
(ii) xðnÞ ¼ sin πn
30 þ cos 3
3πn
(iii) xðnÞ ¼ sin 8 þ cos 3πn
4
ð1Þn n 0
(iv) xðnÞ ¼
0 n<0
4. Check the following for linearity, time-invariance, and causality:
(i) y(n) ¼ 5nx2(n). (ii) y(n) ¼ x(n)sin2n. (iii) y(n) ¼ e–nx(n + 3)
n1
5. Given the input x(n) ¼ u(n) and the output yðnÞ ¼ 12 uðn 1Þ of a system,
(i) Determine the impulse response h(n)
(ii) Is the system stable?
(iii) Is the system causal?
6. Check for stability and causality of a system for the following impulse
responses:
πn
(i) hðnÞ ¼ e2n sin πn
2 uðn 1Þ (ii) hðnÞ ¼ sin 2 uðnÞ
7. Determine if the following signals are periodic, and if periodic, find its period:
3πn
(ii) sin n (b) e jπn/3 (c) sin πn
4 þ sin 4
8. Determine the convolution of the sum of the two sequences:

x1(n) ¼ (3,2,1,2) and x2(n) ¼ (1,2,1,2).
9. Determine the convolution of the sum of the two sequences x1(n) and x2(n), if
x1(n) ¼ x2(n) ¼ cnu(n) for all n, where c is a constant.
10. Determine the impulse response (i.e., when x(n) ¼ δ(n) of a discrete-time system
characterized by the following difference equation:
yðnÞ þ yðn 1Þ 6yðn 2Þ ¼ xðnÞ
11. A discrete-time system is characterized by the following difference equation:
6yðnÞ yðn 1Þ yðn 2Þ ¼ 6xðnÞ
Determine the step response of the system, i.e., x(n) ¼ u(n), given the initial
conditions y(-1) ¼ 1 and y(-2) ¼ -1.
12. A discrete-time system is characterized by the following difference equation:
yðnÞ 5yðn 1Þ þ 6yðn 2Þ ¼ xðnÞ
Determine the response of the system for x(n) ¼ nu(n) and initial conditions
y(-1) ¼ 1 and y(-2) ¼ 0.
13. Determine the response of the system described by the following difference
equation:
yðnÞ þ yðn 1Þ ¼ sin 3n uðnÞ
14. Obtain the state-space representation of a discrete-time system described by the

following differential equation:
2yðnÞ þ 3yðn 1Þ þ yðn 2Þ ¼ ℧ðnÞ
1. Using the function impz, write a MATLAB program to determine the impulse
response of a discrete-time system represented by
yðnÞ 5yðn 1Þ þ 6yðn 2Þ ¼ xðnÞ 2xðn 1Þ
2. Write a MATLAB program to illustrate down-sampling by an integer factor of

4 of a sum of two sinusoidal sequences, each of length 50, with normalized
frequencies of 0.2 Hz and 0.35 Hz.
3. Write a MATLAB program to illustrate up-sampling by an integer factor of 4 of a
sum of two sinusoidal sequences, each of length 50, with normalized frequencies
of 0.2 Hz and 0.35 Hz.
Further Reading
1. Linden, D.A.A.: Discussion of sampling theorem. Proceedings of the IRE. 47, 1219–1226 (1959)
2. Proakis, J.G., Manolakis, D.G.: Digital Signal Processing Principles, Algorithms and Applica-
tions, 3rd edn. Prentice-Hall, India (2004)
3. Crochiere, R.E., Rabiner, L.R.: Multirate Digital Signal Processing. Prentice-Hall, Englewood
Cliffs (1983)
4. Hsu, H.: Signals and Systems, Schaum’s Outlines, 2nd edn, Mc Graw Hill, New York (2011)
5. Mandal, M., Asif, A.: Continuous and Discrete Time Signals and Systems. Cambridge, UK;
New York: Cambridge University Press, (2007)
Chapter 7
Frequency Domain Analysis of Discrete-
Time Signals and Systems
This chapter discusses the transform domain representation of discrete-time

sequences by discrete-time Fourier series (DTFS) and discrete-time Fourier trans-
form (DTFT) in which a discrete-time sequence is mapped into a continuous
function of frequency. We first obtain the discrete-time Fourier series (DTFS)
expansion of a periodic sequence. The periodic convolution and the properties of
DTFS are discussed. The Fourier transform domain representation of discrete-time
sequences are described along with the conditions for the existence of DTFT and its
properties. Later, the frequency response of discrete-time systems, frequency
domain representation of sampling process, and reconstruction of band-limited
signals from its samples are discussed.
7.1 The Discrete-Time Fourier Series
If a sequence x(n) is periodic with period N, then x(n) ¼ x(n + N ) for all n. In analogy
with the Fourier series representation of a continuous periodic signal, we can look
for a representation of x(n) in terms of the harmonics corresponding to the funda-
mental frequency of (2π/N ). Hence, we may write x(n) in the form
X
x ð nÞ ¼ b ej2πkn=N
k k
ð7:1aÞ
It can easily be verified from Eq. (7.1a) that x(n) ¼ x(n + N ). Also, we know that
there are only N distinct values for e j2πkn/N, corresponding to k ¼ 0, 1, . . . . N 1,
these being 1, e j2πn/N, . . ., e j2πn(N 1)/N. Hence, we may rewrite (7.1a) as
XN1
x ð nÞ ¼ k¼0
ak ej2πkn=N ð7:1bÞ
It should be noted that the summation can be taken over any N consecutive values
of k. Eq. (7.1b) is called the discrete-time Fourier series (DTFS) of the periodic

314 7 Frequency Domain Analysis of Discrete-Time Signals and Systems
sequence x(n) and ak as the Fourier coefficients. We will now obtain the expression
for the Fourier coefficients ak. It can easily be shown that {e j2πkn/N} is an orthogonal
sequence satisfying the relation
XN1
j2πkn=N j2πl n=N 0 k¼
6 l
e e ¼ ð0 k; l ðN 1Þ ð7:2Þ
n¼0 N k¼l
Now, multiplying both sides of (7.1b) by ej2πln/N and summing over n between
0 and (N 1), we get
PN1 PN1 PN1
n¼0 xðnÞ ej2πln=N ¼ n¼0 k¼0 ak ej2πkn=N ej2πln=N
PN1 PN1
¼ k¼0 ak n¼0 ej2πkn=N ej2πln=N
¼ al N, using Eq:ð7:2Þ:
Hence,
1 XN1
ak ¼ xðnÞej2πkn=N , k ¼ 0, 1, 2, . . . , N 1 ð7:3Þ
N n¼0
It is common to associate the factor (1/N ) with x(n) rather than ak. This can be
done by denoting Nak by X(k); in such a case, we have
1 XN1
x ð nÞ ¼ X ðkÞ ej2πkn=N ð7:4Þ
N k¼0
where the Fourier coefficients X(k) are given by

XN1 j2πkn
X ðk Þ ¼ n¼0
x ð nÞ e N , k ¼ 0, 1, 2, . . . , N 1 ð7:5Þ
It is easily seen that X(k + N ) ¼ X(k), that is, the Fourier coefficient sequence X(k)
is also periodic of period N. Hence, the spectrum of a signal x(n) that is periodic with
period N is also a periodic sequence with the same period. It is also noted that since
the Fourier series of a discrete periodic signal is a finite sequence, the series always
converges, and the Fourier series gives an exact alternate representation of the
discrete sequence x(n).
7.1.1 Periodic Convolution
In the case of two periodic sequences x1(n) and x2(n) having the same period N,
linear convolution as defined by Eq. (6.29) does not converge. Hence, we define a
different form of convolution for periodic signals by the relation
7.1 The Discrete-Time Fourier Series 315
Table 7.1 Some important properties of DTFS

Property Periodic sequence DTFS coefficients
Linearity ax1(n) þ bx2(n) a and b are constants aX1(k) þ bX2(k)
Time shifting x(n m) ejð N Þkm X ðk Þ
2π
Frequency shifting ejð N Þln xðnÞ X(k l )

2π
Periodic convolution X
N1 X1(k)X2(k)
x1 ðmÞx2 ðn mÞ
m¼0
Multiplication x1(n)x2(n) 1 X
N1
X 1 ðlÞX 2 ðk lÞ
N l¼0
x*(n) X*(k)
x*(n) X*(k)
RefxðnÞg 1
X e ðk Þ ¼ ðX ðk Þ þ X ∗ ðk ÞÞ
j lmfxðnÞg 2
1
X o ðk Þ ¼ ðX ðk Þ X ∗ ðk ÞÞ
2
Symmetry properties x e ð nÞ RefX ðk Þg
1
¼ ½xðnÞ þ x∗ ðnÞ j Im fX ðk Þg
2
xo ðnÞ
1
¼ ½xðnÞ x∗ ðnÞ
2
If xðnÞis real RefX ðk Þg
1
xe ðnÞ ¼ ½xðnÞ þ xðnÞ j Im fX ðk Þg
2
1
xo ðnÞ ¼ ½xðnÞ xðnÞ
2
XN1 XN1
y ð nÞ ¼ x ðmÞx2 ðn mÞ ¼
m¼0 1
x ðn
m¼0 1
mÞx2 ðmÞ ð7:6Þ
The above convolution is called periodic convolution. It may be observed that y

(n) ¼ y(n + N ), that is, the periodic convolution is itself periodic of period N. Some
important properties of the DTFS are given in Table 7.1. In this table, it is assumed
that x1(n) and x2(n) are periodic sequences having the same period N. The proofs are
omitted here, since they are similar to the ones that will be given in Section 7.2 for
the corresponding properties of the DTFT.
Example 7.1 Determine the Fourier series representation for the following discrete-
time signal:
πn
2πn
xðnÞ ¼ 3 sin sin
4 5
Solution πn
2πn
xðnÞ ¼ 3 sin sin
4 5

3 3n π 13nπ
¼ cos cos
2 20 20

3 j 3nπ 3nπ
j 20
13nπ 13nπ
j 20
¼ e 20 þe e j 20
e
4

3 3nπ 17nπ 13nπ 7nπ
¼ ej 20 þ ej 20 ej 20 ej 20
4
X(0) ¼ X(1) ¼ X(2) ¼ X(4) ¼ X(5) ¼ X(6) ¼ X(8) ¼ X(9) ¼ X(10) ¼ X

(11) ¼ X(12) ¼ X(14) ¼ X(15) ¼ X(16) ¼ X(18) ¼ X(19) ¼ 0, X(3) ¼ Xð17Þ ¼ 32,
X(7) ¼ Xð13Þ ¼ 32
Example 7.2 Discrete-time signal x(n) is periodic of period 8, and x(n) ¼ n for
0 n 7.
Solution The sequence is periodic with period N ¼ 8.
XN1
X ðk Þ ¼ n¼0
xðnÞej2πkn=N , k ¼ 0, 1, 2, . . . , N 1
Using above equation, the DTFS coefficients are computed as

X(0) ¼ 28., X(4) ¼ 4
X(1) ¼ 4þ 9.6569 j, X(5) ¼ 4 1.6569 j
X(2) ¼ 4 þ 4 j X(6) ¼ 4 4 j
X(3) ¼ 4 þ 1.6569 j, X(7) ¼ 4 9.6569 j
X(k) ¼ {28, 4þ 9.6569j, 4 þ 4 j, 4 þ 1.6569 j, 4, 4 1.6569 j,
44i,4 49.6569 j}
7.2 Representation of Discrete-Time Signals and Systems

in Frequency Domain
7.2.1 Fourier Transform of Discrete-Time Signals
The discrete-time Fourier transform (DTFT) of a finite energy sequence x(n) is

defined as X1
F½xðnÞ ¼ X ejω ¼ n¼1
xðnÞeðjωnÞ ð7:7Þ
7.2 Representation of Discrete-Time Signals and Systems in Frequency Domain 317
From X(e jω), x(n) can be computed as

ðπ
1
x ð nÞ ¼ X ejω eðjωnÞ dω ð7:8Þ
2π π
Eq. (7.8) is called the inverse Fourier transform.

Convergence of the DTFT
The existence of DTFT of x(n) depends on the convergence of the series in Eq. (7.7).
Now, we look at theXcondition for convergence.
jω 1
Let X k e ¼ k¼1
xðnÞeðjωnÞ denote the partial sum of the weighted
complex exponentials in Eq. (7.7). Then for uniform convergence of X(e jω),

lim X k ejω ¼ X ejω ð7:9Þ
k!1
Hence, for uniform convergence of X(e jω), x(n) must be absolutely summable,
i.e.,
X
1
jxðnÞj < 1, ð7:10Þ
n¼1
Then

jω X1
X1 X1
X e ¼ xðnÞe jωn
jxðnÞj ejωn jxðnÞj < 1 ð7:11Þ
n¼1 n¼1 n¼1
guaranteeing the existence of X(e jω), for all values of ω. Consequently, Eq. (7.10)
is only a sufficient condition for the existence of the DTFT, but is not a necessary
condition.
7.2.2 Theorems on DTFT
We will now consider some important theorems concerning DTFT that can be used
in digital signal processing. All these properties can be proved using the definition of
DTFT. The following notation is adopted for convenience:

X ejω ¼ F½xðnÞ ð7:12aÞ

xðnÞ ¼ F1 X ejω ð7:12bÞ
Linearity If x1(n) and x2(n) are two sequences with Fourier transforms X1(e jω)
and X2(e jω), then the Fourier transform of a linear combination of x1(n) and x2(n) is
given by

F½a1 x1 ðnÞ þ a2 x2 ðnÞ ¼ a1 X 1 ejω þ a2 X 2 ejω ð7:13Þ
where a1and a2 are arbitrary constants.

Time Reversal If x(n) is a sequence with Fourier transform X(e jω), then the Fourier
transform of time reversed sequence x(n) is given by

F½xðnÞ ¼ X ejω ð7:14Þ
Time Shifting If x(n) is a sequence with Fourier transform X(e jω), then the Fourier
transform of the delayed sequence x(n k), where k an integer, is given by

F½xðn k Þ ¼ ejωk X ejω ð7:15Þ
Therefore, time shifting results in a phase shift in the frequency domain.

Frequency Shifting If x(n) is a sequence with Fourier transform X(e jω), then the
Fourier transform of the sequence ejω0 n x(n) is given by

i
F ejω0 n xðnÞ ¼ X ejðωω0 Þ ð7:16Þ
Thus, multiplying a sequence x(n) by a complex exponential ejω0 n in the time

domain corresponds to a shift in the frequency domain.
Differentiation in Frequency If x(n) is a sequence with Fourier transform X(e jω),
then the Fourier transform of the sequence nx(n) is given by
d jω
F½nxðnÞ ¼ j X e ð7:17Þ
dω
Convolution Theorem If x1(n) and x2(n) are two sequences with Fourier trans-
forms X1(e jω)and X2(e jω), then the Fourier transform of the convolution of x1(n) and
x2(n) is given by

F½x1 ðnÞ∗x2 ðnÞ ¼ X 1 ejω X 2 ejω ð7:18Þ
Hence, convolution of two sequences x1(n) and x2(n) in the time domain is equal
to the product of their frequency spectra. In the above equation, since X1(e jω) and
X2(e jω) are periodic in ω with period 2π, the convolution is a periodic convolution.
Windowing Theorem If x(n) and w(n) are two sequences with Fourier transforms
X(e jω) and W(e jω), then the Fourier transform of the product of x(n) and w(n) is given
by
ðπ
1
F½xðnÞwðnÞ ¼ X ejω ∗W ejω ¼ X ejθ W ejðωθÞ dθ ð7:19Þ
2π π
The above result is called the windowing theorem.

Correlation Theorem If x1(n) and x2(n) are two sequences with Fourier transforms
X1(e jω) and X2(e jω), then the Fourier transform of the correlation r x1 x2 ðlÞ of x1(n) and
x2(n) defined by
X1
r x1 x2 ðlÞ ¼ x ðnÞx2 ðn
n¼1 1
lÞ ð7:20aÞ
is given by
hX1 i
F ½ r x1 x2 ð l Þ ¼ F n¼1
x 1 ðn Þx 2 ð n l Þ ¼ X 1 ejω X 2 ejω ð7:20bÞ
which is called the cross energy density spectrum of the signals x1(n) and x2(n).
Parseval’s Theorem If x(n) is a sequence with Fourier transform X(e jω), then the
energy E of x(n) is given by
X1 ðπ
1 jω 2
E¼ jxðnÞj ¼ 2 X e dω ð7:21Þ
1 2π π
where |X(e jω)|2 is called the energy density spectrum.

Proof The energy E of x(n) is defined as
P1 P
E¼ jxðnÞj2 ¼ 1
1
∗
1 xðnÞx ðnÞ
ð
P 1 π ∗ jω jωn
¼ 1
1 x ð n Þ X e e dω ð7:22Þ
2π π
using Eq. (7.8).

Interchanging the integration and summation signs, the above equation can be
rewritten as
ðπ
1 X 1
E¼ X ∗ ejω xðnÞ ejωn dω
2π π 1
ðπ
1 ∗

¼ X ejω X ejω dω
2π π
ðπ
1 jω 2
¼ X e dω
2π π
Thus,
X1 ðπ
1 jω 2
E¼ jxðnÞj ¼ 2 X e dω ð7:23Þ
1 2π π
Table 7.2 Some properties of discrete-time Fourier transforms

Property Sequence DTFT
Linearity a1x1(n) þ a2x2(n) a1X1(e jω) þ a2X2(e jω)
Time shifting x(n k) ejωkX(e jω)
Time reversal x(n) X(ejω)

Frequency shifting ejω0 n xðnÞ X ejðωω0 Þ
Differentiation in the frequency domain nx(n) d
jdω X ðejω Þ
Convolution theorem x1(n) * x2(n) X1(e jω) X2(e jω)
Windowing theorem x1(n)x2(n) X1(e jω) ∗ X2(e jω)
X1
Correlation theorem x ðnÞx2 ðn lÞ X1(e jω)X2(ejω)
1 1
Table 7.3 Some useful x(n) DTFT

DTFT pairs
δ(n) 1
X1
1 (1 < n < 1) 2πδðω þ 2πk Þ
k¼1
anu(n), |a| < 1 1
1aejω
sin ðωc nÞ

1 jωj < ωc
πn
0 ωc < jωj < π
sin ωðLþ1Þ=2 jωL=2
1 0nL e
sin ω=2
0 otherwise
X1
ejω0 n 2πδðω ω0 þ 2πk Þ
k¼1
The above theorems concerning DTFT are summarized in Table 7.2

X1 ðπ
1 jω 2
Parseval’ s theorem jxðnÞj2 ¼ X e dω
1 2π π
Using the definitions of DTFT pair given by (7.7) and (7.8), we may establish the
DTFT pairs for some useful functions. These are given in Table 7.3
7.2.3 Some Properties of the DTFT of a Complex Sequence x

(n)
From Eq. (7.7), the DTFT of a time reversed sequence x(n) can be written as
X1 X1
F½xðnÞ ¼ n¼1
xðnÞejωn ¼ l¼1
xðlÞejωl ¼ X ejω ð7:24aÞ
Similarly, the DTFT of the complex conjugate sequence x*(n) can be

expressed as
X1 X1 ∗
F ½ x ∗ ð nÞ ¼ n¼1
x∗ ðnÞejωn ¼ n¼1
xðnÞejωn ¼ X ∗ ejω ð7:24bÞ
From the above two equations, it can be easily shown that

F½x∗ ðnÞ ¼ X ∗ ejω ð7:25Þ
The sequence x(n) can be represented as a sum of conjugate symmetric sequence

xe (n) and a conjugate antisymmetric sequence xo(n) as
xðnÞ ¼ xe ðnÞ þ xo ðnÞ ð7:26Þ
where
1
xe ðnÞ ¼ ½xðnÞ þ x∗ ðnÞ ð7:27Þ
2
and
1
xo ðnÞ ¼ ½xðnÞ x∗ ðnÞ ð7:28Þ
2
The DTFT X(e jω) can be split into

X ejω ¼ X e ejω þ X o ejω ð7:29Þ
where Xe(e jω) and Xo(e jω) are the DTFTs of xe(n) and xo(n), respectively. Using
Eqs. (7.7), (7.25), and (7.27), Xe(e jω) can be expressed as

X e ejω ¼ F½xe ðnÞ
1 1

¼ ðF½xðnÞþF½x∗ ðnÞÞ ¼ X ejω þ X ∗ ejω ¼ Re X ejω ð7:30Þ
2 2
In a similar way, using Eqs. (7.7), (7.25), and (7.28), Xo(e jω) can be written as
X o ðejω Þ ¼ F½xo ðnÞ

1 1
¼ ðF½xðnÞ F½x∗ ðnÞ ¼ ½Xðejω Þ X ∗ ðejω Þ ¼ jIm½Xðejω Þ
2 2
ð7:31Þ
A complex sequence x(n)can be decomposed into a sum of its real and imaginary
parts as
xðnÞ ¼ xR ðnÞ þ jxI ðnÞ ð7:32Þ

where
1
xR ðnÞ ¼ ½xðnÞ þ x∗ ðnÞ ð7:33Þ
2
and
1
jxI ðnÞ ¼ ½xðnÞ x∗ ðnÞ ð7:34Þ
2
The DTFT of xR (n) can be written as

1
F½ReðxðnÞ ¼ F ðxðnÞ þ x∗ ðnÞÞ
2 ð7:35Þ
1
jω
¼ X e þ X ∗ ejω
2
Similarly, the DTFT of jxI (n) can be expressed as

1
F½jImðxðnÞÞ ¼ F ðxðnÞ x∗ ðnÞÞ
2 ð7:36Þ
1
¼ ½Xðe Þ X ∗ ðejω Þ
jω
2
The above properties of the DTFT of a complex sequence are summarized in
Table 7.4.
7.2.4 Some Properties of the DTFT of a Real Sequence x(n)
Since e–jωn ¼ cosωn – jsinωn, the DTFT X(e jω) given by Eq. (7.7) can be expressed as
X1 X1
X ejω ¼ n¼1
x ð n Þ cos ωn j n¼1
xðnÞ sin ωn ð7:37Þ
The Fourier transform X(e jω) is a complex function of ω and can be written as the
sum of the real and imaginary parts as
Table 7.4 Some properties of Sequence DTFT

DTFT of a complex sequence
x∗(n) X∗(ejω)
x∗(n) X∗(e jω)
xR(n) ¼ Re [x(n)] ∗ jω
2 ½X ðe Þ þ X ðe Þ
1 jω
jxI(n) ¼ j Im [x(n)] 2 ½X ðe Þ
1 jω
X ∗ ðejω Þ
∗ jω
xe ðnÞ ¼ 2 ½xðnÞ þ x ðnÞ
1 Re[X(e )]
∗ j Im [X(e jω)]
x0 ðnÞ ¼ 2 ½xðnÞ þ x ðnÞ
1

X ejω ¼ X R ejω þ j X I ejω ð7:38Þ
From Eq. (7.37), the real and imaginary parts of X(e jω) are given by
X1
X R ejω ¼ n¼1
xðnÞ cos ωn ð7:39Þ
and
X1
X I ejω ¼ n¼1
xðnÞ sin ωn ð7:40Þ
Since cos(ωn) ¼ cosωn and sin(ωn) ¼ sinωn, we can obtain the following
relations from Eqs. (7.39) and (7.40):
X1 jω
X R ejω ¼ n¼1
x ð n Þ cos ωn ¼ X R e ð7:41aÞ
X1
X I ejω ¼ n¼1
xðnÞ sin ωn ¼ X I ejω ð7:41bÞ
indicating that the real part of DTFT is an even function of ω, while the imaginary
part is an odd function of ω. Thus,

X ejω ¼ X ∗ ejω ð7:42Þ
In polar form, X(e jω) can be written as

X ejω ¼ X ejω ejθω ð7:43Þ
where
jω qffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffi
ffi
X e ¼ ½X R ðejω Þ2 þ ½X I ðejω Þ2 ð7:44Þ
and
X I ðejω Þ
θðωÞ ¼ ∠X ejω ¼ phase of X ejω ¼ tan 1 ð7:45Þ
X R ðejω Þ
Using the above relations, it can easily be seen that |X (e jω)| is an even function of
ω, whereas the function θ(ω)is an odd function of ω.
Now, the DTFT of xe (n), the even part of the real sequence x(n) is given by
1 1

F½xe ðnÞ ¼ ðF½xðnÞ þ F½xðnÞÞ ¼ X ejω þ X ejω ¼ X R ejω ð7:46Þ
2 2
Thus, the DTFT of even part of a real sequence is the real part of X (e jω).
Similarly, the DTFT of xo (n), the odd part of the real sequence x(n), is given by
Table 7.5 Some properties of

F½xðnÞ ¼ X ðejω Þ ¼ X R ðejω Þ þ jX I ðejω Þ
DTFT of a real sequence
F½xe ðnÞ ¼ X R ðejω Þ
F½xo ðnÞ ¼ jX I ðejω Þ
XR(e jω) ¼ XR(ejω)
XI(e jω) ¼ XI(ejω)
X(e jω) ¼ X∗(ejω)
|X(e jω)| ¼ |X(ejω)|
∠X(e jω) ¼ ∠ X(ejω)
1
jω
F½xo ðnÞ ¼ X e X ejω ¼ jX I ejω ð7:47Þ
2
Hence, the DTFT of the odd part of a real sequence is jXI (ejω).
The above properties of the DTFT of a real sequence are summarized in
Table 7.5.
Example 7.3 A causal LTI system is represented by the following difference
equation:
yðnÞ ayðn 1Þ ¼ xðn 1Þ
(i) Find the impulse response of the system h(n), as a function of parameter a.
(ii) For what range of values would the system be stable?
Solutions (i) Given
yðnÞ ayðn 1Þ ¼ xðn 1Þ
Taking Fourier transform on both sides of above equation, we get

Y ejω aejω Y ejω ¼ ejω X ejω
From the above relation, we arrive at
Y ðejω Þ ejω
H ejω ¼ ¼
X ðejω Þ 1 aejω
P P
F½an uðnÞ ¼ 1 n¼1 a e
n jωn
¼ 1n¼1 ðae
jω n
Þ
1
¼
1 aejω
From the above equation and time shifting property, the impulse response is
given by

ejω
hðnÞ ¼ F1 H ejω ¼ F1 ¼ an1 uðn 1Þ
1 aejω
(ii) Now,
X1 X1
n¼1
j hð nÞ j ¼ n¼1
jajn1 < 1 for jaj < 1:
Thus, the system is stable for |a| < 1.

Example 7.4 Find the impulse response of a system described by the following
difference equation:
5 1 1
y ð nÞ y ð n 1Þ þ y ð n 2Þ ¼ x ð n 1Þ
6 6 3
Solution Taking Fourier transform on both sides of given difference equation, we

get
5 1 1
Yðejω Þ ejω Yðejω Þ þ e2jω Yðejω Þ ¼ ejω Xðejω Þ
6 6 3
From the above relation, we arrive at
Y ðejω Þ ð1=3Þejω
H ðejω Þ ¼ ¼
X ðe Þ 1 ð5=6Þejω þ ð1=6Þe2jω
jω
2 2
¼
1 ð1=2Þejω 1 ð1=3Þejω
The impulse response h(n) is given by

2 2
hðnÞ ¼ F1 F1
1 ð1=2Þejω 1 ð1=3Þejω

n n
¼ 2 12 13 uðnÞ
Example 7.5 Find the DTFT of xðnÞ ¼ ðn!nþm1 Þ! n

ðm1Þ! a uðnÞ, jaj < 1
Solution Let x1(n) ¼ anu(n)

The Fourier transform of x1(n) is given by
X 1 X
1 n 1
X 1 ejω ¼ ðaÞn ejωn ¼ aejω ¼
n¼0 n¼0
1 aejω
For m ¼ 2,
xðnÞ ¼ ðn þ 1Þan uðnÞ
Using the differentiation property of DTFT, the Fourier transform of nanu(n) is

given by

dX 1 ðejω Þ d 1 aejω
j ¼j ¼
dω dω 1 aejω ð1 aejω Þ2
Using linearity property of the DTFT, the Fourier transform of x(n) is denoted by
aejω 1 1
X ejω ¼ þ ¼
ð1 aejω Þ2 ð1 aejω Þ ð1 aejω Þ2
For m ¼ 3,

ðn þ 2Þðn þ 1Þ n n2 þ 3n þ 2 n
x ð nÞ ¼ a uð nÞ ¼ a uð nÞ
2 2
1
2 n
¼ n a uðnÞ þ 3nan uðnÞ þ 2an uðnÞ
2
Using the differentiation and linearity properties of DTFT, the Fourier transform
of x(n) is given by
" ! #
1 d aejω 3aejω 2
X ðe Þ ¼ j
jω
þ þ
2 dω ð1 aejω Þ2 ð1 aejω Þ2 ð1 aejω Þ
" #
1 aejω ð1 þ aejω Þ 3aejω 2
¼ þ þ
2 ð1 aejω Þ3 ð1 aejω Þ2 ð1 aejω Þ
" #
1 2 1
¼ ¼
2 ð1 aejω Þ3 ð1 aejω Þ3
In general, for m ¼ k, the Fourier transform of x(n) is given by
1
X ejω ¼ , where k is any integer value:
ð1 aejω Þk
Example 7.6 Let G1(e jω) denote the DTFT of the sequence g1(n) shown in Figure 7.1
(a). Express the DTFT of the sequence g2(n) in Figure 7.1b in terms of G1(e jω). Do not
evaluate G1(e jω).
Solution From Figure 7.1(b), g2(n) can be expressed in terms of g1(n) as
g2 ð nÞ ¼ g1 ð nÞ þ g1 ð n 4Þ
Applying DTFT on both sides, we obtain

G2 ejω ¼ G1 ejω þ ej4ω G1 ejω ¼ 1 þ ej4ω G1 ejω
4 g1 (n) g 2 ( n)
3
2
1
0 1 2 3 n 0 1 2 3 4 5 6 7 n
(a) (b)
Figure 7.1 (a) Sequence g1(n). (b) Sequence g2(n)
Example 7.7 Evaluate the inverse DTFT of each of the following DTFTs:
P
1
αe jω
(a) X 1 ðejω Þ ¼ δðω þ 2πkÞ (b) X 2 ðejω Þ ¼ ð1αe jω Þ2
, jα j < 1
k¼1
X 1
Solution (a) X 1 ejω ¼ δðω þ 2πk Þ
k¼1
From Table 7.3,

X1
Fð1Þ ð1 < n < 1Þ ¼ k¼1
2πδðω þ 2πkÞ
Hence,
1
F1 ½δðω þ 2πkÞ ¼ , ð1 < n < 1Þ
2π
jω
αe
(b) X 2 ðejω Þ ¼ ð1αe jω Þ2
, jαj < 1
From Example 7.5,
1 ðn þ m 1Þ! n
$ α uð nÞ
ð1 αejω Þm n!ðm 1Þ!
For m ¼ 2,
1 ðn þ 1Þ! n
$ α uð nÞ
ð1 αejω Þ2 n!ð1Þ!
1
$ ðn þ 1Þαn uðnÞ
ð1 αejω Þ2
· 4
· 3
1 · 1
0 1 5
· ·
-3 -2 -1 2 3 4
· -1
-2 ·
· -3
Figure 7.2 A length-9 sequence x(n)
Then
α
$ ðn þ 1Þαnþ1 uðnÞ
ð1 αejω Þ2
αejω
$ nαn uðn 1Þ
ð1 αejω Þ2
Example 7.8 A length-9 sequence x(n) is shown in Figure 7.2

If the DTFT of x(n) is X(e jω), calculate the following functions without comput-
ing X(e jω).
ðπ ðπ ðπ
jω 2 dX ðejω Þ 2
j0
(a) X(e ) jπ
(b) X(e ) (c) jω
X e dω (d) X e dω (e)
d dω
π π π
Solution From the given data,

x(3) ¼3, x(2) ¼0, x(1) ¼ 1, x(0) ¼ 2, x(1) ¼ 3, x (2) ¼ 4, x (3) ¼ 1, x
(4) ¼ 0, x(5) ¼ 1
(a) X(e j0)
From the definition of Fourier transform,
X
1
X ðejω Þ ¼ xðnÞejωn
n¼1
X
1
X ðej0 Þ ¼ x ð nÞ
n¼1
¼ ½ 3 þ 0 þ 1 2 3 þ 4 þ 1 þ 0 1 ¼ 3
(b) X(e jπ)

From the definition of Fourier transform,
X
1
X ðejπ Þ ¼ xðnÞejπ
n¼1
X
1
X ðejπ Þ ¼ xðnÞ ¼ 3
n¼1
ðπ

(c) X ejω dω
π
From the definition of inverse Fourier transform,

ðπ
1
x ð nÞ ¼ X ejω ejωn dω
2π π
Hence,
ðπ

X ejω ejωn dω ¼ 2πxð0Þ ¼ 4π
π
ðπ
jω 2
(d) X e dω
π
From the definition of Parseval’s theorem,
X
1 ðπ
1 jω 2
2
j x ð nÞ j ¼ X e dω
n¼1
2π
π
Hence,
Ðπ 2 P1
π jX ðejω Þj dω ¼ 2π n¼1 jxðnÞj2
¼ 2π ð9 þ 0 þ 1 þ 4 þ 9 þ 16 þ 1 þ 0 þ 1Þ ¼ 82π
ðπ
dX ðejω Þ 2
(e)
dω dω
π
From differentiation property and Parseval’s theorem of DTFT,

( )
H1 e jw
( )
H 2 e jw
1
1
p/2 w p /3 p
(a) (b)
Figure 7.3 (a) Fourier transform of h1(n) (b) Fourier transform of h2 (n)
( )
H1 e jw
( )
H 2 e jw
1
1
p /3 w p /2 p
(a) (b)
Figure 7.4 (a) Fourier transform of h1(n) (b) Fourier transform of h2(n)
ðπ
X1
dX ðejω Þ 2
jnxðnÞj2
dω dω ¼ 2π
n¼1
π
¼ 2π ½81 þ 0 þ 1 þ 0 þ 9 þ 64 þ 9 þ 0 þ 25 ¼ 189π
Example 7.9 (a) The Fourier transforms of the impulse responses, h1(n) and h2 (n),
of two LTI systems are as shown in Figure 7.3. Find the Fourier
transform of the impulse response of the overall system, when they
are connected in cascade.
(b) The Fourier transforms of the impulse responses h1(n) and h2(n) of two LTI
systems are as shown in Figure 7.4. Find the Fourier transform of the overall
system, when they are connected in parallel.
Solution (a) The impulse response h(n) of the overall system is given by
hðnÞ ¼ h1 ðnÞ∗h2 ðnÞ

Figure 7.5 (a) Fourier ( )

H e Jw
transform of the impulse
response of the cascade
system (b) Fourier transform
of the impulse response of 1
the parallel system
p p w
3 2
(a)
( )
H e Jw
1.0
p p w
3 2
(b)
Then, by the convolution property of the Fourier transform, the Fourier transform
of the impulse response of the cascade system is given by

H 1 ejω H 2 ejω
The Fourier transform of impulse response of the cascade system is shown in

Figure 7.5(a).
(b) The impulse response h(n) of the overall system is given by
hð nÞ ¼ h1 ð nÞ þ h 2 ð nÞ
Hence, the Fourier transform of impulse response of the cascade system is

given by

H 1 ejω þ H 2 ejω
The Fourier transform of the impulse response of the parallel system is shown in
Figure 7.5(b).
7.3 Frequency Response of Discrete-Time Systems
For an LTI discrete-time system with impulse response h(n) and input sequence x(n),
the output y(n) is the convolution sum of x(n) and h(n) given by
X
1
y ð nÞ ¼ hðkÞxðn k Þ ð7:48Þ
k¼1
To demonstrate the eigenfunction property of complex exponential for discrete-

time systems, consider the input x(n) of the form
xðnÞ ¼ ejωn , 1 < n < 1 ð7:49Þ
Then from Eq. (7.48), the output is given by

!
X
1 X
1
jωðnk Þ jωk
y ð nÞ ¼ hðkÞe ¼ hðkÞe ejωn ð7:50Þ
k¼1 k¼1

yðnÞ ¼ H ejω ejωn , ð7:51aÞ
where
X
1
H ejω ¼ hðnÞejωn : ð7:51bÞ
n¼1
H(e jω) is called the frequency response of the LTI system whose impulse
response is h(n), e jωn is an eigenfunction of the system, and the associated eigen-
value is H(e jω). In general H(e jω) is complex and is expressed in terms of real and
imaginary parts as

H ejω ¼ H R ejω þ jH I ejω ð7:52Þ
where HR(e jω) and HI(e jω) are the real and imaginary parts of H(e jω), respectively.
Furthermore, due to convolution, the Fourier transforms of the system input and
output are related by

Y ejω ¼ H ejω X ejω ð7:53Þ
where X(e jω) and Y(e jω) are the Fourier transforms of the system input and output,
respectively. Thus,
Y ðejω Þ
H ejω ¼ ð7:54Þ
X ðejω Þ
7.3 Frequency Response of Discrete-Time Systems 333
The frequency response function H(e jω) is also known as the transfer function of
the system. The frequency response function provides valuable information on the
behavior of LTI systems in the frequency domain. However, it is very difficult to
realize a digital system since it is a complex function of the frequency variable ω. In
polar form, the frequency response can be written as

H ejω ¼ H ejω ejθðωÞ ð7:55aÞ
where |H(e jω)|, the amplitude response term, and θ(ω), the phase-response term, are
given by
jHðejω Þj2 ¼ jH R ðejω Þj2 þ jH I ðejω Þj2 ð7:55bÞ

1 H I ðe Þ
jω
θðωÞ ¼ tan ð7:55cÞ
H R ðejω Þ
Phase and Group Delays

If the input is a sinusoidal signal given by
xðnÞ ¼ cos ðωnÞ, for 1 < n < 1, ð7:56aÞ
then from Eq. (7.55a), the output is

y½n ¼ H ejω0 cos ðωn þ θðωÞÞ ð7:56bÞ

jω θ ð ωÞ

y ½ n ¼ H e 0
cos ω n þ , ð7:57aÞ
ω
jω
¼ H e 0 cos ω n τp ðωÞ
It can be clearly seen that the above equation expresses the phase response as a
time delay in seconds which is called as phase delay and is defined by
θ ð ωÞ
τ P ð ωÞ ¼ ð7:57bÞ
ω
An input signal consisting of a group of sinusoidal components with frequencies
within a narrow interval about ω experiences different phase delays when processed
by an LTI discrete-time system. As such, the signal delay is represented by another
parameter called group delay defined as
dθðωÞ
τ g ð ωÞ ¼ ð7:57cÞ
dω
Example 7.10 Determine the magnitude

n and phase response of a system whose
impulse response is given by hðnÞ ¼ 12 uðnÞ
n
Solution For hðnÞ ¼ 12 uðnÞ, the frequency response is given by
X1 n X1
1 jωn 1 jω n
H ðe Þ ¼
jω
e ¼ e
n¼1
2 n¼1
2
1 1
¼ jω
¼
1 0:5e 1 0:5 cos ω þ j0:5 sin ω
The magnitude response is given by
jω 1 1
H e ¼ qffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffi
ffi ¼ r
ffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffi
ð1 0:5 cos ωÞ2 þ ð0:5Þ2 sin 2 ω 1 þ ð0:5Þ2 2ð0:5Þ cos ω
The phase response is
0:5 sin ω
θðωÞ ¼ tan 1
1 0:5 cos ω
The magnitude and phase values are tabulated in Table 7.6 for various values of ω
and plotted in Figure 7.6(a) and (b), respectively.
Table 7.6 Magnitude and phase

π π
ω 0 4 2
3π
4 π 5π
4
3π
2
7π
4 2π
|H(e jω)| 2 1.3572 0.8944 0.7148 0.67 0.715 0.894 1.3572 2
θ(ω) 00 28.675 26.565 14.640 00 14.640 26.565 28.6750 00
2 30
1.8 20
1.6
Phase, degrees
10
Magnitude
1.4
0
1.2
-10
1
0.8 -20
-30
0 0.2 0.4 0.6 0.8 1 1.2 1.4 1.6 1.8 2 0 0.2 0.4 0.6 0.8 1 1.2 1.4 1.6 1.8 2
ω/π ω/π
(a) (b)
Figure 7.6 (a) Magnitude and (b) phase responses of h(n) of Example 7.10
Figure 7.7 (a) Impulse h2 (n) h1 (n)

response of h1(n)
3 3
(b) impulse response of 2 2 2 2
h2(n) 1 1 1 1
n n
0 2 4 -2 0 2
(a) (b)
Example 7.11 Compute the magnitude and phase responses of the impulse
responses given in Figure 7.7, and comment on the results.
Solution Since h1(n) is an even function of time, it has a real DTFT indicating that
the phase is zero, that is, the phase is a horizontal line; h2(n) is the right-shifted
version of h1(n). Hence, from time shifting property of DTFT, the transform of h2(n)
is obtained by multiplying the transform of h1(n) by e–j2ω. This changes the slope of
the phase linearly and can be verified as follows:
The frequency response of h1(n) is
H 1 ðejω Þ ¼ e2jω þ 2ejω þ 3 þ 2ejω þ e2jω

¼ ðe2jω þ e2jω Þ þ 2ðejω þ ejω Þ þ 3 ¼ 2 cos 2ω þ 4 cos ω þ 3
The magnitude response of H1(e jω) is

jω
H 1 e ¼ 2 cos 2ω þ 4 cos ω þ 3
The phase response of H1(e jω) is zero.

The frequency response of h2(n) is
H 2 ðejω Þ ¼ e2jω H 1 ðejω Þ

¼ e2jω ð2 cos 2ω þ 4 cos ω þ 3Þ
The magnitude response of H2(e jω) is

jω
H 2 e ¼ 2 cos 2ω þ 4 cos ω þ 3
The phase response of H2(e jω) is given by

∠H 2 ejω ¼ ∠e2jω ¼ 2ω:
The magnitude and phase responses of h1(n) and h2(n) are shown in Figure 7.8(a),
(b), (c), and (d). From the magnitude and phase responses of h1(n) and h2(n), it is
observed that h1(n) has zero phase and h2(n) has a linear phase response, whereas
both h1(n) and h2(n) have the same magnitude responses.
Figure 7.8 (a) Magnitude response of h1(n). (b) Phase response of h1(n). (c) Magnitude response
of h2(n). (d) Phase response of h2(n)
Example 7.12 The trapezoidal integration formula is represented by a recursive

difference equation as y(n) – y(n – 1) ¼ 0.5x(n) þ 0.5x(n – 1). Determine H(e jω) of
the trapezoidal integration formula.
Solution Given
yðnÞ yðn 1Þ ¼ 0:5xðnÞ þ 0:5xðn 1Þ

Taking Fourier transform on both sides of the above equation, we get
Y ðejω Þ ejω Y ðejω Þ ¼ 0:5X ðejω Þ þ 0:5ejω X ðejω Þ

Y ðejω Þð1 ejω Þ ¼ 0:5X ðejω Þð1 þ ejω Þ
Y ðejω Þ ð1 þ ejω Þ
H ejω ¼ ¼ 0:5
X ðejω Þ ð1 ejω Þ
" #
ejω=2 ejω=2 þ ejω=2 cos ðω=2Þ
¼ 0:5 jω=2 jω=2 ¼ j0:5
e ðe ejω=2 Þ sin ðω=2Þ
The magnitude response is given by

jω
H e ¼ 0:5 cos ðω=2Þ
sin ðω=2Þ
The phase response is given as follows:

If
π
0 < ω < π, then both cos ω/2 and sin ω/2 are positive, and hence the phase is
2 .

If π < ω < 2π, then cos ω/2 is negative, but sin ω/2 is positive; hence the phase is π2 .
7.3.1 Frequency Response Computation Using MATLAB
The M-file function freqz(h, w) in MATLAB can be used to determine the values of
the frequency response of an impulse response vector h at a set of given frequency
points ω. Similarly, the M-file function freqz(b, a, ω) can also be used to find the
frequency response of a system described by the recursive difference equation with
the coefficients in vectors b and a. From frequency response values, the real and
imaginary parts can be computed using MATLAB functions real and imag, respec-
tively. The magnitude and phase of the frequency response can be determined using
the functions abs and angle as illustrated in the following examples:
Example 7.13 Determine the magnitude and phase response of a system described
by the difference equation, y(n) ¼ 0.5x(n) þ 0.5x(n – 2).
Solution If x(n) ¼ δ(n), then the impulse response h(n) is given by
hðnÞ ¼ 0:5δðnÞ þ 0:5δðn 2Þ
Hence, h(n) sequence is [0.5 0 0.5]. When this sequence is used in Program 7.1
given below, the resulting magnitude and phase responses are as shown in Figure 7.9
(a) and (b), respectively.
Figure 7.9 (a) Magnitude response of h(n) sequence. (b) Phase response of h(n) sequence
Program 7.1
clear;clc;
w=0:0.05:pi;
h=exp(j*w); %set h=exp(jw)
num=0.5+0*h.^-1+0.5*h.^-2;
den=1;
%Compute the frequency responses
H=num/den;
%Compute and plot the magnitude response
mag=abs(H);
figure(1),plot(w/pi,mag);
ylabel(‘Magnitude’);xlabel(‘\omega/\pi’);
%Compute and plot the phase responses
ph=angle(H)*180/pi;
figure(2),plot(w/pi,ph);
ylabel(‘Phase, degrees’);
xlabel(‘\omega/\pi’)
Example 7.14 Determine the magnitude and phase responses of a system described
by the following difference equation:
yðnÞ 2:1291yðn 1Þ þ 1:7834yðn 2Þ 0:5435yðn 3Þ

¼ 0:0534xðnÞ 0:0009xðn 1Þ 0:0009xðn 2Þ þ 0:0534xðn 3Þ
Comment on the frequency response of the system.

Solution The following MATLAB program 7.2 is used and the resultant magnitude
response and phase response are shown in Figure 10(b) and (b), respectively.
Program 7.2
clear;close all;
num=[0.0534 -0.0009 -0.0009 0.0534];% numerator coefficients
den=[1 -2.1291 1.7834 -0.5435];% denominator coefficients
w=0:pi/255:pi;
%Compute the frequency responses
H=freqz(num,den,w);
%Compute and plot the magnitude response
mag=abs(H);
figure(1),plot(w/pi,mag);
ylabel(‘Magnitude’);xlabel(‘\omega/\pi’);
%Compute and plot the phase responses
ph=angle(H)*180/pi;
figure(2),plot(w/pi,ph);
ylabel(‘Phase, degrees’);xlabel(‘\omega/\pi’);
Figure 7.10 (a) Magnitude response (b) phase response
The frequency response shown in Figure 7.10 characterizes a low-pass filter with
nonlinear phase.
Example 7.15 Determine the magnitude and phase responses of a system described
by the following difference equation:
yðnÞ 3:0538yðn 1Þ þ 3:8281yðn 2Þ 2:2921yðn 3Þ þ 0:5507yðn 4Þ

¼ xðnÞ 4xðn 1Þ þ 6xðn 2Þ 4xðn 3Þ þ xðn 4Þ:
Comment on the frequency response of the system.

Solution Program 7.2 with variables num ¼ [ 1 4 6 4 1] and den ¼ [1 3.0538
3.8281 2.2921 0.5507] is used, and the resultant magnitude and phase responses
are shown in Figure 7.11(a) and (b), respectively. It is observed from this figure that
the frequency response characterizes a narrowband band-pass filter.
Example 7.16 An LTI system is described by the following difference equation:
yðnÞ ¼ xðnÞ þ 2xðn 1Þ þ xðn 2Þ
(a) Find the frequency response H(e jω) and group delay grd [H(e jω)] of the system.
(b) Determine the difference equation of a new system such that the frequency
response H1(e jω) of the new system is related to H(e jω) as H1(e jω) ¼ H(e j(ω þ π)).
Solution (a)
yðnÞ ¼ xðnÞ þ 2xðn 1Þ þ xðn 2Þ

hðnÞ ¼ δðnÞ þ 2δðn 1Þ þ δðn 2Þ
H ðejω Þ ¼ 1 þ 2ejω þ e2jω

1 1
¼ 2ejω ðejω Þ þ 1 þ ðejω Þ
2 2
¼ 2ejω ð cos ω þ 1Þ
Hence,
jH ðejω Þj ¼ 2ð cos ω þ 1Þ
∠H ðejω Þ ¼ ω
Therefore,

d∠H ðejω Þ
group delay ¼ grad H ejω ¼ ¼1
dω
(b) By frequency shifting property, ejπnh(n) $ H(e j(ω+π)). Therefore,
h1 ðnÞ ¼ ejπn hðnÞ ¼ ð1Þn hðnÞ

¼ δðnÞ 2δðn 1Þ þ δðn 2Þ
Hence, the difference equation of the new system is
yðnÞ ¼ xðnÞ 2xðn 1Þ þ xðn 2Þ:

Figure 7.11 (a) Magnitude response (b) phase response

7.4 Representation of Sampling in Frequency Domain
As mentioned in Section 6.1, mathematically, the sampling process involves multi-

plying a continuous-time signal xa(t) by a periodic impulse train p(t)
X1
pð t Þ ¼ n¼1
δðt nT Þ ð7:58Þ
As a consequence, the multiplication process gives an impulse train xp(t), which

can be expressed as
xp ðt Þ ¼ xa ðt Þpðt Þ
P1 ð7:59Þ
¼ 1 xa ðt Þδðt nT Þ
Since xa(t) δ(t – nT) ¼ xa(nT) δ(t – nT), the above reduces to
X1
xp ð t Þ ¼ x ðnT Þδðt
1 a
nT Þ ð7:60Þ
If we now take the Fourier transform of (7.59), and use the multiplication
property of the Fourier transform, we get
1
X p ðjΩÞ ¼ ½X a ðjΩÞ∗PðjΩÞ ð7:61Þ
2π
where * denotes the convolution in the continuous-time domain and Xp(jΩ), Xa(jΩ),
and P(jΩ) are the Fourier transforms of xp(t), xa(t), and p(t), respectively. Since p(t) is
periodic with a period T, it can be expressed as a Fourier series
1X1
ejð T Þkt
2π
pð t Þ ¼
T 1
Since the Fourier transform of f ðt Þ ¼ ejΩT t is given by F(jΩ) ¼ 2πδ(Ω – ΩT), we

see that the Fourier transform of p(t) is given by
2π X1
PðjΩÞ ¼ δðΩ kΩT Þ ð7:62Þ
T k¼1
where ΩT ¼ 2πT :
Substitution of (7.62) in (7.61) yields
1h X1 i
X p ðjΩÞ ¼ X a ðjΩÞ∗ δð Ω kΩ T Þ ð7:63Þ
T k¼1
Since the convolution of Xa(jΩ) with a shifted impulse δ(Ω – kΩT) is the shifted
function Xa(j(Ω – kΩT)), the above reduces to
1 X1
X p ðjΩÞ ¼ X ðjΩ jkΩT Þ
k¼1 a
ð7:64Þ
T
7.4 Representation of Sampling in Frequency Domain 345
Eq. (7.64) shows that the spectrum of xp(t) consists of an infinite number of
shifted copies of the spectrum of xa(t), and the shifts in frequency are multiples of
ΩT; that is, Xp(jΩ) is a periodic function with a period of ΩT ¼ 2π/T.
Since the continuous Fourier transform of δ(t – nT) is given by
F½δðt nT Þ ¼ ejΩTn , ð7:65Þ
we have from Eq.(7.60) that

X1
jΩTn
X p ðjΩÞ ¼ x ðnT Þe
n¼1 a
ð7:66Þ
Since
xðnÞ ¼ xa ðnT Þ, 1 < n < 1
and the fact that the DTFT of the sequence x(n) is given by
X1
X ejω ¼ n¼1
xðnÞejωn , ð7:67Þ
we obtain

X ejω ¼ X p ðjΩÞ Ω¼ω=T ð7:68aÞ
or equivalently

X p ðjΩÞ ¼ X ejω ω¼ΩT ð7:68bÞ
Hence, we have from (7.68a) and (7.64) that

1 X1 1 X1 ω 2πk
X ejω ¼ X
k1 a
ðjΩ jkΩ T Þ ¼ X
k1 a
j j ð7:69Þ
T Ω¼ω=T T T T
On the other hand, the above equation can also be expressed as
1 X1
X ejΩT ¼ X ðjΩ jkΩT Þ
k1 a
ð7:70Þ
T
From Eq.(7.69) or (7.70), it can be observed that X(e jω) is obtained by frequency
scaling Xp ( jΩ) using Ω ¼ ω/T.
As mentioned earlier, the continuous-time Fourier transform Xp(jΩ) is periodic
with respect to Ω having a period of ΩT ¼ (2π/T). In view of the frequency scaling,
the DTFT X(e jω) is also periodic with respect to ω with a period of 2π.
(a) (b)
Figure 7.12 (a) Spectrum of an analog signal (b) spectrum of the pulse train
Figure 7.13 Spectrum of an undersampled signal, showing aliasing (fold-over region). Signals in
the fold-over region are not recoverable
7.4.1 Sampling of Low-Pass Signals
Sampling Theorem
If the highest component of frequency in analog signal xa(t) is Ωm, then xa(t) is
uniquely determined by its samples xa(nT), provided that
ΩT 2Ωm ð7:71Þ
where ΩT is called the sampling frequency in radians. Eq. (7.71) is often referred as
the Nyquist condition.
The spectra of the analog signal xa(t) and the impulse train p(t) with a sampling
period T ¼ 2π/ΩT are shown in Figure 7.12(a) and (b), respectively.
Undersampling
If ΩT < 2Ωm, then the signal is undersampled, and the corresponding spectrum
Xp(jΩ) is as shown in Figure 7.13. In this figure, the image frequencies centered at
ΩT will alias into the baseband frequencies, and the information of the desired signal
is indistinguishable from its image in the fold-over region.
7.5 Reconstruction of a Band-Limited Signal from Its Samples 347
Figure 7.14 Spectrum of an oversampled signal
Oversampling
If ΩT > 2Ωm, then the signal is oversampled, and its spectrum is shown in Fig-
ure 7.14. Its spectrum is the same as that of the original analog signal, but repeats
itself at every multiple of ΩT. The higher-order components centered at multiples of
ΩT are called image frequencies.
7.5 Reconstruction of a Band-Limited Signal

from Its Samples
According to the sampling theorem, samples of a continuous-time band-limited

signal (i.e., its Fourier transform Xa(jΩ) ¼ 0 for |Ω| > |Ωm|) taken frequently enough
are sufficient to represent the signal exactly. The original continuous-time signal
xa(t) can be fully recovered by passing the modulated impulse train xp(t) through an
ideal low-pass filter, HLP(jΩ), whose cutoff frequency satisfies Ωm Ωc ΩT/2.
Consider a low-pass filter with a frequency response:

T jΩj Ωc
H LP ðjΩÞ ¼ ð7:72Þ
0 jΩj > Ωc
Applying the inverse continuous-time Fourier transform to HLP(jΩ), we obtain the

impulse response hLP(t) of the ideal low-pass filter given by
ð1 ð Ωc
1 T sin ðΩc tÞ
hLP ðtÞ ¼ H LP ðjΩÞe dΩ ¼
jΩt
ejΩt dΩ ¼ , 1<t <1
2π 1 2π Ωc ðπt=TÞ
ð7:73Þ
For a given sequence of samples x(n), we can form an impulse train xp(t) in which
successive impulses are assigned an area equal to the successive sequence values, i.e.,
X1
xp ð t Þ ¼ n¼1
xðnÞδðt nT Þ ð7:74Þ
The nth sample is associated with the impulse at t ¼ nT, where T is the sampling
period associated with the sequence x(n). Therefore, the output xa(t) of the ideal
low-pass filter is given by the convolution of xp(t) with the impulse response hLP(t) of
the analog low-pass filter:
xa ( t ( x (n( y ( n( yr (t (
ADC H ( e Jw ( DAC
T1 = 0.0001sec T2 = 0.0001sec
(a)
X a ( jW)
1
-10000p 10000p W
(b)
Figure 7.15 (a) Discrete time system (b) spectrum of input xa(t)
X1
xa ð t Þ ¼ n¼1
xðnÞhLP ðt nT Þ ð7:75Þ
Substituting hLP(t) from Eq.(7.73) in Eq. (7.75) and assuming for simplicity that
Ωc ¼ ΩT/2 ¼ π/T, we get
X1 sin ½π ðt nT Þ=T
xa ðt Þ ¼ x ð nÞ ð7:76Þ
n¼1 π ðt nT Þ=T
The above expression indicates that the reconstructed continuous-time signal xa(t)
is obtained by shifting in time the impulse response hLP(t) of the low-pass filter by an
amount nT and scaling it in amplitude by the factor x(n) for all integer values of n in
the range –1 < n < 1 and then summing up all the shifted versions.
Example 7.17 Consider the system shown in Figure 7.15(a), where H(e jω) is an
ideal LTI low-pass filter with cutoff of π /8 rad/sec, and the spectrum of xa(t) is shown
in Figure 7.15(b).
(i) What is the maximum value of T to avoid aliasing in the ADC?
(ii) If 1/T ¼ 10 kHz, then what will be the spectrum of yr(t).
Solution (i) From Figure 7.15(b), Ωm ¼ 10 k π.

The given T1 ¼ 0.0001 sec. Then ΩT ¼ T2π1 ¼ 20 kπ.
The condition to avoid aliasing in the ADC is ΩT ¼ 2Ωm (Figure 7.16)
(ii) T ¼ 10K
1
¼ 0:0001 sec
7.6 Problems 349
Figure 7.16
1
-W -10000 10000 W
(a)
jWT
1 X (e )
T
-W -20000 -10000 10000 20000 W

(b)
X (e jw )
1
T H (e jw )
w 1
·
- -2p -p -p p p 2p w = WT
8 8
(c)
Y (e jw )
1
T
-2p p p 2p w
-
8 8
(d)
Yr (e jWT )
H r ( jW)
T
1
T
2p 10000p 2p
-
8T 8T T
(e)
r
1
p p
-
8 8 T
(f )
7.6 Problems
1. Obtain the DTFS representation of the periodic sequence shown in Figure P7.1
4
4
3 3
2 2
…….
1
1
0 1 2 3 4 5 6 7 8 9 n
Figure P7.1 Periodic sequence with period N ¼ 5
2. Find the Fourier

coefficients in DTFS representation of the sequence
xðnÞ ¼ sin 5π
4 n
3. Find the DTFT for the following sequences:
(a) x1(n) ¼ u(n) – u(n – 5) (b) x2(n) ¼ αn(u(n) – u(n – 8)), |α| < 1
jnj
(c) x3 ðnÞ ¼ n 12 (d) x4(n) ¼ |a|nsin ωn, |α| < 1
4. Let G1(e jω) denote the DTFT of the sequence g1(n) shown in Figure P7.2(a).
Express the DTFTs of the remaining sequences in Figure P7.2 in terms of G1(e jω).
Do not evaluate G1(e jω).
4 g1 (n)
3
2
1
0 1 2 3 n
(a)
g 2 ( n)
g 3 ( n)
0 1 2 3 4 5 6 7 n 0 1 2 3 4 5 6 7 n
Figure P7.2 Sequences g1(n), g2(n), and g3(n)

Further Reading 351
5. Determine the inverse DTFT of each of the following DTFTs:

(a) H1(e jω) ¼ 1 þ 4 cos ω þ 3 cos 2ω
(b) H2(e jω) ¼ (3 þ 2 cos ω þ 4 cos (2ω)) cos (ω/2)ejω/2
(c) H3(e jω) ¼ ejω/4
(d) H4(e jω) ¼ ejω[1 þ 4 cos ω]
6. A continuous-time signal xa(t) has its spectrum Xa(jΩ) as shown in Figure P7.3(a).
The signal xa(t) is input to the system shown in Figure P7.3(b). H(e jω) in
Figure P7.3(b) is an ideal LTI low-pass filter with a cutoff frequency of (π/2).
Sketch the spectrums of x(n), y(n), and yr(t).
- 5000p 5000p W
(a)
xa (t) x(n ) y (n ) yr (t)

ADC ( )
H e Jw DAC
T1 = 0.0001sec T2 = 0.0001sec
(b)
Figure P7.3 (a) Spectrum of signal. (b) Signal reconstruction
Further Reading
1. Morrison, N.: Introduction to Fourier Analysis. Wiley, New York (1994)

3. Oppenheim, A.V., Schafer, W.: Discrete-Time Signal Processing, 2nd edn. Prentice-Hall, Upper
Saddle River (1999)
Chapter 8
The z-Transform and Analysis of Discrete
Time LTI Systems
The DTFT may not exist for all sequences due to the convergence condition, whereas
the z-transform exists for many sequences for which the DTFT does not exist. Also,
the z-transform allows simple algebraic manipulations. As such, the z-transform has
become a powerful tool in the analysis and design of digital systems. This chapter
introduces the z-transform, its properties, the inverse z-transform, and methods for
finding it. Also, in this chapter, the importance of the z-transform in the analysis of
LTI systems is established. Further, one-sided z-transform and the solution of state-
space equations of discrete-time LTI systems are presented. Finally, transformations
between continuous-time systems and discrete-time systems are discussed.
8.1 Definition of the z-Transform
The z-transform of an arbitrary discrete-time signal x(n) is defined as

X1
X ðzÞ ¼ Z ½xðnÞ ¼ n¼1
xðnÞzn ð8:1Þ
where z is a complex variable. For the existence of theX

z-transform, Eq. (8.1) should
1
converge. It is known from complex variables that if n¼1
xðnÞzn is absolutely
convergent, then Eq. (8.1) is convergent. Eq. (8.1) can be rewritten as
X1 X1
X ðzÞ ¼ n¼0
xðnÞzn þ n¼1
xðnÞzn ð8:2Þ
By ratio test, the first series is absolutely convergent if

xðn þ 1Þ zðnþ1Þ xðn þ 1Þ 1

limn !1
¼ limn!1 z < 1
x ð nÞ zn x ð nÞ

354 8 The z-Transform and Analysis of Discrete Time LTI Systems
or

x ð n þ 1Þ

jzj > limn!1 ¼ r 1 ðsayÞ ð8:3aÞ
x ð nÞ
Similarly, the second series in Eq. (8.2) is absolutely convergent if

xðn þ 1Þ 1
limn!1 z < 1
x ð nÞ
or

x ð n þ 1Þ
jzj < limn!1 ¼ r 2 ðsayÞ ð8:3bÞ
x ð nÞ
Thus, in general, Eq. (8.1) is convergent in some annulus
r 1 < j zj < r 2 ð8:4Þ
The set of values of z satisfying the above condition is called the region of
convergence (ROC). It is noted that for some sequences r1 ¼ 0 or r2 ¼ 1. In such
cases, the ROC may not include z ¼ 0 or z ¼ 1, respectively. Also, it is seen that no
z-transform exists if r1 > r2.
The complex variable z in polar form may be written as
z ¼ rejω ð8:5Þ
where r and ω are the magnitude and the angle of z, respectively. Then, Eq. (8.1) can
be rewritten as
X
1 X
1
X rejω ¼ xðnÞðreÞjωn ¼ xðnÞejωn r n ð8:6Þ
n¼1 n¼1
When r ¼ 1, that is, when the contour |z| ¼ 1, a unit circle in the z-plane, then
Eq. (8.5) becomes the DTFT of x(n).
Rational z-Transform
In LTI discrete-time systems, we often encounter with a z-transform which is a ratio
of two polynomials in z:
N ðzÞ b0 þ b1 z1 þ b2 z2 þ þ bM zM

X ðzÞ ¼ ¼ ð8:7Þ
DðzÞ 1 þ a1 z1 þ a2 z2 þ þ aN zN
The zeros of the numerator polynomial N(z) are called the zeros of X(z) and those
of the denominator polynomial D(z) as the poles of X(z). The numbers of finite zeros
and poles in Eq. (8.7) are M and N, respectively. For example, the function
X ðzÞ ¼ ðz1Þzðz2Þ has a zero at z ¼ 0 and two poles at z ¼ 1 and z ¼ 2.
8.2 Properties of the Region of Convergence for the z-Transform 355
8.2 Properties of the Region of Convergence

for the z-Transform
The properties of the ROC are related to the characteristics of the sequence x(n). In
this section, some of the basic properties of ROC are considered.
Property 1: ROC should not contain poles.
In the ROC, X(z) should be finite for all z. If there is a pole p in the ROC, then X(z)
is not finite at this point, and the z-transform does not converge at z ¼ p. Hence, ROC
cannot contain any poles.
Property 2: The ROC for a finite duration causal sequence is the entire z-plane
except for z ¼ 0.
A causal finite duration sequence of length N is such that x(n) ¼ 0 for n < 0 and for
n > N 1. Hence X(z) is of the form
P n
XðzÞ ¼ N1
n¼0 xðnÞz ð8:8Þ
¼ xð0Þ þ xð1Þz1 þ þ xðN 1ÞzNþ1
It is clear from the above expression that X(z) is convergent for all values of
z except for z ¼ 0, assuming that x(n) is finite. Hence, the ROC is the entire z-plane
except for z ¼ 0 and is shown as shaded region in Figure 8.1.
Property 3: The ROC for a noncausal finite duration sequence is the entire
z-plane except for z ¼ 1.
A noncausal finite duration sequence of length N is such that x(n) ¼ 0 for n 0
and for n N. Hence, X(z) is of the form
P1
X ðzÞ ¼ xðnÞzn
n¼N
ð8:9Þ
¼ xðN ÞzN þ þ xð2Þz2 þ xð1Þz
Figure 8.1 ROC of a finite Im(z)

duration causal sequence
Re(z)
Figure 8.2 ROC of a finite Im(z)

duration noncausal
sequence
Re(z)
It is clear from the above expression that X(z) is convergent for all values of
except for z ¼ 1, assuming that x(n) is finite. Hence, the ROC is the entire z-plane
except for z ¼ 1 and is shown as shaded region in Figure 8.2.
Property 4: The ROC for a finite duration two-sided sequence is the entire
z-plane except for z ¼ 0 and z ¼ 1.
A finite duration of length (N2 + N1 þ 1) is such that x(n) ¼ 0 for n < N1 and for
n > N2, where N1 and N2 are positive. Hence, x(z) is of the form
PN 2
X ðzÞ ¼ n¼N 1 xðnÞzn
ð8:10Þ
¼ xðN 1 ÞzN 1 þ þ xð1Þz þ xð0Þ þ xð1Þz1 þ þ xðN 2 ÞzN 2
It is seen that the above series is convergent for all values of z except for z ¼ 0 and
z ¼ 1.
Property 5: The ROC for an infinite duration right-sided sequence is the
exterior of a circle which may or may not include z ¼ 1.
For such a sequence, x(n) ¼ 0 for n < N. Hence, X(z) is of the form
X1
X ðzÞ ¼ n¼N
xðnÞzn ð8:11Þ
If N 0, then the right-sided sequence corresponds to a causal sequence and the

above series converges if Eq (8.3a) is satisfied, that is,

xðn þ 1Þ

jzj > limn!1 ¼ r1 ð8:12Þ
x ð nÞ
Hence, in this case the ROC is the region exterior to the circle |z| ¼ r1 or the region
|z| > r1 including the point at z ¼ 1. However, if N is a negative integer, say,
N ¼ N1, then the series (8.12) will contain a finite number of terms involving
positive powers of z. In this case, the series is not convergent for z ¼ 1, and hence
the ROC is the exterior of the circle |z| ¼ r1 but will not include the point at z ¼ 1.
Figure 8.3 ROC of an Region of

infinite duration causal Im Convergence
sequence
r1 Re
As an example of an infinite duration causal sequence, consider

(
r 1n n 0,
x ð nÞ ¼
0 n < 0:
X
1 X
1 n 1
Then X ðzÞ ¼ r 1n zn ¼ r 1 z1 ¼ ð8:13Þ
n¼0 n¼0
1 r 1 z1
Eq. (8.13) holds only if |r1z1| < 1. Hence, the ROC is |z| > r1. The ROC is
indicated by the shaded region shown in Fig. 8.3 and includes the region |z| > r1. It
can be seen that X(z) has a zero at z ¼ 0 and pole at z ¼ r1. The zero is denoted by O
and the pole by X.
Property 6: The ROC for an infinite duration left-sided sequence is the
interior of a circle which may or may not include z ¼ 0.
For such a sequence, x(n) ¼ 0 for n > N. Hence, X(z) is of the form
XN
X ðzÞ ¼ n¼1
xðnÞzn ð8:14Þ
If N < 0, then the left-sided sequence corresponds to a noncausal sequence and the
above series converges if Eq. (8.3b) is satisfied, that is,

x ð n þ 1Þ

jzj < limn!1 ¼ r2 ð8:15Þ
x ð nÞ
Hence, in this case, the ROC is the region interior to the circle |z| ¼ r2 or the
region |z| < r2 including the point at z ¼ 0.
However, if N is a positive integer, then the series (8.14) will contain a finite
number of terms involving negative powers of z. In this case, the series is not
convergent for z ¼ 0, and hence the ROC is the interior of the circle |z| ¼ r2 but
will not include the point at z ¼ 0.
Figure 8.4 ROC of an Im

infinite duration noncausal
sequence Region of
Convergence
r2 Re
As an example of an infinite duration noncausal sequence, consider

(
0 n 0,
x ð nÞ ¼ ð8:16Þ
r 2n n 1:
Then,
P1 P1
X ðzÞ ¼ n¼1 r n
2 z
n
¼ r 1
2 z
m m
m¼0 r 2 z
1 z ð8:17Þ
X ðzÞ ¼ ¼ for jzj < r 2
1 r 2 z1 z r 2
Hence, the ROC is |z| < r2, that is, the interior of the circle |z| ¼ r2. The ROC and
the pole and zero of X(z) are shown in Fig. 8.4.
Property 7: The ROC of an infinite duration two-sided sequence is a ring in
the z-plane.
In this case, the z-transform X(z) is of the form

X1
X ðzÞ ¼ n¼1
xðnÞzn ð8:18Þ
and converges in the region r1 < |z| < r2, where r1 and r2 are given by (8.3a) and
(8.3b), respectively. As mentioned before, the z-transform does not exist if r1 > r2.
As an example, consider the sequence
(
r 1n n 0,
x ð nÞ ¼ ð8:19Þ
r 2n n < 1:
Then,
z z zð2z r 1 r 2 Þ
X ðzÞ ¼ þ ¼ ð8:20Þ
z r 1 z r 2 ðz r 1 Þ ðz r 2 Þ
Figure 8.5 ROC of an Im Region of

infinite duration two-sided
Convergence
sequence
r1 r2 Re
where the region of convergence is r1< |z| < r2. Thus, the ROC is a ring with a pole on
the interior boundary and a pole on the exterior boundary of the ring, without any
pole in the ROC. There are two zeros, one being located at the origin and the other in
the ROC. The poles and zeros as well as the ROC are shown in Figure 8.5.
Example 8.1 Determine the z-transform and the ROC for the following sequence:
xðnÞ ¼ 2n for n 0
Solution From the definition of the z-transform,
X
1 X
1 X
1 n
X ðzÞ ¼ xðnÞzn ¼ 2n zn ¼ 2 z1
n¼1 n¼0 n¼0
1 1
¼ , 2z < 1
1 2z1
Thus, the ROC is |z| > 2.
Example 8.2 Determine the z-transform and the ROC for the following sequence:
8 n
>
> 1
< 5
> for n 0
x ð nÞ ¼ n
>
> 1
>
: for n < 0
3
X
1 X1 X 1 n
n 1 n n 1
Solution X ðzÞ ¼ xðnÞz ¼ z þ zn
n¼1 n¼0
5 n¼1
3

1 1 1 1
¼ þ , for < j z j and j z j <
1 þ ð1=5Þz1 1 ð1=3Þz1 5 3
respectively.
Thus, the ROC is 15 < jzj < 13
8.3 Properties of the z-Transform
Properties of the z-transform are very useful in digital signal processing. Some
important properties of the z-transform are stated and proved in this section. We
will denote in the following the ROC of X(z) by R(r1 < |z| < r2) and those of X1(z) and
X2(z) by R1 and R2, respectively. Also, the region (1/(r2) < |z| < 1/(r1) is denoted by
(1/R).
Linearity If x1(n) and x2(n) are two sequences with z-transforms X1(z) and X2(z)
and ROCs R1 and R2, respectively, then the z-transform of a linear combination of
x1(n) and x2(n) is given by
Zfa1 x1 ðnÞ þ a2 x2 ðnÞg ¼ a1 X 1 ðzÞ þ a2 X 2 ðzÞ ð8:21Þ
whose ROC is at least (R1 \ R1) and a1 and a2 being arbitrary constants.
Proof
P1
Zfa1 x1 ðnÞ þ a2 x2 ðnÞg ¼ n¼1 fa1 x1 ðnÞ þ a2 x2 ðnÞgzn
P P ð8:22Þ
¼ a1 1n¼1 x1 ðnÞz
n
þ a2 1 n¼1 x2 ðnÞz
n
¼ a1 X 1 ð z Þ þ a2 X 2 ð z Þ ð8:23Þ
The result concerning the ROC follows directly from the theory of complex
variables concerning the convergence of a sum of two convergent series.
Time Reversal If x(n) is a sequence with z-transform X(z) and ROC R, then the
z-transform of the time reversed sequence x(n) is given by

Z fxðnÞg ¼ X z1 ð8:24Þ
whose ROC is 1/R.

Proof From the definition of the z-transform, we have
P1 P1
Z ½xðnÞ ¼ n¼1 xðnÞzn ¼ m¼1 xðmÞ zm
P1 m ð8:25Þ
¼ m¼1 xðmÞðz1 Þ
Hence,

Z ½xðnÞ ¼ X z1 ð8:26Þ
Since (r1 < |z| < r2), we have (1/(r2) < |z1| < 1/(r1)). Thus, the ROC of Z [x(n)]
is 1/R.
8.3 Properties of the z-Transform 361
Time Shifting If x(n) is a sequence with z-transform X(z) and ROC R, then the
z-transform of the delayed sequence x(n k), k being an integer, is given by
Z ½xðn kÞ ¼ zk X ðzÞ ð8:27Þ
whose ROC is the same as that of X(z) except for z ¼ 0 if k > 0 and z ¼ 1 if k < 0
Proof X1
Z fx ð n k Þ g ¼ n¼1
xðn kÞzn ð8:28Þ
Substituting m ¼ n k,
P1 ð m þ k Þ
P1
Z ½ xð n k Þ ¼ m¼1 xðmÞz ¼ zk m¼1 xðmÞzm
P ð8:29Þ
¼ zk 1m¼1 xðmÞz
m
¼ zk X ðzÞ ð8:30Þ
It is seen from Eq. (8.30) that in view of the factor zk, the ROC of Z [x(n k)] is
the same as that of X(z) except for z ¼ 0 if k > 0 and z ¼ 1 if k < 0. It is also observed
that in particular, a unit delay in time translates into the multiplication of the
z-transform by z1.
Scaling in the z-Domain If x(n) is a sequence with z-transform X(z), then Z{anx(n)} ¼
X(a1z) for any constant a, real or complex. Also, the ROC of Z{anx(n)} is |a|R, i.e.,
|a|r1 < |z| < |a|r2.
Proof
X
1
Z fan x ð nÞ g ¼ an xðnÞzn ð8:31Þ
n¼1
X
1 z n z
¼ x ð nÞ ¼X ð8:32Þ
n¼1
a a
Since the ROC of X(z) is r1 < |z| < r2, the ROC of X(a1z) is given by r1 < |a–1z| < r2,
that is,
jajr 1 < jzj < jajr 2
Differentiation in the z-Domain If x(n) is a sequence with z-transform X(z), then
dX ðzÞ
Z fnxðnÞg ¼ z ð8:33Þ
dz
whose ROC is the same as that of X(z).
Proof From the definition,
X1
Z ½ x ð nÞ ¼ n¼1
xðnÞzn
Differentiating the above equation with respect to z, we get
dX ðzÞ X1
¼ ðnÞxðnÞ zn1 ð8:34Þ
dz n¼1
Multiplying the above equation both sides by z, we obtain
dX ðzÞ X1
z ¼ z ðnÞxðnÞzn1 ð8:35Þ
dz n¼1
dX ðzÞ X1
z ¼ nxðnÞzn ¼ Z fnxðnÞg ð8:36aÞ
dz n¼1
Now, the region of convergence ra < |z| < rb of the sequence nx(n) can be found
using Eqs. (8.3a) and (8.3b).

ðn þ 1Þxðn þ 1Þ x ð n þ 1Þ

r a ¼ limn!1
nxðnÞ ¼ limn!1 xðnÞ ¼ r 1
and

ðn þ 1Þxðn þ 1Þ x ð n þ 1Þ

r b ¼ limn!1 ¼n lim ¼ r2
nxðnÞ n!1 xðnÞ
Hence, the ROC of Z[nx(n)] is the same as that of X(z).

By repeated differentiation of Eq. (8.36a), we get the result

dfX ðzÞg k
Z nk xðnÞ ¼ z ð8:36bÞ
dz
It is to be noted that the ROC of Z[nkx(n)] is also the same as that of X(z).
Convolution of Two Sequences If x1(n) and x2(n) are two sequences with z-trans-
forms X1(z) and X2(z), and ROCs R1 and R2, respectively, then
Z ½x1 ðnÞ∗x2 ðnÞ ¼ X 1 ðzÞX 2 ðzÞ ð8:37Þ
whose ROC is at least R1 \ R2.

Proof
X
1
X ðzÞ ¼ xðnÞzn ð8:38Þ
n¼1
8.3 Properties of the z-Transform 363
The discrete convolution of x1(n) and x2(n) is given by

X1 X1
x1 ðnÞ∗x2 ðnÞ ¼ x ðkÞx2 ðn
k¼1 1
kÞ ¼ x ðk Þx1 ðn
k¼1 2
kÞ ð8:39Þ
Hence, the z-transform of the convolution is

X1 hX1 i
Z ½x1 ðnÞ∗x2 ðnÞ ¼ n¼1 k¼1
x 2 ð k Þx 1 ð n k Þ zn ð8:40Þ
Interchanging the order of summation, the above equation can be rewritten

P1 P1 n
Z ½x1 ðnÞ∗x2 ðnÞ ¼ k¼1 x1 ðk Þ n¼1 x2 ðn k Þz
P1 P1 ðmþk Þ
¼ k¼1 x1 ðk Þ m¼1 x2 ðmÞz ð8:41Þ
P1 k
P1 m
¼ k¼1 x1 ðk Þz m¼1 x2 ðmÞz
Hence,
Z ½x1 ðnÞ∗x2 ðnÞ ¼ X 1 ðzÞX 2 ðzÞ ð8:42Þ
Since the right side of Eq. (8.42) is a product of the two convergent sequences
X1(z) and X2(z) with ROCs R1 and R1, it follows from the theory of complex
variables that the product sequence is convergent at least in the region R1 \ R2.
Hence, the ROC of Z[x1(n) ∗ x2(n)] is at least R1 \ R2.
Correlation of Two Sequences If x1(n) and x2(n) are two sequences with z-trans-
forms X1(z) and X1(z), and ROCs R1 and R2, respectively, then

Z ½r x1 x2 ðlÞ ¼ X 1 ðzÞX 2 z1 ð8:43Þ
whose ROC is at least R1 \ (1/R2)

Proof Since
r x1 x2 ðlÞ ¼ x1 ðlÞ∗x2 ðlÞ,

ð8:44Þ
Z ½r x1 x2 ðlÞ ¼ Z ½x1 ðlÞ∗x2 ðlÞ,
¼ Z ½x1 ðlÞZ ½x2 ðlÞ, using Equation ð8:37Þ
1
ð8:45Þ
¼ X 1 ðzÞX 2 ðz Þ, using Equation ð8:24Þ
Since the ROC of X2(z) is R2, the ROC of X2(z1) is 1/R2 from the property
concerning time reversal. Also, since the ROC of X1(z) is R1, it follows from
Eq. (8.45) that the ROC of Z ½r x1 x2 ðlÞ is at least R1 \ (1/R2).
Conjugate of a Complex Sequence If x(n) is a complex sequence with the

z-transform X(z), then
∗
Z ½x∗ðnÞ ¼ ½X ðz∗ Þ ð8:46Þ
with the ROCs of both X(z) and Z[x*(n)] being the same
Proof The z-transform of x*(n) is given by
X1
Z ½x∗ðnÞ ¼ n¼1
x∗ ðnÞzn ð8:47Þ
hX1 i
n ∗
¼ n¼1
xðnÞðz∗ Þ ð8:48Þ
In the R.H.S. of the above equation, the term in the brackets is equal to x (z*).
Therefore, Eq. (8.48) can be written as
Z½x∗ ðnÞ ¼ ½Xðz∗ Þ∗ ¼ X ∗ ðz∗ Þ ð8:49Þ
It is seen from Eq. (8.49) that the ROC of the z-transform of conjugate sequence is
identical to that of X(z).
Real Part of a Sequence If x(n) is a complex sequence with the z-transform
X(z), then
1
Z ½RefxðnÞg ¼ ½X ðzÞ þ X ∗ ðz∗ Þ ð8:50Þ
2
Proof
1
Z ½RefxðnÞg ¼ Z fxðnÞ þ x∗ ðnÞg ð8:51Þ
2
Since the z-transform satisfies the linearity property, we can write Eq. (8.51) as
1 1
Z ½RefxðnÞg ¼ Z ½xðnÞ þ Z ½x∗ ðnÞ ð8:52Þ
2 2
1
¼ ½X ðzÞ þ X ∗ ðz∗ Þ, using ð8:49Þ ð8:53Þ
2
It is clear that the ROC of Z[Re{x(n)}] is the same as that of X(z).
8.4 z-Transforms of Some Commonly Used Sequences 365
Imaginary Part of a Sequence If x(n) is a complex sequence with the z-transform

X(z), then
1
Z ½ImfxðnÞg ¼ ½X ðzÞ X ∗ ðz∗ Þ ð8:54Þ
2j
Proof Now
xðnÞ x∗ ðnÞ ¼ 2jImfxðnÞg ð8:55Þ
Thus,
1
ImfxðnÞg ¼ fx ð nÞ x ∗ ð nÞ g ð8:56Þ
2j
Hence,

1
Z½ImfxðnÞg ¼ Z fxðnÞ x∗ ðnÞg ð8:57Þ
2j
Again, since the z-transform satisfies the linearity property, we can write
Eq. (8.57) as
1 1
Z ½ImfxðnÞg ¼ Z ½xðnÞ Z ½x∗ ðnÞ
2j 2j
ð8:58Þ
1
¼ ½X ðzÞ X ∗ ðz∗ Þ, using ð8:49Þ
2j
Again, it is evident that the ROC of the above is the same as that of X(z). The
above properties of the z-transform are all summarized in Table 8.1.
8.4 z-Transforms of Some Commonly Used Sequences
Unit Sample Sequence The unit sample sequence is defined by

1 for n ¼ 0
δðnÞ ¼ ð8:59Þ
0 elsewhere
By definition, the z-transform of δ(n) can be written as

X1
X ðzÞ ¼ n¼1
xðnÞzn ¼ 1z0 ¼ 1 ð8:60Þ
It is obvious from (8.60) that the ROC is the entire z-plane.

Table 8.1 Some properties of the z-transform

Property Sequence z-Transform ROC
Linearity a1x1(n) þ a2x2(n) a1X1(z) þ a2X2(z) At least R1 \ R2
Time shifting x(n k) zkX(z). Same as R except for
z ¼ 0 if k > 0 and for
z ¼ 1 if k < 0
Time reversal x(n) X(z1) 1
R
n 1
Scaling in the a x(n) X(a z) |a|R
z-domain
Differentiation in nx(n) z dXdzðzÞ R
the z-domain
Convolution x1(n) ∗ x2(n) X1(z)X2(z) At least R1 \ R2
theorem
Correlation P
1 X1(z)X2(z1) At least R1 \ 1/R2
r x1 x2 ðlÞ ¼ x1 ðnÞx2 ðn lÞ
theorem n¼1
Conjugate com- x∗(n) [X(z∗)]∗ R
plex sequence
Real part of a Re[x(n)] 2 ½X ðzÞ
1
þ X ∗ ðz∗ Þ At least R
complex
sequence
Imaginary part of Im[x(n)] 2j ½X ðzÞ
1
X ∗ ðz∗ Þ At least R
a complex
sequence
Time reversal of a x∗(n) X∗(1/z∗) 1
R
complex conju-
gate sequence
Unit Step Sequence The unit step sequence is defined by

1 for n 0
uð nÞ ¼ ð8:61Þ
0 elsewhere
The z-transform of x(n) by definition can be written as

P1
X ðzÞ ¼ n¼1 xðnÞzn ¼ 1 þ z1 þ z2 þ
1 z ð8:62Þ
¼ ¼ for z1 < 1
1 z1 z1
Hence, the ROC for X(z) is |z| > 1
Example 8.3 Find the z-transform of x(n) ¼ δ(n k)
Solution By using the time shifting property, we get
Z ½δðn kÞ ¼ zk Z ½δðnÞ ¼ zk ð8:63Þ
The ROC is the entire z-plane except for z ¼ 0 if k is positive and for z ¼ 1 if k is
negative
Example 8.4 Find the z-transform of x(n) ¼ u(n 1)

Solution We know that Z ½uðnÞ ¼ z1 z
for jzj > 1 from Eq. (8.62)
Hence, using the time shifting property
z 1
Z ½uðn 1Þ ¼ z1 ¼ for jzj > 1 ð8:64Þ
z1 z1
Now, using the time reversal property (Table 8.1), we get
1 z
Z½uðn 1Þ ¼ ¼ for jzj < 1
z1 1 1z
Hence,
z
Z ½uðn 1Þ ¼ for jzj < 1 ð8:65Þ
z1
Example 8.5 Find the z-transform of the sequence x(n) ¼ {bnu(n)}

Solution Let x1(n) ¼ u(n). From Eq. (8.62), Z ½uðnÞ ¼ X 1 ðzÞ ¼ z1
z
for jzj > 1
Using the scaling property, we get
z
Z ½bn uðnÞ ¼ X 1 b1 z ¼ for jzj > jbj
zb
Example 8.6 Find the z-transform of x(n) ¼ nu(n)

Solution Let x1(n) ¼ u(n). Again, using Eq. (8.62), we have
Z ½uðnÞ ¼ X 1 ðzÞ ¼ z1
z
for jzj > 1
Using the differentiation property,
dX ðzÞ
Z ½nxðnÞ ¼ z
dz
we get

dX 1 ðzÞ d z z
Z ½nuðnÞ ¼ z ¼ z ¼ for jzj > 1
dz dz z 1 ðz 1Þ2
Example 8.7 Obtain the z-transform of the following sequence:

(
n2 u ð n Þ
xðnÞ ¼
0 elsewhere
Solution X1 X1
X ðzÞ ¼ n¼1
xðnÞzn ¼ n¼0
n2 uðnÞzn
Let x(n) ¼ n2x1(n), where x1(n) ¼ u(n). Then

z
X 1 ðzÞ ¼ for jzj > 1
z1
Using the differentiation property that

ZT ZT d 2
if xðnÞ $ X ðzÞ, then n2 xðnÞ $ X z X ðzÞ
dz
we get
" #
d d d z z ð z þ 1Þ
X ðzÞ ¼ z z ½X 1 ðzÞ ¼ z ¼
dz dz dz ðz 1Þ2 ðz 1Þ3
The ROC of X(z) is the same as that of u(n), namely, |z| > 1
Example 8.8 Find the z-transform of x(n) ¼ sin ωn u(n)
Solution jωn
e ejωn 1
Zfsin ωn uðnÞg ¼ Z uðnÞ ¼ ½Zfejωn uðnÞg Zfejωn uðnÞg
2j 2j

1 jωn jωn
1 z z
Z e uð nÞ Z e uð nÞ ¼
2j 2j z ejω z ejω
z sin ω
¼ 2 for jzj > 1
z 2z cos ω þ 1
Therefore,
z sin ω
Z f sin ωn uðnÞ ¼ for jzj > 1
z2 2z cos ω þ 1
Example 8.9 Find the z-transform of x(n) ¼ cos ωn u(n).

n jωn jωn i
Solution Z f cos ωn uðnÞ ¼ Z e þe 2 uðnÞ ¼ 12 ½Z fejωn uðnÞg þ Z fejωn uðnÞg

1 jωn 1 z z
½Zfe uðnÞg þ Zfe
jωn
uðnÞg ¼ þ
2 2 z ejω z ejω
zðz cos ωÞ
¼ for jzj > 1
z2 2zcos ω þ 1
Therefore,
zðz cos ωÞ
Z f cos ωn uðnÞ ¼ for jzj > 1
z2 2z cos ω þ 1
Example 8.10 Find the z-transform of the sequence x(n) ¼ [u (n) u (n 5)]
Solution
X
4
z 1 z5 1
XðzÞ ¼ zn ¼ 1 þ z1 þ z2 þ z3 þ z4 ¼ ð1 z5 Þ ¼ 4
n¼ 0
ðz 1Þ z z1
The ROC is the entire z-plane except for z ¼ 0

1
n
Example 8.11 Determine X(z) for the function xðnÞ ¼ 2 uðn 1Þ
Solution From Eq. (8.65), we have
z
Z ½uðn 1Þ ¼ for jzj < 1
z1
Now using the scaling property (Table 8.1),
n
1 2z 1
Z uðn 1Þ ¼ for jzj <
2 2z 1 2
Thus the ROC is jzj < 12

Example 8.12 Consider a system with input x(n) and output y(n). If its impulse
response h(n) ¼ Ax(L n), where L is an integer constant, and A is a known
constant, find Y(z) in terms of X(z).
Solution
hðnÞ ¼ AxðL nÞ
By the convolution property of the z-transform, we have
Y ðzÞ ¼ H ðzÞX ðzÞ
where
X1
H ðzÞ ¼ Z fAxðL nÞg ¼ A n¼1
xððn LÞÞzn
Letting n L ¼ m in the above, we have

P1 P
H ðzÞ ¼ A m ¼ 1 xðmÞzðmþLÞ ¼ AzL 1m¼1 xðmÞz
m
P
¼ AzL 1 m¼1 xðmÞz
m
¼ AzL X ðz1 Þ ¼ AzL X ð1=zÞ

Hence,
Y ðzÞ ¼ AzL X ð1=zÞX ðzÞ
A list of some commonly used z-transform pairs are given in Table 8.2
Initial Value Theorem
If a sequence x(n) is causal, i.e., x(n) ¼ 0 for n < 0, then
xð0Þ ¼ Lt X ½z ð8:66Þ

z!1
Proof Since x(n) is causal, its z-transform X[z] can be written as
X
1
X ½z ¼ xðnÞ:zn ¼ xð0Þ þ xð1Þz1 þ xð2Þz2 þ ð8:67Þ
n¼0
Now, taking the limits on both sides

Lt X ðzÞ ¼ Lt xð0Þ þ xð1Þz1 þ xð2Þz2 þ ¼ xð0Þ ð8:68Þ
z!1 z!1
Hence, the theorem is proved.

Example 8.13 Find the initial value of a causal sequence x(n) if its z-transform X(z)
is given by
0:5z2
X ðzÞ ¼
ðz 1Þðz2 0:85z þ 0:35Þ
Table 8.2 Some commonly used z-transform pairs

x(n) X(z) ROC
δ(n) 1 Entire z-plane
u(n) 1 |z| > 1
1 z1
nu(n) z1 |z| > 1
ð1 z1 Þ2
anu(n 1) 1 |z| < |a|
1 az1
nan{u(n 1)} az1 |z| < |a|
ð1 az1 Þ2
{cosωn} u(n) 1 z1 cos ω |z| > 1
1 2z1 cos ω þ z2
{sinωn} u(n) z1 sin ω |z| > 1
1 2z1 cos ω þ z2
8.5 The Inverse z-Transform 371
Solution The initial value x(0) is given by
0:5z2 0:5z2
xð0Þ ¼ lim X ðzÞ ¼ lim ¼ lim ¼0
z!1 n!1 ðz 1Þ ð z 2 0:85z þ 0:35Þ z!1 zðz2 Þ
8.5 The Inverse z-Transform
The z-transform of a sequence x(n), Z[x(n)], defined by Eq. (8.1), is

X1
X ðzÞ ¼ m¼1
xðmÞzm ð8:69Þ
Multiplying the above equation both sides by zn 1 and integrating both sides on
a closed contour C in the ROC of the z-transform X(z) enclosing the origin, we get
Þ Þ P1
C X ðzÞz
n1
dz ¼ C m¼1 xðmÞzm zn1 dz
Þ P1 ð8:70Þ
¼ C m¼1 xðmÞzmþn1 dz
1
Multiplying both sides of Eq. (8.70) by 2πj , we arrive at
þ þ X
1 1 1
X ðzÞzn1 dz ¼ xðnÞzmþn1 dz ð8:71Þ
2πj C 2πj C
m¼1
By Cauchy integral theorem, we have

þ X
1 1 mþn1 1 for m ¼ n
z dz ¼ ð8:72Þ
2πj m¼1 0 for m ¼
6 n
C þ
1
X ðzÞzn1 dz ¼ xðnÞ:
2πj C
Thus, the inverse z-transform of X(z), denoted by Z1[X(z)], is given by

þ
1 1
Z ½ X ð z Þ ¼ x ð nÞ ¼ X ðzÞzn1 dz ð8:73Þ
2πj C
It should be noted that given the ROC and the z-transform X(z), the sequence x(n)
is unique. Table 8.2 can be used in most of the cases for obtaining the inverse
transform. We will consider in Section 8.6 different methods of finding the inverse
transform.
8.5.1 Modulation Theorem in the z-Domain
The z-transform of the product of two sequences (real or complex) x1(n) and x2(n) is
given by
þ z
1
Z ½x1 ðnÞx2 ðnÞ ¼ X 1 ðvÞX 2 v1 dv ð8:74Þ
2πj C v
where C is a closed contour which

encloses the origin and lies in the ROC that is
common to both X1(v) and X 2 vz .
Proof Let x(n) ¼ x1(n)x2(n)
The inverse z-transform of x1(n) is given by
þ
1
x 1 ð nÞ ¼ X 1 ðvÞ vn1 dv ð8:75Þ
2πj C
Using Eq. (8.75), we get

þ
1
xðnÞ ¼ x1 ðnÞx2 ðnÞ ¼ X 1 ðvÞ vn1 x2 ðnÞdv ð8:76Þ
2πj C
Taking the z-transform of Eq. (8.76), we obtain

þ
P1 n
P1 1
XðzÞ ¼ n¼1 xðnÞz ¼ n¼1 X 1 ðvÞ v n1
x 2 ðnÞdv zn
þ 2πj C ð8:77Þ
1 X1
¼ X 1 ðvÞ½ n¼1 vn x2 ðnÞzn v1 dv
2πj C
Using the scaling property, we have that

X1 z
n¼1
vn x2 ðnÞ zn ¼ X 2
v
Hence, Eq. (8.77) becomes
þ z
1
X ðzÞ ¼ X 1 ðvÞX 2 v1 dv
2πj C v
which is the required result.
8.5.2 Parseval’s Relation in the z-Domain
If x1(n) and x2(n) are complex valued sequences, then

8.5 The Inverse z-Transform 373
X1 þ
∗ 1 ∗ 1 1
½x1 ðnÞx2 ð nÞ ¼ X 1 ð vÞ X 2 v dv ð8:78Þ
n¼1 2πj C v∗
where C is a contour contained in the ROC common to the ROCs of X1(v) and X ∗
2

1
.
v∗
Proof From Eq. (8.77), we have
þ z
1
Z ½x1 ðnÞx2 ðnÞ ¼ X 1 ðvÞX 2 v1 dv
2πj C v
Hence,
þ
1 z∗ 1
Z ½x1 ðnÞx2 ∗ ðnÞ ¼ X 1 ðvÞX 2 ∗ v dv ð8:79Þ
2πj C v∗
where we have used the result concerning the z-transform of a complex conjugate
(see Table 8.1). That is,
X1 þ
1 z∗ 1
½x ðnÞx2 ∗ ðnÞ zn ¼
n¼1 1
X 1 ðvÞX 2 ∗ v dv ð8:80Þ
2πj C v∗
Letting z ¼ 1 in Eq. (8.80), we get

X1 þ
1 1 1
½x ðnÞx2 ∗ ðnÞ ¼
n¼1 1
X 1 ðvÞX 2 ∗ v dv
2πj C v∗
Hence, the theorem.

If x1(n) ¼ x2(n) ¼ x(n) and the unit circle is included by the ROC of X(z), then by
letting v ¼ e jω in (8.78), we get the energy of sequence in the z-domain to be
X1 þ
1 1 1
j x ð nÞ j 2 ¼ X ðzÞX ∗ z dz ð8:81Þ
n¼1 2πj C z∗
For the energy of real sequences in the z-domain, the above expression becomes
X1 þ
2 1 1 1
jx ð n Þ j ¼ X ð z ÞX z z dz ð8:82Þ
n¼1 2πj C
The Parseval’s relation in the frequency domain is given by

X1 ðπ
1
n¼1
jxðnÞj2 ¼ jXðejω Þj2 dω
2π π
Thus,
X1 þ ðπ
2 1 1 1 1
jxðnÞj ¼ XðzÞXðz Þz dz ¼ jXðejω Þj2 dω ð8:83Þ
n¼1 2πj C 2π π
8.6 Methods for Computation of the Inverse z-Transform
8.6.1 Cauchy’s Residue Theorem for Computation

of the Inverse z-Transform
By Cauchy’s residue theorem, the integral in Eq. (8.73) for rational z-transforms
yields Z1[X(z)] ¼ x(n) ¼ sum of the residues of the function [X(z)zn1] at all the
poles pi enclosed by a contour C that lies in the ROC of X(z) and encloses the origin.
The residue at a simple pole pi is given by

res X ðzÞzn1 ¼ lim ðz pi Þ X ðzÞzn1 ð8:84Þ

z¼p z!pi
while for a pole pi of multiplicity m, the residue is given by

1 d m1
res X ðzÞzn1 ¼ lim m1 ðz pi Þm X ðzÞzn1 ð8:85Þ

z¼p ðm 1Þ! z!pi dz
We will now consider a few examples of finding the inverse z-transform using the
residue method.
Example 8.14 Assuming the sequence x(n) to be causal, find the inverse z-transform
of
z ð z þ 1Þ
X ðzÞ ¼
ð z 1Þ 3
Solution Since the sequence is causal, we have to consider the poles of X(z)zn1 for
only n 0. For n 0, the function X(z)zn1 has only one pole at z ¼ 1 of multiplicity
3. Thus, the inverse z-transform is given by
" #
1 d2 z ð z þ 1 Þ
x ð nÞ ¼ lim ð z 1Þ 3 zn1
ð3 1Þ! z!1 dz2 ðz 1Þ3
" #
1 d2 3 zðz þ 1Þ n1
x ð nÞ ¼ lim ð z 1Þ z
ð3 1Þ! z!1 dz2 ðz 1Þ3
1 d2 1
¼ lim 2 ½ðz þ 1Þzn ¼ lim nðn þ 1Þzn1 þ nðn 1Þzn2

2! z!1 dz 2 z!1
¼ n2
8.6 Methods for Computation of the Inverse z-Transform 375
It should be mentioned that if x(n) were not causal, then X(z)zn1 would have
had a multiple pole of order n at the origin, and we would have to find the residue of
X(z)zn1 at the origin to evaluate x(n) for n < 0.
Example 8.15 If x(n) is causal, find the inverse z-transform of
1
X ðzÞ ¼
2ðz 0:8Þðz þ 0:4Þ
Solution Since the sequence is causal, we have to consider the poles of X(z)zn1 for
only n 0. Hence X ðzÞzn1 ¼ 2ðz0:81Þðzþ0:4Þ zn1 , we see that for n 1, X(z)zn1 has
two simple poles at 0.8 and 0.4. However for n ¼ 0, we have an additional pole at
the origin. Hence, we evaluate x(0) separately by evaluating the residues of
X ðzÞz1 ¼ 2ðz0:81Þðzþ0:4Þ. Thus,
1 1 1
xð 0Þ ¼ jz¼0 þ jz¼0:4 þ jz¼0:8
2ðz 0:8Þ z þ 0:4 2zðz 0:8Þ 2zðz þ 0:4Þ
1 1 1
¼ þ þ ¼0
2ð0:8Þð0:4Þ 2ð0:4Þð1:2Þ 2ð0:8Þð1:2Þ
For n > 0,
zn1 zn1
x ð nÞ ¼ jz¼0 :4 þ jz¼0
2ðz 0:8Þ 2ðz þ 0:4Þ :8
ð0:4Þn1 0:8n1 1 n1
¼ þ ¼ : 0:8 ð0:4Þn1
2ð1:2Þ 2ð1:2Þ 2:4
Hence for any n 0,
1 n1
x ð nÞ ¼ : 0:8 ð0:4Þn1 uðn 1Þ
2:4
8.6.2 Computation of the Inverse z-Transform Using

the Partial Fraction Expansion
Partial fraction expansion is another technique that is useful for evaluating the
inverse z-transform of a rational function and is a widely used method. To apply
the partial fraction expansion method to obtain the inverse z-transform, we may
consider the z-transform to be a ratio of two polynomials in either z or in z1. We
now consider a rational function X(z) as given in Eq. (8.7). It is called a proper
rational function if M > N; otherwise, it is called an improper rational function. An
improper rational function can be expressed as a proper rational function by dividing
the numerator polynomial N(z) by its denominator polynomial D(z) and expressing X
(z) in the form
X
MN
N 1 ðzÞ
X ðzÞ ¼ f k zk þ ð8:86Þ
k¼0
DðzÞ
where the order of the polynomial N1(z) is less than that of the denominator
polynomial. The partial fraction expansion can be now made on N1(z)/D(z). The
z
inverse z-transform of the terms in the sum is obtained from the pair δ½n $ 1 (see
Table 8.1) and the time-shift property (see Table 8.2).
Let X(z) be a proper rational function expressed as
N ðzÞ b0 þ b1 z1 þ b2 z2 þ þ bM zM

X ðzÞ ¼ ¼ ð8:87Þ
DðzÞ 1 þ a1 z1 þ a2 z2 þ þ aN zN
For simplification, eliminating negative powers, Eq. (8.87) can be rewritten as
NðzÞ b0 zN þ b1 zN1 þ b2 zN2 þ þ bM zNM

XðzÞ ¼ ¼ ð8:88Þ
DðzÞ zN þ a1 zN1 þ a2 z2 þ þ aN
Since X(z) is a proper fraction, so will be [X(z)/z]. If all the poles pi are simple,
then, [X(z)/z] can be expanded in terms of partial fractions as
X ðzÞ X N
ci
¼ ð8:89Þ
z i¼1
z pi
where

X ðzÞ
c i ¼ ð z pi Þ ð8:90Þ
z z¼v
If [X(z)/z] has a multiple pole, say at pj, with a multiplicity of k, in addition to

(Nk) simple poles at pi, then the partial fraction expansion given in Eq. (8.89) has
to be modified as follows.
X ðzÞ cj1 cj2 cjk XNk ci

¼ þ 2 þ þ k þ ð8:91Þ
z z pj z pj z pj i¼1 z p
i
where ci is still given by (8.90) and cjk by

1 dðkjÞ k X ðzÞ
cjk ¼ z p ð8:92Þ
ðk jÞ! dzkj j
z z¼pj
Hence,
cj1 z cj2 z cjk z XNk ci z

X ðzÞ ¼ þ 2 þ þ k þ ð8:93Þ
z pj z pj z pj i¼1 z p
i
Then inverse z-transform is obtained for each of the terms on the right-hand side
of (8.91) by the use of Tables 8.1 and 8.2. We will now illustrate the method by a few
examples.
Example 8.16 Assuming the sequence x(n) to be right-sided, find the inverse
z-transform of the following:
z
X ðzÞ ¼
ð z aÞ ð z bÞ
Solution The given function has poles at z ¼ a and z ¼ b. Since X(z) is a right-sided
sequence, the ROC of X(z) is the exterior of a circle around the origin that includes
both the poles. Now X(z)/z can be expressed in partial fraction expansion as
X ðzÞ a 1 b 1
¼
z a bz a a bz b
Hence,
a 1 b 1
X ðzÞ ¼
a b 1 az1 a b 1 bz1
We can now find the inverse transform of each term using Table 8.2 as
a b
x ð nÞ ¼ an uð nÞ bn uðnÞ
ab ab
Example 8.17 Assuming the sequence x(n) to be causal, find the inverse z-transform
of the following:
10z2 3z
X ðzÞ ¼
10z2 9z þ 2
Solution Dividing the numerator and denominator by z2, we can rewrite X(z) as
10 3z1
¼
10 9z1 þ 2z2
4 5
¼ 1

2z 5 2z1
2 1
¼
1 0:5z1 1 0:4z1
Each term in the above expansion is a first-order z-transform and can be recog-
nized easily to evaluate the inverse transform as
Z 1 fX ðzÞg ¼ xðnÞ ¼ 2ð0:5Þn uðnÞ ð0:4Þn uðnÞ:
Example 8.18 Assuming the sequence x(n) to be causal, determine the inverse
z-transform of the following:
z ð z þ 1Þ
X ðzÞ ¼
ð z 1Þ 3
Solution Since X(z)/z can be written in partial fraction expansion as
X ðzÞ zþ1 A B C
¼ ¼ þ þ
z ðz 1Þ3 z 1 ðz 1Þ2 ðz 1Þ3
Solving for A, B, and C, we get A ¼ 0, B ¼ 1, C ¼ 2. Hence, X(z) can be expanded

as
z 2z
X ðzÞ ¼ 2
þ
ð z 1Þ ðz 1Þ3
Making use of Table 8.2, the inverse z-transform of X(z) can be written as
Z 1 fX ðzÞg ¼ xðnÞ ¼ nuðnÞ þ nðn 1ÞuðnÞ ¼ n2 uðnÞ
Example 8.19 If x(n) is a right-handed sequence, determine the inverse z-transform

for the function:
1 þ 2z1 þ z3
X ðzÞ ¼
ð1 z1 Þð1 0:5z1 Þ
Solution
1 þ 2z1 þ z3 z3 þ 2z2 þ 1
X ðzÞ ¼ ¼
ð1 z1 Þð1 0:5z1 Þ zðz 1Þðz 0:5Þ
Now, X(z)/z can be written in partial fraction expansion form as
X ðzÞ z3 þ 2z2 þ 1 A B C D
¼ 2 ¼ þ 2þ þ
z z ðz 1Þðz 0:5Þ z z ðz 1Þ ðz 0:5Þ
Solving for A, B, C, and D, we get A ¼ 6, B ¼ 2, C ¼ 8, D ¼ 13. Hence,
z3 þ 2z2 þ 1 2 8z 13z
X ðzÞ ¼ ¼6þ þ
zðz 1Þðz 0:5Þ z ðz 1Þ ðz 0:5Þ
Since the sequence is right-handed and the poles of X(z) are located z ¼ 0, 0.5,
and 1, the ROC of X(z) is |z| > 1. Thus, from Table 8.2, we have
Z 1 fX ðzÞg ¼ xðnÞ ¼ 6δðnÞ þ 2δðn 1Þ þ 8uðnÞ 13ð0:5Þn uðnÞ
Example 8.20 Assuming h(n) to be causal, find the inverse z-transform of
ð z 1Þ 2
H ðzÞ ¼
ðz2 0:1z 0:56Þ
Solution Expanding H(z)/z as
H ðzÞ ð z 1Þ 2 A B C
¼ ¼ þ þ
z zðz 0:8Þðz þ 0:7Þ z ðz 0:8Þ ðz þ 0:7Þ
Solving for A, B, and C, we get A ¼ 1.78, B ¼ 0.033, and C ¼ 2.75

Therefore, H(z) can be expanded as
0:0333z 2:7524z
H ðzÞ ¼ 1:7857 þ þ
ðz 0:8Þ ðz þ 0:7Þ
Hence,
Z 1 fH ðzÞg ¼ hðnÞ ¼ 1:7857δðnÞ þ 0:0333ð0:8Þn uðnÞ þ 2:7524ð0:7Þn uðnÞ
8.6.3 Inverse z-Transform by Partial Fraction Expansion

Using MATLAB
The M-file residue z can be used to find the inverse z-transform using the power
Series expansion.
The coefficients of the numerator and denominator polynomial written in
descending powers of z for Example 8.20 can be
num= [1 -2 1];
den= [1 -0.1 -0.56];
The following MATLAB statement determines the residue (r), poles (p), and
direct terms (k) of the partial fraction expansion of H(z).
[r,p,k]= residuez(num,den);
After execution of the above statements, the residues, poles, and constants
obtained are
Residues: 0.0333 2.7524
Poles: 0.8000 –0.7000
Constants: 1.7857
The desired expansion is
0:0333z 2:7524z
H ðzÞ ¼ 1:7857 þ þ ð8:94Þ
ðz 0:8Þ ðz þ 0:7Þ
8.6.4 Computation of the Inverse z-Transform Using

the Power Series Expansion
The z-transform of an arbitrary sequence defined by Eq. (8.1) implies that X(z) can be
expressed as power series in z1 or z. In this expansion, the coefficient of the term
indicates zn the value of the sequence x(n). Long division is one way to express X(z)
in power series.
Example 8.21 Assuming h(n) to be causal, find the inverse z-transform of the
following:
z2 þ 2z þ 1
H ðzÞ ¼
z2 þ 0:4z 0:12
Solution We obtain the inverse z-transform by long division of the numerator by

the denominator as follows:
1 þ 1:6z1 þ 0:48z2 þ 0z3 þ 0:0576z4 þ
z þ 0:4z 0:12
2
j z2 þ 2z þ 1
z2 þ 0:4z 0:12
1:6z þ 1:12
1:6z þ 0:64 0:192z1
0:48 þ 0:192z1
0:48 þ 0:19z1 0:0576z2
0:0576z2
0:0576z2 þ 0:02304z3 0:006912z4

0:02304z3 þ 0:006912z4
............
Hence, H(z) can be written as
H ðzÞ ¼ 1:0 þ 1:6z1 þ 0:48z2 þ 0z3 þ 0:0576z4 þ
implying that
fh½ng ¼ f1:0; 1:6; 0:48; 0 0:0576; . . .g for n 0
Example 8.22 Find the inverse z-transform of the following:

X ðzÞ ¼ log 1 þ bz1 , jbj < jzj
Solution We know that power series expansion for log(1 þ u) is
u2 u3 u4 u5
logð1 þ uÞ ¼ u þ þ
2 3 4 5
X1
ð1Þnþ1 un
¼ , j uj < 1
n¼1
n
Letting u ¼ bz1, X(z) can be written as
X 1
ð1Þnþ1 bn zn
X ðzÞ ¼ log 1 þ bz1 ¼ , jbj < jzj
n¼1
n
From the definition of z-transform of x(n), we have
X
1
X ðzÞ ¼ xðnÞzn
n¼1
Comparing the above two expressions, we get x(n), i.e., the inverse z-transform of
X(z) ¼ log (1 þ bz1) to be
(
bn
x ð nÞ ¼ ð1Þnþ1 n n>0 ð8:95Þ
0 n0
Example 8.23 Find the inverse z-transform of

z
X ðzÞ ¼ , for jzj > jbj
zb
Solution The sequence is a right-sided causal sequence as the region of conver-

gence is |z| > |b|. We can use the long division as we did in Example 8.21 to express z/
(zb) as a series in powers of z1. Instead, we will use binomial expansion.
z 1
X ðzÞ ¼ ¼
z b 1 bz1

¼ 1 þ bz1 þ b2 z2 þ for bz1 < 1
P
¼ 1 n 1
n¼0 b z for jzj > jbj
Hence,

z
Z 1 fX ðzÞg ¼ xðnÞ ¼ Z 1 ¼ bn uðnÞ:
zb
Example 8.24 Find the inverse z-transform of

z
X ðzÞ ¼ , for jzj < jbj
zb
Solution Since the region of convergence is |z| < |b|, the sequence is a left-sided
sequence. We can use the long division to obtain z/(zb) as a power series in z.
However, we will use the binomial expansion.
z z 1
X ðzÞ ¼ ¼
zb b1 ðz=bÞ
z
z z z 2
¼ 1þ þ þ for <1
b b b b
P
¼ 1 n n
n¼1 b z for jzj < jbj
Hence,

z
Z 1 fX ðzÞg ¼ xðnÞ ¼ Z 1 ¼ bn uðn 1Þ
zb
Example 8.25 Using the z-transform, find the convolution of the sequences:
x1 ðnÞ ¼ f1; 3; 2g and x2 ðnÞ ¼ f1; 2; 1g
Solution
Step 1: Determine z-transform of individual signal sequences
P2 1
X 1 ð z Þ ¼ Z ½ x 1 ð nÞ ¼ n¼0 x1 ðnÞz ¼ x1 ð0Þ þ x1 ð1Þz1 þ x1 ð2Þz2
¼ 1 3z1 þ 2z 2
and
P2 1
X 2 ð z Þ ¼ Z ½ x 2 ð nÞ ¼ n¼ 0 x2 ðnÞz ¼ x2 ð0Þ þ x2 ð1Þz1 þ x2 ð2Þz2
¼ 1 þ 2z1 þ z 2
Step 2: Obtain X(z) ¼ X1(z)X2(z)
X ðzÞ ¼ ð1 3z1 þ 2z2 Þð1 þ 2z1 þ z2 Þ

¼ 1 z1 3z2 þ z3 þ 2z4
Step 3: Obtain the inverse z-transform of X(z)

xðnÞ ¼ Z 1 1 z1 3z2 þ z3 þ 2z4 ¼ f1; 1; 3; 1; 2g
8.6.5 Inverse z-Transform via Power Series Expansion

Using MATLAB
The M-file impz can be used to find the inverse z-transform using the power series
expansion.
The coefficients of the numerator and denominator polynomial for Example 8.21
can be written as
num = [1 2 1];
den = [1 0.4 -0.12];
The following statement can be run to obtain the coefficients of the inverse z-
transform:
h = impz(num,den);
where h is the vector containing the coefficients of the inverse z-transform. The
first 11 coefficients of the inverse z-transform of Example 8.21 obtained after
execution of the above MATLAB statements are
Columns 1 through 9
1.0000 1.6000 0.4800 0 0.0576 -0.0230 0.0161 -0.0092 0.0056
Columns 10 through 11
-0.0034 0.0020

Using the z-Transform
Example 8.26 Determine the impulse response of the system described by the
difference equation:
yðnÞ 3yðn 1Þ 4yðn 2Þ ¼ xðnÞ þ 2xðn 1Þ:
Assume that the system is relaxed initially.

Solution Let X(z) ¼ Z[x(n)] and Y(z) ¼ Z[y(n)]. Taking z-transform on both sides
and using the time shifting property, we get

1 3z1 4z2 Y ðzÞ ¼ 1 þ 2z1 X ðzÞ
Since X(z) ¼ 1, we have
1 þ 2z1
YðzÞ ¼
1 3z1 þ 4z2
YðzÞ zþ2 ð6=5Þ ð1=5Þ
¼ ¼
z ðz 4Þðz þ 1Þ z 4 z þ 1
ð6=5Þ ð1=5Þ
YðzÞ ¼
1 4z1 1 þ z1
We now take inverse transform of the above and use Table 8.2 to obtain y(n),
which is the impulse response of the system as
hðnÞ ¼ yðnÞ ¼ ð6=5Þ4n uðnÞ ð1=5Þð1Þn uðnÞ
Example 8.27 Determine the response y(n), n 0 of the system described by the
second-order difference equation
yðnÞ 3yðn 1Þ 4yðn 2Þ ¼ xðnÞ þ 2xðn 1Þ
for the input x(n) ¼ 4n u(n)

Solution Applying z-transform to both sides of the equation, we have

Y ðzÞ 1 3z1 4z2 ¼ X ðzÞ 1 þ 2z1
Given that x(n) ¼ 4n u(n), we have
1
X ðzÞ ¼
1 4z1
Substituting for X(z) in the expression for Y(z) and simplifying, we get
Y ðzÞ ð z 2 þ 2Þ
¼
z ðz 4Þ2 ðz þ 1Þ
or
Y ðzÞ 1 26 24
¼ þ þ
z 25ðz þ 1Þ 25ðz 4Þ 5ðz 4Þ2
8.7 Analysis of Discrete-Time LTI Systems in the z-Transform Domain 385
Hence,
z 26z 24z
Y ðzÞ ¼ þ þ
25ðz þ 1Þ 25ðz 4Þ 5ðz 4Þ2
By applying inverse z-transforms, we get
1 6 26
y ð nÞ ¼ ð1Þn uðnÞ þ nð4Þn uðnÞ þ ð4Þn uðnÞ
25 5 25
Example 8.28 Find the impulse response of the system
yðnÞ ¼ 3yðn 1Þ þ 2yðn 2Þ þ xðnÞ
Solution Taking z-transforms on both sides of the above equation, and using the
factZ[δ(n)] ¼ 1, we get
1 z2
Y ðzÞ ¼ ¼
1 3z1
2z2 3z 2
z2
0:86 0:135
Y ðzÞ ¼ þ
1 3:56z1 1 þ 0:56z1
Hence, the impulse response is given by
hðnÞ ¼ yðnÞ ¼ 0:86ð3:56Þn uðnÞ þ 0:135ð0:561Þn uðnÞ
8.7 Analysis of Discrete-Time LTI Systems

in the z-Transform Domain
8.7.1 Transfer Function
It was stated in Chapter 6 that an LTI system can be completely characterized by its
impulse response h(n). The output signal y(n) of a LTI system and the input signal x
(n) are related by convolution as
yðnÞ ¼ hðnÞ∗xðnÞ ð8:96Þ
Taking z-transform on both sides of the above equation and using the convolution
property, we get
Y ðzÞ ¼ H ðzÞX ðzÞ ð8:97Þ
indicating the z-transform of the output sequence y(n) is the product of the z-trans-
forms of the impulse response h(n) and the input sequence x(n). The quantities h(n)
and H(z) are two equivalent descriptions of a system in the time domain and
z-domain, respectively. The transform H(z) is called the transfer function or the
system function and expressed as
Y ðzÞ
H ðzÞ ¼ ð8:98aÞ
X ðzÞ
Or equivalently,
PM
bk zk
H ðzÞ ¼ P
k¼0
N
ð8:98bÞ
1þ k¼1 ak zk
where the constants ak and bk are real.

The above transfer function is a ratio of polynomials in z1 and, hence, is a
rational transfer function or system function.
Example 8.29 The following are known about a LTI discrete-time system:
(i) y(n) ¼ δ(n) þ a(0.25)nu(n) for x(n) ¼ (0.5)nu(n)
(ii) y(n) ¼ 0 for all n if x(n) ¼ (2)n for all n
Find the value of the constant a.
Solution It is given that for the input x(n) ¼ (0.5)nu(n), the output of the LTI system
is y(n) ¼ δ(n) þ a(0.25)nu(n). From this fact, the transfer function H(z) is given by
Y ðzÞ
H ðzÞ ¼
X ðzÞ
1 þ a 0:25z1
¼
ð1 0:25z1 Þð1 0:5z1 Þ
It is also given that the output y(n) ¼ 0 for the input x(n) ¼ (2)n for all n. Since
the function z0n is an eigenfunction for a discrete-time LTI system, the output to this
input is H ðz0 Þz0n . From this, it can be inferred that H(2) ¼ 0. Using this in the above
transfer function, the value of a is calculated to be 1.125
8.7.2 Poles and Zeros of a Transfer Function
As mentioned earlier, the zeros of a system function H(z) are the values of z for
which H(z) ¼ 0, while the poles are the values of z for which H(z) ¼ 1. Since H(z) is
a rational transfer function, the number of finite zeros and the number of finite poles
are equal to the degrees of the numerator and denominator polynomials,
respectively.
In MATLAB, tf2zp command can be used to find the zeros, poles, and gains of a
rational transfer function. z plane command can be used for plotting pole-zero plot of
a rational transfer function.
Example 8.30 Determine the pole-zero plot using MATLAB for the system
described by the system function
Y ðzÞ z1
H ðzÞ ¼ ¼ 2
X ðzÞ 8z 6z þ 1
Solution The coefficients of the numerator and denominator polynomial can be

written as
numerator = [0 1 -1];
denominator = [8 -6 1];
The following MATLAB statement yields the poles and zeros and gain of the
system:
[z,p,gain] = tf2zp (numerator, denominator)

zeros, z = 1
poles, p = [0.500 0.250] and gain = 0.1250
The MATLAB command z-plane (z, p) plots the poles and zeros as shown in
Figure 8.6.
Figure 8.6 Pole-zero plot of Example 8.30

8.7.3 Frequency Response from Poles and Zeros
By factorizing the numerator and denominator polynomials of Eq. (8.98b), the

transfer function can be written in pole-zero form as
Q
M
ðz zi Þ
b0 zðNMÞ N
i¼1
HðzÞ ¼ ð8:99Þ
Q
ðz pi Þ
i¼1
where zi and pi are the zeros and poles of H(z). It should be noted that the zeros are
either real or occur in conjugate pairs. The frequency response of the system can be
obtained by letting z ¼ e jω in the transfer function H(z), that is,

H ejω ¼ H ðzÞz¼ejω
Hence,
Q
M
ðejω zi Þ
Hðejω Þ ¼ b0 ejωðNMÞ i¼1 ð8:100Þ
QN
ðejω pi Þ
i¼1
The contribution of the zeros and poles to the system frequency response can be
visualized from the above expression.
The magnitude of the frequency response can be expressed by
Q
M
jðejω zi Þj
jHðe Þj ¼
jω
jb0 jjejω jðNMÞ i¼1 ð8:101Þ
QN
jðejω pi Þj
i¼1
The zeros contribute to pulling down the magnitude of the frequency response,
whereas the poles contribute to pushing up the magnitude of the frequency response.
The size of decrease or increase in the magnitude response depends on how far the
zero or the pole is from the unit circle. A peak in |H(e jω)| appears at the frequency of
a pole very close to the unit circle.
To illustrate this, consider the following example.
Example 8.31 Consider a system with the transfer function
0:1ðz2 þ 2z þ 1Þ
H ðzÞ ¼ ð8:102Þ
1:2z2 þ 1
The numerator and denominator polynomials coefficients in descending powers

of z can be written as
num=[1 2 1];
den=[1.2 0 1];
Then, as used in Example 8.30, using the MATLAB commands tf2zp and
z-plane, the pole zero plot can be obtained as shown in Figure 8.7(a). The magnitude
and phase responses of the above system transfer function are obtained using the
above num and den vectors using the MATLAB command freqz. The magnitude and
phase responses are shown in Figure 8.7(b) and (c), respectively.
Figure 8.7(a) indicates that the system has zeros of order 2 at z ¼ 1 and two
poles on the imaginary axis close to the unit circle. In the magnitude response of
Figure 8.7(b), a peak occurs at ω ¼ π/2. This can be attributed to the fact that the
frequency of the poles is π/2. The magnitude response is small at high frequencies
due to the zeros.
8.7.4 Stability and Causality
The stability of a LTI system can be expressed in terms of the transfer function or the
impulse response of the system. It is known from Section 6.4.5 that a necessary and
sufficient condition for a LTI system to be BIBO (bounded-input bounded-output)
stable is that its impulse response be absolutely summable, i.e.,
X
1
jhðnÞj < 1 ð8:103Þ
n¼1
X1
H ðzÞ ¼ n¼1
hðnÞ zn ð8:104Þ
X
1 X
1
j H ðzÞj jhðnÞzn j ¼ jhðnÞjjzn j ð8:105Þ
n¼1 n¼1
On the unit circle (i.e., |z| ¼ 1), the above expression becomes
X
1
jH ðzÞj j hð nÞ j ð8:106Þ
n¼1
Therefore, for a stable system, the ROC of its transfer function H(z) must include
the unit circle. Thus we have the following theorem.
BIBO Stability Theorem
A discrete LTI system is BIBO stable if and only if the ROC of its system function
includes the unit circle, |z| ¼ 1.
Figure 8.7 (a) Pole-zero

plot, (b) magnitude
response, (c) phase response
We know from Section 6.4.5 that for a discrete LTI system to be causal h(n) ¼ 0
for n < 0. Thus, the sequence should be right-sided. We also know from Section 8.2
that the ROC of a right-sided sequence is the exterior of a circle whose radius is
equal to the magnitude of the pole that is farthest from the origin. At the same time,
we also know that for a right-sided sequence, the ROC may or may not include the
point z ¼ 1. But we know from Section 8.2 that a causal system cannot have a pole
at infinity. Thus, in a causal system, the ROC should include the point z ¼ 1. Thus,
we may summarize the result for causality by the following theorem:
Causality Theorem
A discrete LTI system is causal if and only if the ROC of its system function is the
exterior of a circle including z ¼ 1. An alternate way of stating this result is that a
system is causal if and only if its ROC contains no poles, finite or infinite.
Thus the conditions for stability and causality are quite different. A causal system
could be stable or unstable, just as a noncausal system could be stable or unstable.
Also, a stable system could be causal or noncausal just as an unstable system could
be causal or noncausal. However, we can conclude from the above two theorems that
a causal stable system must have a system function whose ROC is |z| ¼ r, where
r < 1. Hence, we can summarize this result as follows.
Condition for a System to Be Both Causal and Stable
A causal LTI system is BIBO stable if and only if all its poles are within the unit
circle.
As a consequence, for a LTI system with a system function H(z) to be stable and
causal, it is necessary that the degree of the numerator polynomial in z not exceed
that of the denominator polynomial. As such, an FIR system is always stable,
whereas if an IIR system is not designed properly, it may be unstable.
Example 8.32 Given the system function
zð4z 3Þ
H ðzÞ ¼
z 13 ðz 4Þ
Find the various regions of convergence for H(z), and state whether the system is
stable and/or causal in each of these regions. Also, find the impulse response h(n) in
each case.
Solution The system function can be expressed in partial fraction in the form
z 3z 1 1
H ðzÞ ¼ þ ¼ þ3
z31 ð z 4Þ 1 3z
1 1 1 4z1
The system function has two zeros, viz., z ¼ 0, 34, and two poles at z ¼ 13 , 4:
Hence, there are three regions of convergence: (i) jzj < 13, (ii) 13 < jzj < 4, and
(iii) |z| > 4. Let us consider each of these regions separately.
(i) jzj < 13

In this region, there are no poles including the origin, but has poles exterior to
it. Hence, the system is noncausal. Also, it is an unstable system, since the ROC does
not include the unit circle. By using Table 8.2, we get
n
1 n
hðnÞ ¼ þ 3ð4Þ uðn 1Þ
3
(ii) 1
3 < j zj < 4
This region includes the unit circle and hence the system is stable. However, since
the pole |z| ¼ 4 is exterior to this region, it is noncausal, and the corresponding
sequence is two-sided. Again by using Table 8.2, we have
n
1
hð nÞ ¼ uðnÞ 3ð4Þn uðn 1Þ
3
(iii) |z| > 4

This region does not include the unit circle, and hence the system is unstable.
However, in this region, there are no poles, finite or infinite, and hence, the system is
causal. The impulse response of the system is obtained from H(z) using Table 8.2 as
n
1
hð nÞ ¼ uð nÞ þ 3ð 4Þ n uð nÞ
3
Example 8.33 The rotational motion of a satellite was described by the difference
equation
yðnÞ ¼ yðn 1Þ 0:5 yðn 2Þ þ 0:5 xðnÞ þ 0:5 xðn 1Þ
Is the system stable? Is the system causal? Justify your answer.

Solution Taking the z-transform on both sides of the given difference equation, we
get
Y ðzÞ ¼ z1 Y ðzÞ 0:5z2 Y ðzÞ þ 0:5X ðzÞ þ 0:5z1 X ðzÞ

Y ðzÞ 0:5ð1 þ z1 Þ 0:5ðz þ 1Þz
H ðzÞ ¼ ¼ ¼
X ðzÞ 1 z1 þ 0:5z2 ðz2 z þ 0:5Þ
The poles of the system are at z ¼ 0.5 0.5j as shown in Figure 8.8
All poles of the system are inside the unit circle. Hence, the system is stable. It is
causal since the output only depends on the present and past inputs.
Figure 8.8 Poles of

Example 8.33
´
0.5
0.5
-0.5
´
Example 8.34 Consider the difference equation
7 2
yðnÞ yðn 1Þ þ yðn 2Þ ¼ xðnÞ
3 3
(a) Determine the possible choices for the impulse response of the system. Each
choice should satisfy the difference equation. Specifically indicate which choice
corresponds to a stable system and which choice corresponds to a causal system.
(b) Can you find a choice which implies that the system is both stable and causal? If
not, justify your answer.
Solution (a) Taking the z-transform on both sides and using the shifting theorem,
we get

7 2
1 z1 þ z2 Y ðzÞ ¼ X ðzÞ
3 3
Y ðzÞ 1
¼
X ðzÞ 7 1 2 2
1 z þ z
3 3
2
z
H ðzÞ ¼
1
ð z 2Þ z
3
The system function H(z) has a zero of order 2 at z ¼ 0 and two poles at z ¼ 1/3,
2. Hence, there are three regions of convergence, and thus, there are three possible
choices for the impulse response of the system. The regions are
(i) R1: jzj < 13, (ii) R2: 13 < jzj < 2, and (iii) R3: |z| > 2.
The region R1 is devoid of any poles including the origin and hence corresponds
to an anti-causal system, which is not stable since it does not include the unit circle.
Region R2 does include the unit circle and hence corresponds to a stable system;
however, it is not causal in view of the presence of the pole z ¼ 2. Finally, the region
R3 does not have any poles including at infinity and hence corresponds to a causal
system; however, since R3 does not include the unit circle, the system is not stable.
(b) There is no ROC that would imply that the system is both stable and causal.
Therefore, there is no choice for h(n) which make the system both stable and
causal.
Example 8.35 A system is described by the difference equation
yðnÞ þ yðn 1Þ ¼ xðnÞ, yðnÞ ¼ 0, for n < 0:
(i) Determine the transfer function and discuss the stability of the system.
(ii) Determine the impulse response h(n) and show that it behaves according to the
conclusion drawn from (i).
(iii) Determine the response when x(n) ¼ 10 for n 0. Assume that the system is
initially relaxed.
Solution (i) Taking the z-transforms on both sides of the given equation, we get
Y ðzÞ þ Y ðzÞz1 ¼ X ðzÞ
Hence,
Y ðzÞ z
H ðzÞ ¼ ¼
X ðzÞ z þ 1
The pole is at z ¼ 1, that is, on the unit circle. So the system is marginally stable
or oscillatory.
(ii) Since h(n) ¼ 0 for n < 0,

z
hðnÞ ¼ Z 1 ¼ ð1Þn uðnÞ
zþ1
This impulse response confirms that the impulse response is oscillatory.

(iii) Since
xðnÞ ¼ 10 for n 0,
10z
X ðzÞ ¼
z1
Thus,
z 10
Y ðzÞ ¼ H ðzÞX ðzÞ ¼
zþ 1z 1
or
Y ðzÞ 5 5
¼ þ
z zþ1 z1
Therefore,
yðnÞ ¼ Z 1 ½Y ðzÞ ¼ ½5ð1Þn þ 5uðnÞ
8.7.5 Minimum-Phase, Maximum-Phase, and Mixed-Phase

Systems
A causal stable transfer function with all its poles and zeros inside the unit circle is
called a minimum-phase transfer function. A causal stable transfer function with all
its poles inside the unit circle and all the zeros outside the unit circle is called a
maximum-phase transfer function. A causal stable transfer function with all its poles
inside the unit circle and with zeros inside and outside the unit circle is called a
mixed-phase transfer function. For example, consider the systems with the following
transfer functions:
Y ðzÞ z þ 0:4
H 1 ðzÞ ¼ ¼ ð8:107Þ
X ðzÞ z þ 0:3
Y ðzÞ 0:4z þ 1
H 2 ðzÞ ¼ ¼ ð8:108Þ
X ðzÞ z þ 0:5
Y ðzÞ ð0:4z þ 1Þ ðz þ 0:4Þ
H 3 ðzÞ ¼ ¼ ð8:109Þ
X ðzÞ ðz þ 0:5Þ ðz þ 0:3Þ
The pole-zero plot of the above transfer functions are shown in Figure 8.9 (a), (b),
and (c), respectively. The transfer function H1(z) has a zero at z ¼ 0.4 and a pole at
z ¼ 0.3, and they are both inside the unit circle. Hence, H1(z) is a minimum-phase
function. The transfer function H2(z) has a pole inside the unit circle, at z ¼ 0.5,
and a zero at z ¼ 2.5, outside the unit circle. Thus, H2(z) is a maximum-phase
function. The transfer function H3(z) has two poles, one at z ¼ 0.3 and the other
at z ¼ 0.5, and two zeros one at z ¼ 0.4, inside the unit circle, and the other
at z ¼ 2.5, outside the unit circle. Hence, H3(z) is a mixed-phase function.
8.7.6 Inverse System
Let H(z) be the system function of a linear time-invariant system. Then its inverse
system function HI(z) is defined, if and only if the overall system function is unity
when H(z) and HI(z) are connected in cascade, that is, H(z) HI(z) ¼ 1, implying
Figure 8.9 Pole-zero plot of (a) a minimum-phase function, (b) a maximum-phase function, and
(c) a mixed-phase function
1
H I ðzÞ ¼ ð8:110Þ
H ðzÞ
In the time domain, this is equivalently expressed as
hI ðnÞ∗hðnÞ ¼ δðnÞ ð8:111Þ
Example 8.36 A system is described by the following difference equation:
yðnÞ ¼ xðnÞ e8α xðn 8Þ
where the constant α > 0. Find the corresponding inverse system function to recover
x(n) from y(n). Check for the stability and causality of the resulting recovery system,
justifying your answer.
Solution
Y ðzÞ
Y ðzÞ ¼ X ðzÞ e8α z8 X ðzÞ; ¼ 1 e8α z8
X ðzÞ
The corresponding inverse system
1 X ðzÞ
H 1 ðzÞ ¼ ¼
ð1 e8α z8 Þ Y ðzÞ
The recovery system is both stable and causal, since all the poles of the system
HI(z) are inside the unit circle.
8.7.7 All-Pass System
Consider a causal stable Nth-order transfer function of the form
aN þ aN1 z1 þ þ zN M ðzÞ

H ðzÞ ¼ ¼ ð8:112Þ
1 þ a1 z1 þ þ aN zN D ðzÞ
Now Dðz1 Þ ¼ 1 þ a1 z þ a2 z2 þ þ aN zN
¼ zN ½aN þ aN1 z1 þ þ zN ð8:113Þ
¼ zN MðzÞ
or

M ðzÞ ¼ zN D z1 ð8:114Þ
Hence,
Dðz1 Þ
H ðzÞ ¼ zN ð8:115Þ
D ðzÞ
and
D ðzÞ
H z1 ¼ zN ð8:116Þ
Dðz1 Þ
Therefore,

H ðzÞ H z1 ¼ 1 ð8:117Þ
Thus,

jH ðωÞj2 ¼ H ejω H ejω ¼ 1 ð8:118Þ
for all values of ω.

In other words, H(z) given by (8.112) passes all the frequencies contained in the
input signal to the system, and hence such a transfer function is an all-pass transfer
function, and the corresponding system is an all-pass system. It is also seen from
(8.112) that if z ¼ pi is a zero of D(z), then z ¼ (1/pi) is a zero of M(z). That is, the
poles and zeros of an all-pass function are reciprocal of one another. Since all the
poles of H(z) are located within the unit circle, all the zeros are located outside the
unit circle.
If x(n) is the input sequence and y(n) the output sequence for an all-pass system,
then
Y ðzÞ ¼ H ðzÞX ðzÞ: ð8:119Þ
Thus,

Y ejω ¼ H ejω X ejω : ð8:120Þ
Since |H(e jω)| ¼ 1, we get

jω jω
Y e ¼ X e ð8:121Þ
We know from Parseval’s relation that the output energy of a LTI system is given
by
X1 ð
21 π jω 2
n¼1
jyðnÞj ¼ Y e dω ð8:122Þ
2π π
ð
1 π jω 2
¼ X e dω ð8:123Þ
2π π
Hence,
X1 X1
n¼1
jyðnÞj2 ¼ n¼1
jxðnÞj2 ð8:124Þ
Thus, the output energy is equal to the input energy for an all-pass system. Hence,
an all-pass system is that it is a lossless system.
Example 8.37 A discrete-time system with poles at z ¼ 0.6 and z ¼ 0.7 and
zeros at z ¼ 1/0.6 and z ¼ 1/0.7 is shown in Figure 8.10. Demonstrate algebra-
ically that magnitude response is constant.
Solution For given pole-zero pattern, the system function is given by
0:42 þ 1:3z1 þ z2

H ap ðzÞ ¼
1 þ 1:3z1 þ 0:42z2
Substituting z ¼ e jω in the above transfer function, we get
Figure 8.10 Pole-zero plot of a second-order all-pass system
0:42 þ 1:3ejω þ e2jω

H ap ejω ¼
1 þ 1:3ejω þ 0:42e2jω
0:42 þ 1:3ejω þ e2jω
H ap ðejω Þ ¼
1 þ 1:3ejω þ 0:42e2jω
2
H ap ðωÞ ¼ H ðejω ÞH ðejω Þ ¼ 1
8.7.8 All-Pass and Minimum-Phase Decomposition
Consider an Nth-order mixed-phase system function H(z) with m zeros outside the
unit circle and (nm) zeros inside the unit circle. Then H(z) can be expressed as
1 1
H ðzÞ ¼ H 1 ðzÞ z1 a∗
1 z a∗
2 z a∗
m ð8:125Þ
where H1(z) is a minimum-phase function as its N poles and (nm) zeros are inside
the unit circle. Eq. (8.125) can be equivalently expressed as
H ðzÞ ¼ H 1 ðzÞð1 z1 a1 Þð1 z1 a2 Þ ð1 z1 am Þ

1 1 1
z a∗ z a∗ 1 z a∗ ð8:126Þ
1 m
ð1 z1 a1 Þð1 z1 a2 Þ ð1 z1 am Þ
In the above equation, the factor H1(z)(1z1a1)(1z1a2) (1z1am) is also a

minimum-phase function, since |a1|, |a2|, . . ., |am| are less than 1, the zeros are inside
1 1
z1 a∗1 z a∗ 2 z a∗ m
the unit circle, and the factor is all-pass.
ð1 z1 a1 Þð1 z1 a2 Þ ð1 z1 am Þ
Thus, any transfer function H(z) can be written as
H ðzÞ ¼ H min ð zÞH ap ðzÞ ð8:127Þ
Hmin(z) has all the poles and zeros of H(z) that are inside the unit circle in addition
to the zeros that are conjugate reciprocals of the zeros of H(z) that are outside the unit
circle, while Hap(z) is an all-pass function that has all the zeros of H(z) that lie outside
the unit circle along with poles to cancel the conjugate reciprocals of the zeros of
H(z) that lie outside the unit circle, which are now contained as zeros in Hmin(z).
Example 8.38 A signal x(n) is transmitted across a distorting digital channel char-
acterized by the following system function:
ð1 0:5z1 Þð1 1:25ej0:8π z1 Þð1 1:25ej0:8π z1 Þ

H d ðzÞ ¼
ð1 0:81z2 Þ
Consider the compensating system shown in Figure 8.11. Find H1C(z) such that the
overall system function G1(z) is an all-pass system.
Solution
H d ðzÞ ¼ H dmin1 ðzÞH ap ðzÞ
ð1 0:5z1 Þ
H dmin1 ðzÞ ¼ ð1:25Þ2 1 0:8ej0:8π z1 1 0:8ej0:8π z1
ð1 0:8z2 Þ
ðz1 0:8ej0:8π Þðz1 0:8ej0:8π Þ
H ap ðzÞ ¼
1 0:8ej 0:8π z1 ð1 0:8ej0:8π z1 Þ
1 ð1 0:81z2 Þ
H 1C ðzÞ ¼ ¼
H dmin1 ðzÞ ð1:25Þ2 ð1 0:5z1 Þð1 0:8eJ0:8π z1 Þð1 0:8eJ0:8π z1 Þ
Then,
G1 ðzÞ ¼ H d ðzÞH 1C ðzÞ ¼ H ap ðzÞ
is an all-pass system.
Figure 8.11 Compensating G1 ( z )

system
x (n ) xd (n) xca (n)

H d (z ) H1C (z)
8.8 One-Sided z-Transform 401
8.8 One-Sided z-Transform
The unilateral or one-sided z-transform, which is appropriate for problems involving

causal signals and systems, is evaluated using the portion of a signal associated with
nonnegative values of time index (n 0). It gives considerable meaning to assume
causality in many applications of the z-transforms.
Definition
The one-sided z-transform of a signal x[n] is defined as
X1
Z þ ½xðnÞ ¼ X þ ðzÞ ¼ n¼0
xðnÞzn ð8:128Þ
which depends only on x(n) for (n 0). It should be mentioned that the two-sided z-
transform is not useful in the evaluation of the output of a non-relaxed system. The
one-sided transform can be used to solve for systems with nonzero initial conditions
or for solving difference equations with nonzero initial conditions. The following
special properties of X+ (z) should be noted.
1. The one-sided transform X+ (z) of x(n) is identical to the two-sided transform X(z)
of the sequence x(n)u(n). Also, since x(n)u(n) is always causal, its ROC and hence
that of X+ (z) are always the exterior of a circle. Hence, it is not necessary to
indicate the ROC of a one-sided z-transform.
2. X+ (z) is unique for a causal signal, since such a signal is zero for n < 0.
3. Almost all the properties of the two-sided transform are applicable to the
one-sided transform, one major exception being the shifting property.
Shifting Theorem for X+ (z) When the Sequence is Delayed by k
If
Z þ ½xðnÞ ¼ X þ ðzÞ,
then
Xk
Z þ ½xðn kÞ ¼ zk X þ ðzÞ þ n¼1
xðnÞzn , k > 0 ð8:129Þ
However, if x(n) is a causal sequence, then the result is the same as in the case of
the two-sided transform and
Z þ ½xðn kÞ ¼ zk X þ ðzÞ ð8:130Þ
Proof By definition,
X1
Z þ ½xðn kÞ ¼ n¼0
xðn kÞ zn
Letting (nk) ¼ m, the above equation may be written as

hX1 X1 i
Z þ ½xðn kÞ ¼ zk m¼0
x ðm Þ z m
þ m¼k
x ðm Þ z m
h Xk i
¼ zk X þ ðzÞ þ n¼1
x ð n Þzn
which proves (8.129). If the sequence x(n) is causal, then the second term on the right
side of the above equation is zero, and hence we get the result (8.130).
Shifting Theorem for X+ (z) When the Sequence is Advanced by k
If
Z þ ½xðnÞ ¼ X þ ðzÞ,
then
h Xk1 i
Z þ ½xðn þ kÞ ¼ zk X þ ðzÞ n¼0
x ðn Þz n
, k>0 ð8:131Þ
Proof By definition
X1
Z þ ½xðn þ kÞ ¼ n¼0
xðn þ kÞzn
Letting (n + k) ¼ m, the above equation may be written as

P1
Z þ ½xðn þ kÞ ¼ zk m¼k xðmÞzm

hP Pk1 i
1 m m
¼ zk m¼0 xðmÞz m¼0 xðmÞz
h P i
¼ zk X þ ðzÞ k1
n¼0 xðnÞz
n
thus establishing the result (8.131).

Final Value Theorem
If a sequence x(n) is causal, i.e., x(n) ¼ 0 for n < 0, then
limn!1 xðnÞ ¼ limz!1 ðz 1ÞX ðzÞ ð8:132Þ
The above limit exists only if the ROC of (z1) X(z) exists.
Proof Since the sequence x(n) is causal, we can write its z-transform as follow
X1
Z ½xðnÞ ¼ n¼0
xðnÞ zn ¼ xð0Þ þ xð1Þz1 þ xð2Þz2 þ : ð8:133Þ
Also
X1
Z ½xðn þ 1Þ ¼ n¼0
xðn þ 1Þ zn ¼ xð1Þ þ xð2Þz1 þ xð3Þz2 þ : ð8:134Þ
8.8 One-Sided z-Transform 403
Hence, we see that
Z ½xðn þ 1Þ ¼ z½Z ½xðnÞ xð0Þ ð8:135Þ
Thus,
Z ½xðn þ 1Þ Z ½xðnÞ ¼ ðz 1ÞZ ½xðnÞ zxð0Þ
Substituting (8.133) and (8.135) for the L.H.S., we have
½xð1Þ xð0Þ þ ½xð2Þ xð1Þz þ ½xð3Þ xð2Þz2 þ ¼ ðz 1ÞX ðzÞ zxð0Þ
Taking the limit as z ! 1, we get
½xð1Þ xð0Þ þ ½xð2Þ xð1Þ þ ½xð3Þ xð2Þ þ ¼ limz!1 ðz 1ÞX ðzÞ xð0Þ
Thus,
xð0Þ þ xð1Þ ¼ limz!1 ðz 1ÞX ðzÞ xð0Þ
or
xð1Þ ¼ limz!1 ðz 1ÞX ðzÞ
Hence,
limn!1 xðnÞ ¼ limz!1 ðz 1ÞX ðzÞ
It should be noted that the limit exists only if the function (z1) X(z) has an ROC
that includes the unit circle; otherwise, system would not be stable and the limn!1x
(n) would not be finite.
Example 8.39 Find the final value of x(n) if its z-transform X(z) is given by
0:5z2
X ðzÞ ¼
ðz 1Þ ð z 2 0:85z þ 0:35Þ
Solution The final value or steady value of x(n) is given by
0:5
xðnÞ ¼ limz!1 ðz 1ÞX ðzÞ ¼ ¼1
ð1 0:85 þ 0:35Þ
The result can be directly verified by taking the inverse transform of the
given X(z)
Example 8.40 The following facts are given about a discrete-time signal x(n) with X
(z) as its z-transform:
(i) x(n) is real and right-sided.
(ii) X(z) has exactly two poles.
(iii) X(z) has two zeros at the origin.

π
(iv) X(z) has a pole at z ¼ 0:5ej3 .
(v) X ð1Þ ¼ 83.
Determine X(z) and specify its region of convergence.
Solution It is given that x(n) is real and that X(z) has exactly two poles with one of
jπ
the pole at z ¼ 12 e 3 . Since, x [n] is real, the two poles must be complex conjugates of
jπ
each other. Thus, the other pole of the system is at Z ¼ 12 e 3 . Also, it is given that X
(z) has two zeros at the origin. Therefore, X(z) will have the following form:
Kz2
X ðzÞ ¼
1 jπ 1 jπ
z e 3 z e 3
2 2
Kz2
¼
1 1
z2 z þ
2 4
for some constant K to be determined. Finally, it is given that Xð1Þ ¼ 83. Substituting
this in the above equation, we get K ¼ 2. Thus,
2z2
X ðzÞ ¼
z2 12 z þ 14
Since x[n] is right-sided, its ROC is jzj > 12 (note that both poles have
magnitude 12).
8.8.1 Solution of Difference Equations with Initial

Conditions
The one-sided z-transform is very useful in obtaining solutions for difference

equations which have initial conditions. The procedure is illustrated with an
example.
Example 8.41 Find the step response of the system

1
y ð nÞ y ð n 1Þ ¼ x ð nÞ
2
with the initial condition y(1) ¼ 1

Solution Taking one-sided z-transforms on both sides of the given equation and
using (8.129), we have
8.9 Solution of State-Space Equations Using z-Transform 405

þ 1 1 þ
Y ðzÞ ½z Y ðzÞ þ yð1Þ ¼ X þ ðzÞ
2
Substituting for X+ (z) and y(1), we have

1 1 þ 1 1
1 z Y ðzÞ ¼ þ
2 2 1 z1
Hence,
1 1 1
Y þ ðzÞ ¼ þ
2 1 1 1 1
1 z 1 z ð1 z1 Þ
2 2
2 1 1
¼ 1

1z 2 1
1 z1
2
Taking the inverse transform, we get

" #
1
Z 1
þ YðzÞ ¼ yðnÞ ¼ 2 Þnþ1 uðnÞ
2
8.9 Solution of State-Space Equations Using z-Transform
For convenience, the state-space equations of a discrete-time LTI system from

Chapter 6 are repeated here:
X ðn þ 1Þ ¼ A X ðnÞ þ b℧ðnÞ ð8:136aÞ

yðnÞ ¼ cX ðnÞ þ d℧ðnÞ ð8:136bÞ
Taking unilateral z-transform Eqs. (8.136a) and (8.136b), we obtain
zX ðzÞ zX ð0Þ ¼ AX ðzÞ þ b℧ðzÞ ð8:137aÞ

Y ðzÞ ¼ cX ðzÞ þ d℧ðzÞ ð8:137bÞ
where
2 3
X 1 ðzÞ
6 X 2 ðzÞ 7
6 7
X ðzÞ ¼ 6
6 ⋮ 7
7
4 X N1 ðzÞ 5
X N ðzÞ
Eq. (8.137a) can be rewritten as

½zI AX ðzÞ ¼ zX ð0Þ þ b℧ðzÞ ð8:138Þ
where I is the identity matrix.

From Eq. (8.138), we get
X ðzÞ ¼ ½zI A1 zX ð0Þ þ ½zI A1 b℧ðzÞ ð8:139Þ
The inverse one-sided z-transform of Eq. (8.139) yields

h i h i
z1 ½X ðzÞ ¼ z1 ½zI A1 zX ð0Þ þ z1 ½zI A1 b℧ðzÞ ð8:140Þ
h i
z1 ½zI A1 zX ð0Þ ¼ An X ð0Þ ð8:141Þ
By using convolution theorem, we obtain

h i Xn1
z1 ½zI A1 b℧ðzÞ ¼ k¼0
An1k b℧ðk Þ ð8:142Þ
Thus,
Xn1
z1 ½X ðzÞ ¼ X ðnÞ ¼ An X ð0Þ þ k¼0
An1k b℧ðkÞ n > 0: ð8:143Þ
Substituting Eq. (8.143) into Eq. (8.136b), we get

Xn1
yðnÞ ¼ cAn X ð0Þ þ k¼0
An1k b℧ðkÞ þ d℧ðnÞ n > 0: ð8:144Þ
Example 8.42 Consider an initially relaxed discrete time system with the following
state-space representation. Find y(n).
" # 2 3" # " #
x 1 ð n þ 1Þ 0 1 x 1 ð nÞ 0
¼4 1 4 5 þ ℧ðnÞ
x 2 ð n þ 1Þ x 2 ð nÞ 1
3 3

1 4 x1 ðnÞ
yðnÞ ¼ þ ℧ðnÞ
3 3 x2 ðnÞ
8.9 Solution of State-Space Equations Using z-Transform 407
Solution " #
0 1 0 1 4
A¼ 1 4 ; b¼ ; c¼ ; d ¼ 1:
1 3 3
3 3 2 3
" " 0 1 ##
z 0 z 1
½zI A ¼ 1 4 ¼ 41 45
0 z z
3 3 3 3
2 3
" #1 4
z 1 z 1
6 7
1
½zI A1 ¼ 1 ¼ 4
3
5
z 4 1 1
3 3 ð z 1Þ z z
3 3
2 3
4
6 z 1 7
6 3 7
6 1 1 7
6 ð z 1Þ z ðz 1Þ z 7
6 3 7
¼6 3 7
6 7
6 1=3 z 7
6 7
4 1 1 5
ð z 1Þ z ðz 1Þ z
3 3
2 3
1=2 3=2 3=2 3=2
6 z1 þ
6 1 z1 17
6 z z 7
¼6 3 377
6 1=2 1=2 7
6 þ
1=2 3=2
7
4 15
z1 1 z1
z z
h 3i 3
1 1
A ¼ z ½zI A z
n
2 3
1 3 3 3
z z z z
6 2 þ 2 2 2 7
6 17
6z 1 1 z1 7
6 z z 7
6 3 3 7
¼ z1 6 7
6 1 1 3 1 7
6 z z z z 7
6 2 þ 2 2 2 7
4 15
z1 1 z1
z z
2 3 3
3
1 3 1 n 3 3 1 n
6 2 þ 2 3
2 2 3 7
6 7
¼6 7
4 1 1 1 n 3 1 1 n5
þ
2 2 3 2 2 3
2 1 33 2 3 3 3

6 7 n 6 7
¼ 4 2 2 5 þ 13 4 2 2 5
1 3 1 1

2 2 2 2
Since the system is initially relaxed, cAnX(0) ¼ 0

82 3 2 39
>>
<6
1 3 3 3 > >
1 4 2 27 1n1k 6 2 2 7=
b¼ 6 7 6 7
4 1 35 þ 3
n1k
cA 41
3 3 > > 1 5>
: > ;
2 2 2 2
2 3 2 3
1 3 " # 3 3 " #
7 0
1 4 6 6 2 27
7 0 n1k 1 4 6 62 27
¼ 4 5 þ 13
3 3 1 3 1 3 3 41 15 1

2 2 2 2
n1k
3 1 1
¼
2 6 3
Hence,
X
n1
3 1 1 n1k
y ð nÞ ¼ ð1Þk þ 1
k¼0
2 6 3
" #
X
n 1
3 1 1 nk
¼ ð1Þk þ 1
k¼0
2 2 3
X n1 nk
n1
3 1X 1
¼ þ1
k¼0
2 2 k¼0
3

P 3 1 1 n Xn1 k
¼ n1k¼0 3 þ1
2 2 3 k¼0

3 1 1 n 1 3n
¼ n þ 1 n > 0:
2 2 3 13
n
3 1 1
¼ nþ ð1 3n Þ þ 1 n > 0:
2 4 3
8.10 Transformations Between Continuous-Time Systems

and Discrete-Time Systems
The transformation of continuous-time system to discrete-time system arises in

various situations. In this section, two techniques, namely,
(i) Impulse invariance technique
(ii) Bilinear transformation technique
are discussed for transforming continuous-time system to discrete-time system.
8.10 Transformations Between Continuous-Time Systems and Discrete-Time Systems 409
8.10.1 Impulse Invariance Method
In this method, the impulse response of an analog filter is uniformly sampled to

obtain the impulse response of the digital filter, and hence this method is called the
impulse invariance method. The process of designing an IIR filter using this method
is as follows:
Step 1: Design an analog filter to meet the given frequency specifications. Let Ha(s)
be the transfer function of the designed analog filter. We assume for simplicity
that Ha(s) has only simple poles. In such a case, the transfer function of the analog
filter can be expressed in partial fraction form as
XN Ak
H a ðsÞ ¼ ð8:145Þ
k¼1 s pk
where Ak is the residue of H(s) at the pole pk.

Step 2: Calculate the impulse response h(t) of this analog filter by applying the
inverse Laplace transformation on H(s). Hence,
XN
ha ðt Þ ¼ k¼1
Ak epk t ua ðt Þ ð8:146Þ
Step 3: Sample the impulse response of the analog filter with a sampling period T.
Then, the sampled impulse response h(n) can be expressed as
hðnÞ ¼ ha ðt Þjt¼nT
PN n ð8:147Þ
¼ k¼1 ðAk epk T Þ uðnÞ
Step 4: Apply the z-transform on the sampled impulse response obtained in Step 3, to
form the transfer function of the digital filter, i.e., H(z) ¼ Z[h(n)]. Thus, the
transfer function H(z) for the impulse invariance method is given by
XN Ak
H ðzÞ ¼ ð8:148Þ
k¼1 1 epk T z1
This impulse invariant method can be extended for the case when the poles are
not simple.
Example 8.43 Consider a continuous system with the transfer function:
sþb
H ðsÞ ¼
ðs þ bÞ2 þ c2
Solution The inverse Laplace transform of H(s) yields

ebt cos ðct Þ for t 0
hð t Þ ¼
0 otherwise
Sampling h(t) with sampling period T, we get

(
ebnT cos ðcnT Þ for n 0
hðnT Þ ¼
0 otherwise
X
1
H ðzÞ ¼ ebnT cos ðcnT Þzn
n¼0
X1
1
¼ ebnT zn ejcnT þ ejcnT
n¼0
2
1X 1 h n n i
¼ eðbjcÞT Z 1 þ eðbþjcÞT Z 1
2 n¼0

1 1 1
¼
2 1 eðbjcÞTz1 1 eðbþjcÞTz1
1
1 ebT cos ðcT Þz
¼ 2
1 2ebT cos ðcT Þz1 þe2bTz
Disadvantage of Impulse Invariance Method

The frequency responses of the digital and analog filters are related by

1 X1 ω þ 2πk
H ejω ¼ H a j ð8:149Þ
T k¼1 T
From Eq. (8.149), it is evident that the frequency response of the digital filter is
not identical to that of the analog filter due to aliasing in the sampling process. If the
analog filter is band-limited with
ω ω

Ha j ¼ 0 ¼ jΩj π=T ð8:150Þ
T T
then the digital filter frequency response is of the form
1 ω
H ejω ¼ H a j jωj π ð8:151Þ
T T
In the above expression, if T is small, the gain of the filter becomes very large.
This can be avoided by introducing a multiplication factor T in the impulse invariant
transformation. In such a case, the transformation would be
hðnÞ ¼ Tha ðnT Þ ð8:152Þ
and H(z) would be

8.10 Transformations Between Continuous-Time Systems and Discrete-Time Systems 411
XN Ak
H ðzÞ ¼ T ð8:153Þ
k¼1 1 epk T z1
Also, the frequency response is
1 ω
H ejω ¼ H a j j ωj π ð8:154Þ
T T
Hence, the impulse invariance method is appropriate only for band-limited filters,
i.e., low-pass and band-pass filters, but not suitable for high-pass or band-stop filters
where additional band limiting is required to avoid aliasing. Thus, there is a need for
another mapping method such as bilinear transformation technique which avoids
aliasing.
8.10.2 Bilinear Transformation
In order to avoid the aliasing problem mentioned in the case of the impulse invariant
method, we use the bilinear transformation, which is a one-to-one mapping from the
s-plane to the z-plane; that is, it maps a point in the s-plane to a unique point in the
z-plane and vice versa. This is the method that is mostly used in designing an IIR
digital filter from an analog filter. This approach is based on the trapezoidal rule.
Consider the bilinear transformation given by
2 ð z 1Þ
S¼ ð8:155Þ
T ð z þ 1Þ
Then a transfer function Ha (s) in the analog domain is transformed in the digital
domain as

H ðzÞ ¼ H a ðsÞ 2 ðz1Þ ð8:156Þ
S¼ T ðzþ1Þ
Also, from Eq. (8.155), we have
2 ð1 þ sÞ
Z¼ ð8:157Þ
T ð1 sÞ
We now study the mapping properties of the bilinear transformation. Consider a

point s ¼ σ + jΩ in the left half of the s-plane. Then, from Eq. (8.157),

ð1 σ þ jΩÞ

jzj ¼ >1 ð8:158Þ
ð1 þ σ jΩÞ
Hence, the left half of the s-plane maps into the interior of the unit circle in the
z-plane (see Figure 8.12). Similarly, it can be shown that the right-half of the s-plane
Left half s-plane

Im z jΩ
z-plane
-1 1 ߪ
0
Figure 8.12 Mapping of the s-plane into the z-plane by the bilinear transformation
maps into the exterior of the unit circle in the z-plane. For a point z on the unit circle,
z ¼ e jω, we have from Eq. (8.155)
2 ðejω 1Þ 2 ω
S¼ ¼ j tan ð8:159Þ
T ð e þ 1Þ
jω T 2
Thus,
2 ω
Ω¼ tan ð8:160Þ
T 2
or

ΩT
ω ¼ 2 tan 1 ð8:161Þ
2
showing that the positive and negative imaginary axes of the s-plane are mapped,
respectively, into the upper and lower halves of the unit circle in the z-plane. We thus
see that the bilinear transformation avoids the problem of aliasing encountered in the
impulse invariant method, since it maps the entire imaginary axis in the s-plane onto
the unit circle in the in the z-plane. Further, in view of the mapping, this transfor-
mation converts a stable analog filter into a stable digital filter.
Example 8.44 Design a low-pass digital filter with 3 dB cutoff frequency at 50 Hz
and attenuation of at least 10 dB for frequency larger than 100 Hz. Assume a suitable
sampling frequency.
Solution Assume the sampling frequency as 500 Hz. Then,
2π f c 2π 50
ωc ¼ ¼ ¼ 0:2π
FT 500
2π f s 2π 100
ωs ¼ ¼ ¼ 0:4π
FT 500
T ¼ 1=500 ¼ 0:002
Prewarping of the above normalized frequencies yields

8.11 Problems 413

0:2π 2 tan ð0:1π Þ
ΩC ¼ tan ¼ ¼ 325
2 T

0:4π 2 tan ð0:2π Þ
Ωs ¼ tan ¼ ¼ 727
2 T
Substituting these values in ðΩs =Ωc Þ2N ¼ 100:1αs 1 and solving for N, we get

log 101 1 0:9542
N¼ ¼ ¼ 1:3643:
2logð0:727=0:325Þ 0:6993
Hence, the order of the Butterworth filter is 2. The normalized low-pass

Butterworth filter for N ¼ 2 is given by
1
H N ðsÞ ¼ pffiffiffi
s2 þ 2s þ 1
The transfer function Hc(s) corresponding to Ωc ¼ 0.325 is obtained by substitut-
ing s ¼ (s/Ωc) ¼ (s/0.325) in the expression for HN(s); hence,
0:1056
H a ðsÞ ¼
s2 þ 0:4595s þ 0:1056
The digital transfer function H(z) of the desired filter is now obtained by using the
bilinear transformation (8.157) in the above expression:
H ðzÞ ¼ H a ðsÞj 2 ðz1Þ

s¼ T ðzþ1Þ
0:1056z2 þ 0:2112z þ 0:1056

H ðzÞ ¼
1000459:6056z2 1999999:7888z þ 999540:6056
8.11 Problems
1. Find the z-transform of the sequence x(n) ¼ {1,2,3,4,5,6,7}.

2. Find the z-transform and ROC of the sequence x(n) tabulated below.
n 2 1 0 1 2 3 4
x(n) 1 2 3 4 5 6 7
3. Find the z-transform of the signal x(n) ¼ [3(3)n4(2)n].

4. Find the z-transform of the sequence x(n) ¼ (1/3)n1u(n1).
5. Find the z-transform of the sequence

1, 0nN1
x ð nÞ ¼
0, otherwise
6. Find the z-transform of the following discrete-time signals, and find the ROC for
each.
n n
(i) xðnÞ ¼ 12 uðnÞ þ 3 14 uðn 1Þ

(ii) xðnÞ ¼ 14 δðnÞ þ δðn 2Þ 13 δðn 3Þ
n
(iii) xðnÞ ¼ ðn þ 0:5Þ 12 uðn 1Þ 13 δðn 3Þ
7. Find the z-transform of the sequence x(n) ¼ nan1u(n1).
8. Find the z-transform of the sequence x(n) ¼ (1/4)n+1u(n).
9. Find the z-transform of the signal x(n) ¼ [(4)n+13(2)n1].
10. Determine the z-transform and the ROC for the following time signals. Sketch
the ROC, poles, and zeros in the z-plane.
π

(i) xðnÞ ¼ sin 3π4 n 8 u½ n 1
π

(ii) xðnÞ ¼ ðn þ 1Þ sin 3π4 n þ 8 u½n þ 2:
11. Find the inverse z-transform of the following, using partial fraction expansions:
(i) X ðzÞ ¼ ðzþ0:2
zþ0:5
Þðz2Þ , jzj > 2
1
(ii) X ðzÞ ¼ 1þ3z1þz
1 þ2z2 , jzj > 2
z2 þz
(iii) X ðzÞ ¼ ðz3Þðz2Þ , jzj > 3
zðzþ1Þ
(iv) X ðzÞ ¼ , jzj > 12
ðz12Þðz13Þ
12. Find the inverse z-transform of the following using the partial fraction
expansion.
(i) X ðzÞ ¼ ðz1Þzðz4Þ , jzj < 1
(ii) X ðzÞ ¼ ðz1zÞðþ2z3

2
z3Þðz4Þ , for ðaÞ jzj > 4 and ðbÞ jzj < 1
(iii) X ðzÞ ¼ 3z2 4zþ1
z
, jzj < 13
13. Determine all the possible signals that can have the following z-transform:
z2
X ðzÞ ¼
z2 0:8z þ 0:15
8.11 Problems 415
14. Find the stability of the system with the following transfer function:
z
H ðzÞ ¼
z3 1:4z2 þ 0:65z 0:1
15. The transfer function of a system is given as
z þ 0:5
H ðzÞ ¼
ðz þ 0:4Þðz 2Þ
Specify the ROC of H(z) and determine h(n) for the following conditions:
(i) The system is causal.
(ii) The system is stable.
(iii) Can the given system be both causal and stable?
16. A causal LTI system is described by the following difference equation:
1 1
y ð nÞ y ð n 2Þ ¼ x ð n 2Þ x ð nÞ
4 4
Determine whether the system is an all-pass system.
17. In the system shown in Figure P8.1, if S1 is a causal LTI system with system
function,

1 1 3 1
H ðzÞ ¼ 1 z 1 z 1 3z1
2 4
Determine the system function for a system S2 so that the overall system is an
all-pass system.
x (n ) y (n )
S1 S2
Figure P8.1 Cascade connection of two systems S1 and S2
18. The transfer function of a system is given by
1
H ðzÞ ¼
z2 þ 5z þ 6
Determine the response when x(n) ¼ u(n). Assume that the system is initially
relaxed.
19. A causal LTI system is described by the following difference equation:
y ð nÞ y ð n 1Þ y ð n 2Þ ¼ x ð n 1Þ
Is it a stable system? If not, find a noncausal stable impulse response that

satisfies the difference equation.
20. Using the one-sided z-transform, solve the following difference equation:

1
y ð nÞ yðn 2Þ ¼ uðnÞ, yð1Þ ¼ 0, yð2Þ ¼ 2
9
21. Consider a continuous system with the transfer function
1
H ðsÞ ¼
Determine the transfer function and pole-zero pattern for the discrete-time
system by using the impulse invariance technique.
22. Design a low-pass Butterworth filter using Bilinear transformation for the
following specifications:
Passband edge frequency: 1000 Hz
Stopband edge frequency: 3000 Hz
Sampling frequency: 8000 Hz
23. Consider an initially relaxed discrete time with the following state-space repre-
sentation. Find y(n).
" # 2 3
x 1 ð n þ 1Þ 0 1 " x ð nÞ # " 0 #
1
¼ 4 1 35 þ ℧ðnÞ
x 2 ð n þ 1Þ x 2 ð nÞ 1
8 4
" #
1 3 x 1 ð nÞ
y ð nÞ þ ℧ð n Þ
8 4 x 2 ð nÞ
1. Write a MATLAB program using the command residuez to find the inverse of the
following by partial fraction expansion:
Further Reading 417
16 4z1 þ z2
X ðzÞ ¼
8 þ 2z1 2z2
2. Write a MATLAB program using the command impz to find the inverse of the
following by power series expansion:
15z3
X ðzÞ ¼
15z3 þ 5z2 3z 1
3. Write a MATLAB program using the command z-plane to obtain a pole-zero plot
for the following system:
1 þ 13 z1 þ 57 z2 32 z3

H ðzÞ ¼
1 þ 52 z1 13 z2 35 z3
4. Write a MATLAB program using the command freqz to obtain magnitude and
phase responses of the following system:
1 3:0538z1 þ 3:8281z2 2:2921z3 þ 0:5507z4

H ðzÞ ¼
1 4z1 þ 6z2 4z3 þ z4
Further Reading
1. Lyons, R.G.: Understanding Digital Signal Processing. Addison-Wesley, Reading (1997)

2. Oppenheim, A.V., Schafer, R.W.: Discrete-Time Signal Processing, 2nd edn. Prentice-Hall,
Upper Saddle River (1999)
4. Hsu, H.: Signals and Systems, Schaum’s Outlines, 2nd edn. McGraw-Hill, New York (2011)
5. Kailath, T.: Linear Systems. Prentice-Hall, Englewood Cliffs (1980)
6. Zadeh, L., Desoer, C.: Linear System Theory. McGraw-Hill, New York (1963)
Index
A B
Aliasing, 273, 347, 410–412 Band-pass filter, 230, 257–260, 269, 411
All-pass decomposition, 399–400 Band-stop filter, 231, 260, 261, 263, 264
All-pass system, 397–400, 415 Basic continuous-time signals
Amplitude, 5 complex exponential function, 24
Amplitude demodulation, 163–165 ramp function, 22
Amplitude modulation (AM), 155, 162–164 real exponential function, 23–24
Analog filter design, 237, 238, 240, 242, 244, rectangular pulse function, 22–23
250, 252–255, 258, 260–263 signum function, 23
band-pass, 227, 230, 252–264, 269, 411 sinc function, 24–27, 137
band-stop, 31, 227, 231, 252–264, 411 unit impulse function, 21–22, 136, 187
Butterworth low-pass filter, 62, 163, 228, unit step function, 20, 22, 29, 98, 188
232–237, 249, 263, 413 Basic sequences
Chebyshev analog low-pass filter arbitrary, 21, 50, 77, 113, 139, 176, 178,
type 1 Chebyshev low-pass filter, 237, 282, 286, 318, 353, 360, 380
238, 240, 255, 258, 261 exponential and sinusoidal, 277, 278, 301
type 2 Chebyshev filter, 242, 244, unit sample, 286, 365
250, 253, 261 unit step, 20, 49, 68, 286
elliptic analog low-pass filter, 227, Bessel filter, 248–250
245–247, 251 BIBO stability theorem, 284, 285, 389, 391
high-pass, 31, 227–229, 231, 252–264, Bilateral Laplace transform, 171, 172
269, 411 Bilinear transformation, 411, 413, 416
low-pass, 227, 229, 231–264 Block diagram representation
notch, 31, 266–267 described by differential equations, 82–93
specifications of low-pass filter, 232, 259, Butterworth analog low-pass filter, 233–237
261, 263
transformations
low-pass to band-pass, 252 C
low-pass to band-stop, 252, 260, Cauchy’s residue theorem, 374–375
262, 263 Causal and stable conditions, 391
low-pass to high-pass, 252, 254 Causality, 41, 77, 82, 86–87, 107, 204–206,
low-pass to low-pass, 252, 253 208, 285, 297, 298, 311, 391, 401
Analog filter types comparison, 249–252 Causality for LTI systems, 77, 294–297
Application examples, vii, 10, 30, 162–164 Causality theorem, 391
Associative property, 51–52, 292 Characteristic equation, 300

K. D. Rao, Signals and Systems, https://doi.org/10.1007/978-3-319-68675-2
420 Index
Chebyshev analog low-pass filter, 237–245 Convolution of two sequences, 151, 318, 362
Classification of signals Convolution sum, 271, 287, 289, 332
analog and digital signals, 5, 271–276, 346 Convolution theorem, 220, 318, 320,
causal, non-causal and anti-causal signals, 12 366, 406
continuous time and discrete time signals, 5 Correlation, 30, 31, 33, 319, 363, 366
deterministic and random signals, 20 Correlation of discrete-time signals, vii
energy and power signals, 13–20 Correlation of two sequences, 363
even and odd signals, 9–12, 141 Correlation theorem, 319
periodic and aperiodic signals, 6–9
Commutative property, 46
Complex Exponential Fourier Series, 111–128 D
Computation of convolution integral using Direct form I, 96
MATLAB, 70–74 Direct form II, 95–97
Computation of convolution sum using Discrete-time Fourier series (DTFS)
MATLAB, 291 Fourier coefficients, 350
Computation of linear convolution multiplication, 315
graphical method, 289 periodic convolution, 313–316, 318
matrix method, 288 symmetry properties, 315
Conjugate of complex sequence, 364 Discrete-time Fourier transform (DTFT)
Continuous Fourier Transform linearity, 315
convergence of Fourier transform, 135–136 Discrete time LTI systems in z-domain,
Continuous Fourier transform properties 385–400
convolution property, 151 Discrete-time signal, 271, 315–331, 414
differentiation in frequency, 146, 147 Discrete-time signals classification
differentiation in time, 143 energy and power signals, 13, 279–281
duality, 154 finite and infinite length, 276
frequency shifting, 143 periodic and aperiodic, 6–9, 278
integration, 148 right-sided and left-sided, 277
linearity, 139, 151, 158, 160 symmetric and anti-symmetric, 276
modulation, 155, 158 Discrete-time system characterization
Parseval’s theorem, 149, 150, 157 non-recursive difference equation, 298
symmetry properties, 119, 158 recursive difference equation, 298, 336
time and frequency scaling, 143, 158 Discrete-time systems classification
time reversal, 158 causal, 284, 298
time shifting, 142, 158, 168 linear, 282, 286–289, 291–297
Continuous time signal, 113–133 stable, 284
complex exponential Fourier series, 111–128 time-invariant, 283–284
convergence of Fourier series, 113 Discrete transformation, 281
properties of Fourier series, 113–128 Distributive property, 50–51, 75
trigonometric Fourier series, 128–133, 166 Down-sampler, 306
symmetry conditions, 129–133
Continuous-time systems
causal system, 48, 77 E
invertible system, 49, 79 Elementary operations on signals, 1–5
linear systems, 42–48 Elliptic analog low-pass filter, 246
memory and memoryless system, 49 Energy and power signals, 13–20, 279
stable system, 49, 78 Examples of real world signals and systems
time–invariant system, 43–48, 105 audio recording system, 32
Convergence of the DTFT, 317 global positioning system, 33
Convolution integral heart monitoring system, 34–36
associative property, 51–52 human visual system, 36
commutative property, 50 location-based mobile emergency
distributive property, 50–51, 75 services system, 33–34
graphical convolution, 58–70 magnetic resonance imaging, 36–37
Index 421
F Impulse response, 49, 53–57, 62, 66–68, 73–75,

Filtering, 31, 227, 233, 235, 236 77–79, 88, 92–95, 106, 109, 153, 161,
Final value theorem, 187, 200, 222 193, 202, 204, 217, 227, 229, 267,
Fourier transform, 318 286–288, 294–297, 324, 330, 332
Fourier transform of discrete-time signals, Impulse step responses, 304, 305
158, 315, 317–331, 335, 342 Initial Value Theorem, 186–187, 370–371
convergence of the DTFT, 317 Input-output relationship, 271, 282, 286–288,
properties of DTFT 293
for a complex sequence, 320–322 Interconnected systems, 74–76
for a real sequence, 322–331 Inverse discrete Fourier transform, 111, 135,
theorems on DTFT 136, 138
convolution theorem, 318, 320 Inverse Fourier transform, 317
correlation theorem, 319, 320 Inverse Laplace transform
differentiation in frequency, 158, 318 partial fraction expansion, 194–202, 209,
frequency shifting, 315, 318, 320, 342 211, 375–379, 416
linearity, 317, 320, 326 partial fraction expansion using MATLAB,
Parseval’s theorem, 319, 320, 328, 329 201–202
time reversal, 318, 320 partial fraction expansion with multiple
time shifting, 320, 324, 335 poles, 195–201
windowing theorem, 318 partial fraction expansion with simple
Frequency division multiplexing (FDM), 32, poles, 195, 376
164, 166 Inverse system, 79–81, 205, 395, 396
Frequency response computation using Inverse z-transform
MATLAB, 338, 341, 342, 346 Cauchy’s residue theorem, 374–375
Frequency response from poles and zeros, modulation theorem, 372
264–265, 388–389 Parseval’s relation, 126, 372–374, 398
Frequency response of continuous time partial fraction expansion, 375–379, 414
systems, 111, 159–162 partial fraction expansion using MATLAB,
Frequency response of discrete-time systems 379–380
computation using MATLAB, 338, 341, power series expansion, 379–383
342, 346 power series expansion using MATLAB, 383
Frequency shifting, 115, 143, 145, 158, 315,
318, 342
L
Laplace transform, 117, 144, 151, 155,
G 172, 174–191, 200, 208, 215,
Generation of continuous-time signals using 222, 402, 403
MATLAB, 28–30 block diagram representation, 218–219
Graphical convolution, 58–70 definition of, 171–225
existence of, 172
inter connection of systems, 218
H properties, 171, 178–187
Half-wave symmetry, 119–124, 126, convolution in the frequency domain,
127, 132 117, 155, 181, 182
High-pass filter, 231, 257 convolution in the time domain,
151, 181
differentiation in the s-domain, 180,
I 184, 190
Imaginary part of a sequence, 365 differentiation in the time domain,
Impulse and step responses, 286 144, 179, 184
Impulse and step responses computation division by t, 180
using MATLAB, vii, 92, 225, final value theorem, 187, 200, 222,
304, 305, 312 402, 403
422 Index
Laplace transform (cont.) Modulation property, 155–158

initial value theorem, 186–187 Modulation theorem, 372
integration, 184, 187 Multiplexing and demultiplexing, 32
linearity, 178, 184, 208
properties of even and odd functions,
182–183 N
shifting in the s-domain, 178, 184, Nonperiodic signals, 111, 133–158
189–191 Non-recursive difference equation, 298
time scaling, 179, 184 Notch filter, 266–267
time shifting, 178, 184, 188
region of convergence, 174–176, 203,
208, 223 O
finite duration signal, 174 One-sided z-transform, 384, 393, 401, 404–405
left sided signal, 175–177 properties
right sided signal, 172, 175–177 shifting theorem, 384, 393, 401
strips parallel to the jΩ axis, 174 final value theorem, 402–404
two sided signal, 176, 177 solution of linear difference equations
region of convergence (ROC), 173 with initial conditions, 401, 404–405
relationship to Fourier transform, 172–173
representation of Laplace transform in the
s-plane, 173, 188 P
system function, 202, 204 Parseval’s relation, 126, 372–374, 398
block diagram representation, 218–219 Parseval’s theorem, 118, 149, 150, 157, 319,
interconnection of systems, 218 320, 328
table of properties, 184 Partial fraction expansion, 378
transfer Function, 202–204, 223, 235 Partial fraction expansion using MATLAB,
unilateral Laplace transform, 171, 172, 201–202, 379–380
183–186, 215 Particular solution, 85, 89, 90, 300–303
differentiation property, 183–186, 215 Periodic convolution, 76, 107, 117, 119,
Linear constant coefficient difference 313–316, 318
equations, 298, 299 Phase and group delays, 333
Linear constant-coefficient differential Poles and zeros, 173, 192, 227, 235, 240,
equations, 82–84, 159, 171, 207–210 242, 243, 246, 265, 386–388, 395,
Linearity, 41, 43, 44, 50, 86, 105, 113, 119, 398, 400, 414
139, 140, 151, 158, 160, 178, 184, Pole-zero pairing, 388
193, 208, 217, 282, 285, 311, 315, Pole-zero placement, 227, 264–267
317, 320, 326, 360, 364–366 Power series expansion using MATLAB, 417
Linearity property of the Laplace transform, Properties of the convolution integral, 50–57, 151
217 Properties of the convolution sum, 291–295
Low-pass to band-pass, 257 Properties of the impulse function
Low-pass filter, 164, 203, 225, 227–229, sampling property, 21, 52, 67, 136
231–264, 267, 268, 341, 351 scaling property, 22
LTI discrete-time systems, vii, 271, 286–289, shifting property, 21, 324, 335
291–297, 304, 332, 333, 354, 386 Proposition, 7, 18, 278
LTI systems with and without memory, 77
Q
M Quantization and coding, 274–276
Matrix method, 288 Quantization error, 275
Maximum-phase systems, 395
Minimum-phase decomposition, 399–400
Minimum-phase systems, 395, 399 R
Mixed-phase systems, 395 Rational transfer function, 386
Modulation, 158 Rational z-transform, 354, 374
Modulation and demodulation, 31, 170 Real part of a sequence, 364
Index 423
Reconstruction of a band-limited signal from inverse, 79–82, 205, 395, 396

its samples, 350 maximum-phase, 395
Recursive difference equation, 298, 336, 338 minimum-phase, 395
Region of convergence (ROC), 173–181, mixed-phase, 395, 399
184, 185, 188, 192, 195, 197, 199, stable, 49, 78, 204–210, 218, 219, 223,
203–208, 210, 222, 354–364, 366, 285, 296, 297, 311, 324, 325, 389,
369–371, 373, 377, 379, 389, 391, 391–394, 397, 415, 416
392, 401–404, 413–415
Representation of signals in terms of
impulses, 41–42 T
Theorems on DTFT, 317–320, 329
Time-invariant, 41, 43–48, 50, 77–82, 88,
S 283, 286–289, 291–297, 395
Sampling Time reversal, 3–5, 115, 119, 158, 318, 320,
continuous time signals, vii, 5, 271, 273, 360
305, 344 Time scaling, 2–3, 115, 119, 179, 184
discrete time signals, 271–276, 305–307 Time shifting, 2, 17, 18, 113, 119, 120,
frequency domain, vii, 306, 344–347 142, 158, 168, 178, 184, 188,
Nyquist rate, 273 315, 318, 324, 335
Sampling frequency, 271, 273–275, 346, Transfer function, 202–204, 223, 227,
412, 416 234–241, 243–246, 248, 252, 254,
Sampling in frequency-domain 256–259, 261, 263, 267, 385–386,
aliasing, 273, 347, 410 388, 389, 394, 395, 397–399, 409,
over-sampling, 347 411, 413, 415
sampling theorem, 273, 346 Transformation, 31, 42, 111, 227, 252–255,
under sampling, 346 257, 260, 263, 281, 285, 286, 325,
Sampling period, 271, 273, 306, 346, 409 408–413, 416
Sampling theorem, 273–274, 346, 347
Scalar multiplication, 93
S-domain, 178, 180, 184, 189–191 U
Single-side-band (SSB) AM, 164 Under sampling, 346
Singularity functions, 41, 95 Unilateral Laplace transform
Solution of difference equations differentiation property, 183–186, 208
characteristic equation, 300 Unit doublet, 95, 97
complementary solution, 85, 90, 300 Unit impulse response, 95
using MATLAB, 91–92 Unit ramp, 22
particular solution, 300–303 Unit sample sequence, 286, 365
Solution of linear differential equations Unit step response, 68
using MATLAB, 216 Unit step sequence, 286
Stability and causality, 208, 295–297, 311, Unstable system, 78, 391, 392
389, 391, 396 Up-sampling, 312
Stability and causality of LTI systems in terms
of the impulse response, 295–297
Stability for LTI systems, 77–79 W
Step and impulse responses, 49 Windowing theorem, 318, 320
Step response, 49, 68, 89, 91–94, 106,
107, 110, 286, 302, 304, 305,
311, 404 Z
Systems Zero locations, 206, 264, 266
all-pass, 397–400, 415 Zero-order hold, 272–274
causal, 48, 77, 78, 86, 107, 172, 204–210, Z-transform
219, 223, 284, 288, 294–298, 324, definition of, 353, 359, 360, 381
391–394, 397, 415, 416 properties of, 360–366
424 Index
Z-transform (cont.) imaginary part of a sequence, 365

region of convergence (ROC), 354–359, linearity, 360, 364–366
361–366, 368–371, 374, 377, 379, real part of a sequence, 364
413, 414 scaling in the z-domain, 361
Z-transform properties time reversal, 360, 363, 366
conjugate of a complex sequence, 364 time shifting, 361, 366
convolution of two sequences, 362 Z-transforms of commonly-used sequences
correlation of two sequences, 363 unit step sequence, 366
differentiation in the z-domain, 361

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K.

Signals and Systems

ISBN 978-3-319-68674-5 ISBN 978-3-319-68675-2 (eBook)

Library of Congress Control Number: 2017958547

Mathematics Subject Classiﬁcation (2010): 94A12; 94A05; 93C55; 93C20; 35Q93

© Springer International Publishing AG, part of Springer Nature 2018

Printed on acid-free paper

detailed in Chapter 7. Chapter 8 describes the z-transform and analysis of LTI

Hyderabad, India K. Deergha Rao

2.5.8 Determining Impulse Response and Step

4.1.2 The Unilateral Laplace Transform . . . . . . . . . . . . . . . . . 172

5 Analog Filters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 227

6.5 Characterization of Discrete-Time Systems . . . . . . . . . . . . . . . . . 297

8.5.1 Modulation Theorem in the z-Domain . . . . . . . . . . . . . . 372

1.1 What is a Signal?

1.2 What is a System?

A system is deﬁned mathematically as a transformation that maps an input signal x(t)

1.3 Elementary Operations on Signals

In many practical situations, signals related by a modiﬁcation of the independent

© Springer International Publishing AG, part of Springer Nature 2018 1

Figure 1.1 Schematic representation of a system

1.3.1 Time Shifting

1.3.2 Time Scaling

The compression or expansion of a signal is known as time scaling. The time-scaling

Figure 1.3 Illustration of time scaling

compressed in time by a factor 2, x(2t) is obtained as shown in Figure 1.3(b). If the

1.3.3 Time Reversal

1.4 Classiﬁcation of Signals

Signals can be classiﬁed in several ways. Some important classiﬁcations of signals

1.4.1 Continuous-Time and Discrete-Time Signals

Continuous-time signals are deﬁned for a continuous of values of the independent

1.4.2 Analog and Digital Signals

Figure 1.5 (a) Continuous-time signal, (b) discrete-time signal

1.4.3 Periodic and Aperiodic Signals

(v) xðt Þ ¼ ejð5tþ2Þe

Figure 1.6 (a) Periodic signal, (b) aperiodic signal

(i) For x1(t) to be periodic,

The fundamental period of x2(t) is

The fundamental period of the signal x(t) is 4T1 ¼ 3T2 ¼ 2 seconds.

Example 1.5 Consider the signals

Determine whether x3(t) ¼ x1(t)x2(t) is periodic or aperiodic. If periodic, ﬁnd the

1 jð7πt=5Þ 1 jð3πt=5Þ 1 jð3πt=5Þ 1 jð7πt=5Þ ejð13πt=5Þ

1.4.4 Even and Odd Signals

The even part and odd part of a signal are

yðt Þ ¼ xe ðt Þxo ðt Þ

Hence, y(t) is an odd signal.

yðt Þ ¼ xe ðt Þxe ðt Þ

Hence, y(t) is an even signal.

(iii) Multiplication of an odd signal by an odd signal produces an even signal.

yðt Þ ¼ x0 ðt Þxo ðt Þ

Hence, y(t) is an even signal.

From Figure 1.6(b), it is also obvious that

xðt Þ þ xðt Þ ej2t þ ej2t

x(t) x(t) x(t)

(a) (b) (c)

1.4.5 Causal, Noncausal, and Anticausal Signal

1.4.6 Energy and Power Signals

The total energy of a continuous-time signal x(t) can be deﬁned as

for a complex valued signal

The energy of the signal is given by

Figure 1.10(a) Energy x(t)

Figure 1.10(b) Power x(t)

1 T=2 1 cos ð4πtÞ