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Rao K.D. Signals and Systems (Springer, 2018) (ISBN 9783319686745) (O) (434s) MNW
Rao K.D. Signals and Systems (Springer, 2018) (ISBN 9783319686745) (O) (434s) MNW
Deergha Rao
Signals
and Systems
K. Deergha Rao
This book is published under the imprint Birkhäuser, www.birkhauser-science.com by the registered
company Springer International Publishing AG part of Springer Nature.
The registered company address is: Gewerbestrasse 11, 6330 Cham, Switzerland
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
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
x Contents
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
Index . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 419
Chapter 1
Introduction
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.
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.
()
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
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.
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
x(t) x(-t)
2 2
-2 2
t -2 2
t
(a) (b)
Figure 1.4 Illustration of 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
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
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
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 x ðt 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)
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 þ T
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
2π 1
T1 ¼ ¼
4π 2
The fundamental period of x2(t) is
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.
ejðπtÞ ejðπtÞ
x2 ð t Þ ¼
2j 2j
Then,
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
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 x e ð t Þ þ x o ð t Þ ð1:3Þ
xðt Þ þ xðt Þ
x e ðt Þ ¼ ð1:3aÞ
2
xðt Þ xðt Þ
xo ðt 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)
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
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
1.4 Classification of Signals 11
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
t t
t
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 Classification of Signals 13
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
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 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
0 1 2 Time
14 1 Introduction
-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|
8 , 1 < t < 1.
(ii) x(t) ¼ 2e
< 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
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
1.4 Classification of Signals 17
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
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
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
1.4 Classification of Signals 19
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
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
Hence,
A21 A22
Ps ¼ þ ð1:13Þ
2 2
20 1 Introduction
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
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
t
1.5 Basic Continuous-Time Signals 21
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).
• Shifting property
ð1
xðt Þδðt t 0 Þdt ¼ xðt 0 Þ ð1:18Þ
1
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
r ðt Þ ¼ tuðt Þ ð1:22bÞ
- 1 0
1 t
0 t
-1
1 jt j T 1
xð t Þ ¼ ð1:23Þ
0 jt j > T 1
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
Hence
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
sin ðπt Þ
Sincðt Þ ¼ ð1:28Þ
πt
which is shown in Figure 1.19
1.5 Basic Continuous-Time Signals 25
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
x(t)
t
1
(c)
(c )
u(1-t)
1
t
Example 1.13 Determine and plot the even and odd components of the following
continuous-time signal
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
1.5 Basic Continuous-Time Signals 27
( )
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
(b) 4þjt
δ ð t 1Þ
Ð3jt
1
(c) 1 ð4t 3ÞÞδðt 1Þdt
Solution
ð0Þ
(a) sin
0þ3 δðt Þ ¼ 0
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)
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))
xlabel ('t (seconds)');
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)
clear all;clc;
u=inline('(t>=-5)& (t<5)','t');
t=-10:0.01:10;
plot(t,u(t))
xlabel ('t (seconds)');
ylabel ('Ámplitude')
axis([-10 10 -2 2])
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.
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
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
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
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.
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
1
0
−1
0 0.2 0.4 0.6 0.8 1 1.2 1.4 1.6 1.8 2
⫻10 4
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
⫻10 4
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).
Figure 1.30 (a) MRI imaging system, (b) MRI image with brain tumor
1.9 Problems
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
1
Ámplitude
-1
-2
-3
-4
-5
-10 -8 -6 -4 -2 0 2 4 6 8 10
t (seconds)
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
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
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
Principle of homogeneity:
Principle of superposition:
Linearity:
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
Time-invariance:
yðt t 0 Þ ¼ ðt t 0 Þxðt t 0 Þ
44 2 Continuous-Time Signals and Systems
For example, for an input x1(t) ¼ x(t t0), the output y1(t) can be written as
Solution
Linearity:
Let x1(t) and x2(t) be two distinct inputs applied to the system, then
If an input equal to sum of the inputs ax1(t), bx2(t),x(t) ¼ ax1(t) þ bx2(t) is applied,
then
Time-invariance:
For example, for an input x1(t) ¼ x(t t0), the output y1(t) can be written as
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
If an input equal to sum of the inputs ax1(t), bx2(t),
x(t) ¼ ax1(t) þ bx2(t) is applied, then
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
Hence, the system is linear.
(ii) 3
yð t Þ ¼ xð t Þ
2
Let x1(t) and x2(t) be two distinct inputs applied to the system, then
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
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
46 2 Continuous-Time Signals and Systems
For example, for an input x1(t) ¼ x(t t0), the output y1(t) can be written as
Now, for an input x1(t) ¼ x(t 6), the output y1(t) can be written as
Ð3
y 1 ð t Þ ¼ ℜ½ x 1 ð t Þ ¼ 3 δðt 6Þdt ¼ 0
yðt 6Þ 6¼ y1 ðt Þ
For example, for an input x1(t) ¼ x(t t0), the output y1(t) can be written as
For example, for an input x1(t) ¼ x(t t0), the output y1(t) can be written as
dxðt t 0 Þ
yð t t 0 Þ ¼
dt
For example, for an input x1(t) ¼ x(t t0), the output y1(t) can be written as
dxðt t 0 Þ
y 1 ð t Þ ¼ ℜ½ x 1 ð t Þ ¼
dt
y ð t t 0 Þ ¼ y1 ð t Þ
2 4 t
48 2 Continuous-Time Signals and Systems
(t)
2 4 8 t
-2
-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.
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
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.
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.
A system is said to be invertible if the input can be recovered from its output.
Otherwise the system is noninvertible system.
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):
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
From the linearity property of the system, Eq. (2.12) can be rewritten as
ð1
yð t Þ ¼ xðτÞℜ½δðt τÞdτ ð2:13Þ
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
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
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
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.
Proof By definition
ð1
xðt Þ∗δðt Þ ¼ xðτÞδðt τÞ dτ ð2:31Þ
1
Hence
and
ð1
xðt t 1 Þ∗hðt t 2 Þ ¼ xðτ t 1 Þhðt τ t 2 Þ dτ ð2:35Þ
1
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
54 2 Continuous-Time Signals and Systems
-2 0 2 tt
2.3 The Convolution Integral 55
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
1 2 4 5 t
56 2 Continuous-Time Signals and Systems
1 2 4 5 t
y(t-2)
t
1 2 4 5
t
1 2 4 5
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
1
1
t-3 0 t+1 0
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).
2.3 The Convolution Integral 59
(0)= ( )h(0 − ) = 0
h( − )| =0 ( )
1
-4 0 4 8
-4 -3 0 4 8
-4 -2 0 4 8
-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
-4 -2 0 4 8
-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
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
Figure 2.11 (continued)
62 2 Continuous-Time Signals and Systems
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
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
64 2 Continuous-Time Signals and Systems
( ) 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
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 2.13 If x(t) and h(t) are odd signals, then show that
y(t) ¼ x(t) * h(t) is an even signal.
66 2 Continuous-Time Signals and Systems
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
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
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
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
68 2 Continuous-Time Signals and Systems
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
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
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.
70 2 Continuous-Time Signals and Systems
h (t) x (t)
1
1
t -2T -T 0 T 2T t
-1 1
y(t)
-3 -2 -1 0 1 2 3
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
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;
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.
72 2 Continuous-Time Signals and Systems
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];
[y ty]=convint(x,tx,h,th);
figure;
plot(ty,y);
xlabel('Time')
ylabel('Amplitude');
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);
[y ty]=convint(x,tx,h,th);
figure;
plot(ty,y);
xlabel('Time')
ylabel('Amplitude');
74 2 Continuous-Time Signals and Systems
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
(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.
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
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
h ð t Þ ¼ h1 ð t Þ þ h 2 ð t Þ ð2:44Þ
(a) (b)
Figure 2.22 (a) Parallel connection of two systems. (b) Equivalent system
76 2 Continuous-Time Signals and Systems
Example 2.20 An LTI system consists of two subsystems in cascade. The impulse
responses of the subsystems are, respectively, 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
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 y ðt þ T Þ
ðT
Solution (i) yðt þ T Þ ¼ x1 ðτÞx2 ðt þ T τÞdτ
0
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
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
ð1
hðτÞdτ < 1 ð2:49Þ
1
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
2.4 Properties of Linear Time-Invariant Continuous-Time System 79
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
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
Hence, the input-output relation for the inverse system shown in Figure 2.24 is
dyðt Þ
xðtÞ ¼ ð2:53Þ
dt
Example 2.24
X
1
hð t Þ ¼ hk δðt kT Þ
k¼0
X
1
gð t Þ ¼ gk δðt kT Þ
k¼0
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
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,
1
g0 ¼ ,
h0
g0 h1 h1 h1
g1 ¼ ¼ ¼ 2 ,
h0 h0 h0 h0
! !
1 1 h1 1 h2 h21
g0 h 2 þ g 1 h 1 þ g 2 h 0 ¼ h2 2 h1 ¼
h0 h0 h0 h0 h0 h20
Delay
T -a
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
xðt Þ ¼ yðt=2Þ
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
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
dV 1 dV 0
r 1 r 2 c1 r 1 r 2 c1 þ ðr1 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 Systems Described by Differential Equations 85
The general solution of Eq. (2.54) for a particular input x(t) is given by
yð t Þ ¼ y c ð t Þ þ y p 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
yp ðt Þ ¼ e2t
yc ðt Þ ¼ Bekt
dyc ðt Þ
þ yc ð t Þ ¼ 0
dt
86 2 Continuous-Time Signals and Systems
yields
Bkekt þ Bekt ¼ 0
ðk þ 1ÞBekt ¼ 0
Thus, k ¼ 1 and
yc ðt Þ ¼ Bet
Now,
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 0 ð2:61Þ
2.5.4 Causality
2.5.5 Time-Invariance
dyðt Þ
þ yðt Þ ¼ xðt Þ yð0Þ ¼ 0 ð2:63Þ
dt
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
y1 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
dy2 ðt Þ
þ y2 ðt Þ ¼ x2 ðt Þ ð2:68Þ
dt
and
y2 ð τ Þ ¼ 0 ð2:69Þ
dy1 ðt τÞ
þ y1 ð t τ Þ ¼ x1 ð t τ Þ ¼ x2 ð t Þ
dt
88 2 Continuous-Time Signals and Systems
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.
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
To obtain complementary solution, let us assume
yc ðt Þ ¼ Bekt
Bkekt þ Bekt ¼ 0
ðk þ 1ÞBekt ¼ 0
2.5 Systems Described by Differential Equations 89
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,
dhðt Þ
þ hðt Þ ¼ δðt 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
L dyðtÞ
þ yðtÞ ¼ xðtÞ
R dt
which can be rewritten as
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
þ hðt Þ ¼ δðt Þ
dt L L
dhc ðtÞ R
þ hc ðtÞ ¼ 0
dt L
To obtain complementary solution, let us assume
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
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
þ hðt Þ ¼ δðt Þ
dt L L
2.5 Systems Described by Differential Equations 91
yields
R R R duðtÞ R R R
B eLt uðtÞ þ BeLt þ B eLt uðtÞ ¼ δðtÞ
L dt L L
Lt 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
Ðt ÐtR R
sðtÞ ¼ hðτÞdτ ¼ 0 eLτ dτ
0
L
Lτ t
R R
Lt
¼ e j0 ¼ 1 e uðtÞ
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
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
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) ' ]);
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
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))];
[y ty]=convint(x,tx,h,th);
figure,plot(th,h)
xlabel('Time')
ylabel('Amplitude');
figure, plot(ty,y);
xlabel('Time')
ylabel('Amplitude');
The impulse response and step response obtained from the above program are
shown in Figure 2.29(a) and (b), respectively.
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
94 2 Continuous-Time Signals and Systems
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
(a)
(b)
XN XN
n¼0
an yðNkÞ ðt Þ ¼ n¼0
bn xðNnÞ ðt Þ ð2:72Þ
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
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).
96 2 Continuous-Time Signals and Systems
(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
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
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
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
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
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 ðt 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
100 2 Continuous-Time Signals and Systems
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
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
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
– -
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 x 2 x3
dt
The voltage V c1 is taken as the output y(t):
yð t Þ ¼ x 1
1Ω
i1(t ) iL(t )
1H
+
+
vs (t ) 1F vc (t )
−
1Ω
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 ðt Þ ¼ i L ðt 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
This equation can be rewritten as
104 2 Continuous-Time Signals and Systems
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
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
This equation can be rewritten as
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 x 2 ð t Þ
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
2.9 Problems
x (t ) y (t )
(i)
(ii)
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
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.
t
-2 -1 0 1 2
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
16. Determine the impulse response y(t) of the OP-Amp circuit shown in
Figure P2.4.
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
1 1
0 5 0 0.5 1.5
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
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
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.
Hence,
ðT0
T0 m¼n
ejðnmÞΩ0 t dt ¼ ð3:8Þ
0 0 m 6¼ n
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
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
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
Letting τ ¼ t t0
114 3 Frequency Domain Analysis of Continuous-Time Signals and Systems
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
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
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 .
3.1 Complex Exponential Fourier Series Representation of the Continuous. . . 115
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
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
116 3 Frequency Domain Analysis of Continuous-Time Signals and Systems
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
X1
an jnΩ0 t
¼ e
n¼1
jnΩ0
Since Ω0 ¼ T2π0
ð X1
T0
xðt Þdt ¼ an ejnΩ0 t
n¼1
jn2π
3.1 Complex Exponential Fourier Series Representation of the Continuous. . . 117
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
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
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
X1 ðT0
1 jnT2π t
P¼ a∗
n¼1 n T
xðt Þe 0 dt ð3:20Þ
0 0
Thus,
3.1 Complex Exponential Fourier Series Representation of the Continuous. . . 119
ð X
1
1
P¼ jxðt Þj2 dt ¼ jan j2
T0 T0 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
120 3 Frequency Domain Analysis of Continuous-Time Signals and Systems
(
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
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
¼ xðt ÞejnΩ0 t dt
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.
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
: xðt ÞejnΩ0 t dt for odd n
T0 0
For odd n
ð
2 T 0 =2
an ¼ xðt ÞejnΩ0 t dt
T0 0
ð
2 3
¼ xðt ÞejnΩ0 t dt
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 Ð T 0 =2
: 2 xðt ÞejnΩ0 t dt for odd n
T0 0
For odd n
x(t)
2 4 6 8 t
-1
ð T 0 =2
2
an ¼ xðt ÞejnΩ0 t dt
T0 0
ð4
2
¼ xðt ÞejnΩ0 t dt
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
X
1
xð t Þ ¼ an ejnΩ0 t
n¼1
where
By Parseval’s theorem,
X
1
E¼ j an j 2
n¼1
pffiffiffiffiffiffiffiffiffiffiffiffiffiffiffi
2 pffiffiffiffiffiffiffiffiffiffiffiffiffiffiffi
2
¼ 2 2 þ 22 þ 32 þ 62 þ 3 2 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
xð t Þ ¼ an ejnΩ0 t
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.
124 3 Frequency Domain Analysis of Continuous-Time Signals and Systems
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
xð t Þ ¼ an ejnΩ0 t
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 ej ð 2 Þ
π
2j 2 2
an¼ 0 for all other n.
The magnitudes of Fourier coefficients are
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
3.1 Complex Exponential Fourier Series Representation of the Continuous. . . 125
(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
xð t Þ ¼ an ejnΩ0 t
n¼1
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.
3.1 Complex Exponential Fourier Series Representation of the Continuous. . . 127
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
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,
128 3 Frequency Domain Analysis of Continuous-Time Signals and Systems
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
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
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
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
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
132 3 Frequency Domain Analysis of Continuous-Time Signals and Systems
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 n2 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
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)
For n ¼ 0,
ð 1=4
1 1
a0 ¼ cos ð2πt Þdt ¼
1 1=4 π
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
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
134 3 Frequency Domain Analysis of Continuous-Time Signals and Systems
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
3.3 The Continuous Fourier Transform for Nonperiodic Signals 135
Eq. (3.27) is referred to as the Fourier transform of x(t), and Eq. (3.30) is called
the inverse Fourier transform.
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
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
jΩ0 t
0ejΩ0 t
(i) sin ðΩ0 tÞ ¼ e 2j
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
− 1 0
1
t
138 3 Frequency Domain Analysis of Continuous-Time Signals and Systems
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
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
3.3 The Continuous Fourier Transform for Nonperiodic Signals 139
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
XðjΩÞ ¼ pffiffiffi π
b
Substituting b ¼ 2σ1 2
XðjΩÞ ¼ pffiffiffi π
pffiffiffiffiffi σb2 Ω2 =2
¼ σ 2π e
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
X
1
xð t Þ ¼ δðt kT Þ
k¼1
140 3 Frequency Domain Analysis of Continuous-Time Signals and Systems
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
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 Re½X ðjΩÞ þ jIm X 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
Proof
Ð 1 ∗
X ∗ ðjΩÞ ¼ 1 xðt Þe
jΩt
dt
Ð1 ∗
¼ 1 x ðt Þe dt
jΩt
(ii) If x(t) is a continuous-time real valued and has odd symmetry, then
Proof Since x(t) is real x(t) ¼ x*(t) and x(t) has even symmetry x(t) ¼ x(t), we get
142 3 Frequency Domain Analysis of Continuous-Time Signals and Systems
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
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
Proof ð1
F½xðt t 0 Þ ¼ xðt t 0 ÞejΩt dt
1
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,
1 Ω
F½xðat Þ ¼ X j
j aj a
uðt Þ ¼ xe ðt Þ þ xo ðt Þ
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
(ii)
ejΩ0 t þ ejΩ0 t
cos ðΩ0 tÞuðtÞ ¼ uðtÞ
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
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 jðΩ Ω0 Þ 2 b þ jðΩ þ Ω0 Þ
b þ jΩ
¼ b>0
ðb þ jΩÞ2 þ Ω0 2
d
F½jtxðt Þ ¼ X ðjΩÞ ð3:41Þ
dΩ
ð1
X ðjΩÞ ¼ xðt ÞejΩt dt
1
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!
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
148 3 Frequency Domain Analysis of Continuous-Time Signals and Systems
" #
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
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
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 ð1
1 ∗ jΩt
E¼ xð t Þ X ðjΩÞe d Ω dt ð3:47Þ
1 2π 1
Thus,
ð1 ð1
2 1
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 < jΩj 2
:
0 otherwise
can be written as
3.3 The Continuous Fourier Transform for Nonperiodic Signals 151
where
1 jΩj 1
X 1 ðjΩÞ ¼
0 jΩj > 1
1 jΩj 2
X 2 ðjΩÞ ¼
0 jΩj > 2
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,
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
ð1 ð1
Y ðjΩÞ ¼ F½yðt Þ ¼ ½x1 ðτÞx2 ðt τÞ dτejΩt dt ð3:51Þ
1 1
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 ðt Þ ¼ x2 ðt Þ ¼
0 jt j > 1
yðt Þ ¼ x1 ðt Þ∗x2 ðt Þ
By definition,
-2 2 t
3.3 The Continuous Fourier Transform for Nonperiodic Signals 153
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
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
154 3 Frequency Domain Analysis of Continuous-Time Signals and Systems
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
3.3 The Continuous Fourier Transform for Nonperiodic Signals 155
Ω Ωt F 1 jt j Ω
xðt Þ ¼ sin c $ X ðjΩÞ ¼
π π 0 jt j > Ω
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
F½x2 ðt Þ ¼ X 2 ðjΩÞ ¼ F
½ sin 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,
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 jΩj Ω0
X ðjΩÞ ¼
0 j Ωj > Ω0
W
−W − W 0 −W −W + W 0 0 W −W0 W W + W0
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
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
Y ðjΩÞ
H ðjΩÞ ¼ ð3:57Þ
X ðjΩÞ
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
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 v 0 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
162 3 Frequency Domain Analysis of Continuous-Time Signals and Systems
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
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)
Ω
−Ω 0 Ω
Ω
−Ω Ω
0.5
Ω
−Ω − Ω −Ω −Ω Ω Ω − Ω Ω Ω Ω
c(t)
Figure 3.15 Amplitude demodulation
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
0.5
Ω
−Ω −Ω Ω Ω Ω Ω
Ω
Ω Ω
1
1 1
Ω Ω Ω
Ω
0.5
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
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
10. Plot the magnitude and phase spectrum of the periodic signal shown in
Figure P3.4
168 3 Frequency Domain Analysis of Continuous-Time Signals and Systems
16. Find Fourier transform of the following signal shown in Figure P3.5
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
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.
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.
170 3 Frequency Domain Analysis of Continuous-Time Signals and Systems
Further Reading
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.
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.
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 .
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
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.
The Laplace transform is a ratio of polynomials in the complex variable, which can
be represented by
N ð sÞ
X ð sÞ 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.
174 4 Laplace Transforms
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.
t
4.2 Properties of the Region of Convergence 175
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.
t
176 4 Laplace Transforms
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.
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
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
178 4 Laplace Transforms
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
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
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 ð sÞ
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
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
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
Then,
ð σþj1
dx 1
¼ sX ðsÞest ds ð4:30Þ
dt 2πj σj1
180 4 Laplace Transforms
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
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
d
X ð sÞ ¼ L fx1 ð t Þ g ð4:34Þ
ds
which can be rewritten as
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 ð sÞ ð4:37Þ
1 s
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
ð cþj1
1
Lfx1 ðt Þx2 ðt Þg ¼ X 1 ðpÞX 2 ðs pÞdp ð4:42Þ
2πj cj1
1
Lfxð t Þ g ¼ x2 ðt Þest X 1 ðpÞept dp dt ð4:45Þ
0 2πj cj1
where
ð1
X 2 ðs p Þ ¼ x2 ðt ÞeðspÞt dt ð4:47Þ
0
The above properties of the Laplace transform are summarized in Table 4.1.
Even Property
If x(t) is an even function such that x(t) ¼ x(–t), then X(s) ¼ X(s).
4.4 Properties of the Laplace Transform 183
Proof ð1
X ð sÞ ¼ xðt Þ est dt ð4:49Þ
1
Consider
ð1
X 1 ðsÞ ¼ xðt Þ est dt ð4:50Þ
1
Consider
ð1
X 1 ðsÞ ¼ xðt Þ est dt ð4:53Þ
1
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
184 4 Laplace Transforms
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
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
Solution
(a) X ðsÞ ¼ ðsþ2Ks
Þðs2Þ ; X ðsÞ ¼ ðsþ2Þðs2Þ ¼ X ðsÞ;
Ks
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 p 2 Þ ð s p 3 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 ð sÞ ¼
ð s p 1 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
186 4 Laplace Transforms
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
1
X ðsÞ ¼
s2 p1ffiffi2s þ 14 s2 þ p1ffiffi2s þ 14
For a signal x(t) with Laplace transform X(s) and x(t) ¼ 0 for t < 0, then
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
dx st
lim e ¼0 ð4:62Þ
s!1 dt
is valid. Hence, we get
For a signal x(t) with Laplace transform X(s) and x(t) ¼ 0 for t < 0, then
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
Therefore,
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
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 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!
1
X 1 ðsÞ ¼ Lfuðt Þg ¼
s
and for n ¼ 2, x2(t) ¼ tu(t).
190 4 Laplace Transforms
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
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
4.5 Laplace Transforms of Elementary Functions 191
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
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
192 4 Laplace Transforms
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
and its pole-zero plot and ROC are as shown in Figure 4.6.
4.5 Laplace Transforms of Elementary Functions 193
Re
1 1.5 2
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
yð t Þ ¼ hðτÞxðt τÞdτ
1
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
for an input
xðt Þ ¼ 2 þ 4e3t uðt Þ
2 4 6ð s þ 1Þ
X ð sÞ ¼ þ ¼
s s þ 3 s ð s þ 3Þ
2 3 1 6
Y ðsÞ ¼ þ ¼
s s þ 1 s þ 3 sðs þ 1Þðs þ 3Þ
Y ð sÞ 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
Y 1 ðsÞ 1
X 1 ðsÞ ¼ ¼ ℜeðsÞ > 0
H ðsÞ s
x1 ðt Þ ¼ uðt Þ:
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
N ð sÞ
X ð sÞ 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.
Consider a rational Laplace transform X(s) with repeated poles of the form
N ð sÞ
X ð sÞ ¼ ð4:76Þ
ðs p1 Þr ðs 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
196 4 Laplace Transforms
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
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
1 k1 k2
¼ þ
sðs 1Þ s s1
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
4.6 Computation of Inverse Laplace Transform Using Partial Fraction Expansion 197
Thus,
s2 s þ 1
X ð sÞ 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
Using Table 4.2, we obtain
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
198 4 Laplace Transforms
k 11 ¼ 3; k12 ¼ 3
Hence,
s2 s þ 1 3 3
X ð sÞ ¼ ¼1 þ
ð s þ 1Þ 2 s þ 1 ð s þ 1Þ 2
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
s2 þ s þ 1
X ð sÞ 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
4.6 Computation of Inverse Laplace Transform Using Partial Fraction Expansion 199
Example 4.18 Determine x(t) for the following conditions if X(s) is given by
1
X ðsÞ ¼
ð s þ 2 Þ ð s þ 3Þ
1 1 1
X ðsÞ ¼ ¼
ð s þ 2 Þ ð s þ 3Þ ð s þ 2 Þ ð s þ 3Þ
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
ð s þ 2Þ ð s þ 3Þ
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)
200 4 Laplace Transforms
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 3 Þ 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
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).
sþ1
X ð sÞ 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
202 4 Laplace Transforms
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)
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
yð t Þ ¼ hð t Þ ∗ xð t Þ ð4:81Þ
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
Y ðs Þ
H ðsÞ ¼ ð4:83Þ
X ð sÞ
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
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 r2 sc2 ÞV 0
Substituting the above Eq. for V 1 , the input-output relation can be written as
r1 r1
Vi ¼ 1 þ þ r 1 sc1 ð1 þ r 2 sc2 Þ r 1 sc1 V 0
r2 r2
204 4 Laplace Transforms
Thus,
V 0 ðsÞ 1
¼
V i ðsÞ r1 r1
1 þ þ r 1 sc1 ð1 þ r2 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
1
r 1 r 2 c1 c2
H ð sÞ ¼
r 1 þ r2 1
s2 þ sþ
r 1 r 2 c1 r 1 r 2 c1 c2
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
4.8 Analysis of Continuous-Time LTI Systems Using the Laplace Transform 205
Im
s plane
Re
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.
206 4 Laplace Transforms
Im
s plane
Re
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 2 Þ 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.
4.8 Analysis of Continuous-Time LTI Systems Using the Laplace Transform 207
Im
s-plane
Re
-2 1 3
Im
s-plane
Re
-2 1 3
Figure 4.12 Pole-zero plot and ROC of the noncausal, stable system
s-plane Im
Re
-2 1 3
Figure 4.13 Pole-zero plot and ROC of the system neither causal nor stable
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
Example 4.23 Consider a continuous LTI system described by the following dif-
2
ferential equation ddt2y dy
dt 6y ¼ x.
Y ðs Þ 1
H ðsÞ ¼ ¼
X ð sÞ s 2 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
5 ð s 3 Þ 5 ð s þ 2Þ
Im
Re
-2 3
(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
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
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 2 Þ
4.8 Analysis of Continuous-Time LTI Systems Using the Laplace Transform 211
k1 k2
YðsÞ ¼ þ
ð s 3Þ ð s 2 Þ
k 1 ð s 2Þ þ k 2 ð s 3Þ
¼
ð s 3Þ ð s 2 Þ
k1 þ k2 ¼ 2
2k1 3k2 ¼ 9
k1 ¼ 3, k2 ¼ 5:
Thus,
3 5
YðsÞ ¼ þ
ð s 3Þ ð s 2 Þ
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
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 1 Þ 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
2 2
YðsÞ ¼ þ
ð s þ 1Þ ð s þ 1Þ 3
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)
214 4 Laplace Transforms
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
k11 k12 k2
Vc ðsÞ ¼ þ þ
ð s þ 1 Þ ð s þ 1Þ 2 ð s þ 4Þ
Thus,
14 4 5
Vc ðsÞ ¼ þ
9 ð s þ 1 Þ 3 ð s þ 1Þ 2 9 ð s þ 4 Þ
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).
Solution (a) The Is and IL are related by the following differential equation
dIL
þ IL ¼ IS
dt
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
1 1
IL ðsÞ ¼
2 ð s þ 1Þ 2 ð s þ 3Þ
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
Thus,
1
IL ðsÞ ¼
sþ1
216 4 Laplace Transforms
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
Example 4.29 Find the solution of linear differential equation considered in Exam-
ple 4.25 using MATLAB
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
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 ð t Þ ¼ h1 ð t Þ þ h 2 ð t Þ ð4:93Þ
and from linearity property of the Laplace transform, the associated system function
is (Figure 4.18)
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
(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
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.
220 4 Laplace Transforms
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
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
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 τ Þ
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
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
4.11 Problems
1
XðsÞ ¼
s2 þ1
4.11 Problems 223
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
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.
12. Determine the differential equation characterizing the system represented by the
following block diagram
Y(s)
-5 6
-7 -12
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
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 impulse response of the ideal low-pass filter corresponds to the inverse
Fourier transform of the frequency response shown in Figure 5.1.
Hence,
ð Ωc
1 sin Ωc t
hð t Þ ¼ ejΩc t dΩ ¼ ð5:2Þ
2π Ωc πt
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 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, jΩj Ωc
H HP ðΩÞ ¼ ð5:6Þ
1, j Ω j > Ωc
and is shown in Figure 5.4. The frequency Ωc is called the cutoff frequency
5.1 Ideal Analog Filters 229
From Figure 5.3, the frequency response of the ideal high-pass filter can also be
expressed as
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
Ωc2 Ωc2 t Ωc1 Ωc1 t
hBP ðt Þ ¼ sinc sinc ð5:11Þ
π π π π
Band-Stop Filter
From Figure 5.6, the frequency response of the ideal band-stop filter can also be
expressed as
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
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.
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-
1
jH a ðjΩÞj2 ¼ ð5:15Þ
1 þ ðΩ=Ωc Þ2N
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
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
1 j 1 j
s2 ¼ pffiffiffi þ pffiffiffi ; s3 ¼ pffiffiffi pffiffiffi
2 2 2 2
Hence,
5.2 Practical Analog Low-Pass Filter Design 235
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
1
H N ðsÞ ¼
ð s þ 1 Þ 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
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
236 5 Analog Filters
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,
5.2 Practical Analog Low-Pass Filter Design 237
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
1
H N ðsÞ ¼
ðs2 þ 0:76537s þ 1Þðs2 þ 1:8477s þ 1Þ
Ωs 10000π
Ωc ¼
1=2N ¼
1=8 ¼ 9935
10 1
4
104 1
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
1
jH ðΩÞj2 ¼
ð5:28Þ
1 þ ε2 T 2N Ω=Ωp
(
cos ðN cos 1 ΩÞ, j Ωj 1
T N ð ΩÞ ¼
ð5:29Þ
cosh Ncosh1 Ω , j Ωj > 1
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
H0
H N ðsÞ ¼ ð5:38Þ
Π k ðs p k Þ
where
5.2 Practical Analog Low-Pass Filter Design 239
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
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
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
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
5.2 Practical Analog Low-Pass Filter Design 241
Figure 5.10 Magnitude response of typical Type 1 Chebyshev low-pass filter with 1 dB passband
ripple
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
242 5 Analog Filters
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
1 pffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffi pffiffiffiffiffiffiffiffiffiffiffiffiffiffiffi
¼ 100:1αs 1 ¼ 104 1 ¼ 99:995
δs
Hence,
1
sinh1 ¼ 5:28829
δs
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
H 0 ¼ 0:03
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
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.
244 5 Analog Filters
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
Figure 5.11 Magnitude response of typical Type 2 Chebyshev low-pass filter with 20 dB stopband
ripple
0:03ðs2 þ 1:3333Þ
H N ðsÞ ¼
ðs 3 þ 0:6746s2 þ 0:2271s þ 0:04Þ
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 10 0:1αp
1
Nffi ð5:45Þ
log10 ð1=ρÞ
pffiffiffiffiffiffiffiffiffiffiffiffiffi
k0 ¼ 1 k2 ð5:47Þ
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
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
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:
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:
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
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
5.2 Practical Analog Low-Pass Filter Design 249
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.
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);
250 5 Analog Filters
[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);
ph2=angle(h2);ph2=unwrap(ph2);
ph3=angle(h3);ph3=unwrap(ph3);
ph4=angle(h4);ph4=unwrap(ph4);
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.
5.2 Practical Analog Low-Pass Filter Design 251
Figure 5.15 A comparison of various types of analog low-pass filters: (a) magnitude response and
(b) phase response
252 5 Analog 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
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
(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
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
254 5 Analog Filters
(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
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
5.2 Practical Analog Low-Pass Filter Design 255
(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
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
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
1
H N ðsÞ ¼
ð s þ 1 Þ ð s 2 þ s þ 1Þ
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
5.2 Practical Analog Low-Pass Filter Design 257
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
b p2 Ω
Bp ¼ Ω b p1 ð5:54aÞ
qffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffi
b mp ¼ Ω
Ω b p1 Ω b p2 ð5:54bÞ
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
Ω b p1 Ω
b p2 Ω b s2
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
Passband ripple: 2 dB
Stopband ripple: 10 dB
Solution We have
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
5.2 Practical Analog Low-Pass Filter Design 259
Substituting these values in Eq. (5.23), the order of the filter is given by
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
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-
ple 5.7 using MATLAB
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’);
[B,A]=butter(N,Wn,‘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
260 5 Analog Filters
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
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
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
b s2 Ω
Ω b s1 Ωb p1 1
Ω0p ¼ Ωs ¼ Ωs ð5:62aÞ
b s1 Ω
Ω b s2 Ωb2 A 1
p1
and
b s2 Ω
Ω b s1 Ωb p2 1
Ω00p ¼ Ωs ¼ Ωs ð5:62bÞ
b s1 Ω
Ω b s2 Ωb 2 A2
p2
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
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
Passband ripple: 3 dB
Stopband ripple: 15 dB
262 5 Analog Filters
(a)
(b)
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
Substituting these values in Eq. (5.23), the order of the filter is given by
101:5 1
log 100:3 1
N ¼ 0:9125
2 logð6:5397Þ
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-
ple 5.9 using MATLAB
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’);
[B,A]=butter(N,Wn,‘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
264 5 Analog Filters
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
Frequency response of a system can be obtained by evaluating H(s) for all values of
s ¼ jΩ.
1
H ðsÞ ¼
ð5:64Þ
s ðα þ jΩÞ s ðα jΩÞ
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)
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)
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.
266 5 Analog Filters
(a)
(b) (c)
Figure 5.21 (a) Zero locations of H(s). (b) Magnitude response. (c) Phase response
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)
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
268 5 Analog Filters
(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
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
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)
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
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
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
274 6 Discrete-Time Signals and Systems
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
fH
F T ¼ 2ðΔf Þ ¼ ð6:8Þ
c
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
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,
x kþ1 b
b xk ¼ Δ k ¼ 1, 2, , L:
xkþ1 xk ¼ Δ for finite xk , xkþ1 : ð6:10Þ
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
quantization step2 A2 1 A2
σ 2e ¼ ¼ 2n ¼ 22n ð6:13Þ
12 12 2 12
276 6 Discrete-Time Signals and Systems
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.
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
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.
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
278 6 Discrete-Time Signals and Systems
· · ·
· · ·
··· ···
· · ·
· · · · · · · · ·
-5 -4 -3 -2 -1 0 1 2 -3 4 5 6 7 8 -9 10 11 12
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
5π
mental period N ¼ 16.
6.2 Classification of Discrete-Time Signals 279
X
1
E¼ jxðnÞj2 ð6:17Þ
n¼1
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
1 X1 XN
P ¼ limN!1 xðnÞ2 ¼ limN!1 1 a2n
2N þ 1 1 2N þ 1 0
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
280 6 Discrete-Time Signals and Systems
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
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
The energy E is infinite and also the average power P is infinite. Hence, the signal
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
The energy E is infinite and also the average power P is infinite. Hence, the signal
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.
where ℜ is an operator.
282 6 Discrete-Time Signals and Systems
Linear Systems
A system is said to be linear if and only if it satisfies the following conditions:
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
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
(ii) The outputs y1(n) and y2(n) for inputs x1(n) and x2(n) are given by
The output y(n) due to an input x(n) ¼ ax1(n) þ bx2(n) is then given by
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
y 1 ð n Þ ¼ x 1 ð n n0 Þ
y 2 ð n Þ ¼ x 2 ð n n0 Þ
The output y(n) due to an input x(n) ¼ ax1(n) þ bx2(n) is then given by
ℜ½xðn n0 Þ ¼ yðn n0 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
y1 ð nÞ ¼ x ð k n0 Þ
k¼1
284 6 Discrete-Time Signals and Systems
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
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
is not causal, as the output at time instant n depends on future values of the input.
But, the system defined by
y ð n Þ ¼ x ð n Þ 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
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
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, ℜ½ax1 ðnÞ þ bx2 ðnÞ ¼ ð1Þn ax1 ðnÞ þ ð1Þn bx2 ðnÞ ¼ ay1 ðnÞ þ by2 ðnÞ
Hence, it is linear.
Therefore, it is linear.
yðnÞ ¼ ℜ½xðnÞ ¼ xðn2 Þ
ℜ½xðn 1Þ 6¼ yðn 1Þ
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 ℜ ½ 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.
Let the response of the system due to input δ(n k) be hk(n), that is,
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
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
288 6 Discrete-Time Signals and Systems
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
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.
6.4 Linear Time-Invariant Discrete-Time Systems 289
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 0 7 6 7 6 7
6 7 6 6 0 76 6 7 6 45 7
6 7¼6 76 7¼6 7
6 yð2Þ 7 6 0 3 0 7 6 7 6 7
4 5 4 6 54 3 5 4 0 5
yð3Þ 0 0 3 6 0 9
Thus,
6 h (n ) 6
x ( n)
3
0
n n
0 1 1 2
-3
-3
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
Starting with the convolution sum given by (6.30), namely, y(n) ¼ x(n) * h(n), we
can establish the following properties:
292 6 Discrete-Time Signals and Systems
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
y1 ( n) y (n )
x( n)
x( n) y(n)
h1 (n) h2( n) ≡ h(n) = h1(n)* h2 (n)
y1 ( n)
h1( n)
y(n) x( n) y(n)
x( n)
≡ h1(n) + h2 (n)
h2 ( n)
y2 ( n)
y(- n)
y1 ( n)
h (n)
x (n) y2 ( n)
h1 (n)
x ( n) y ( n)
h1 (n ) h2 (n ) h2 (n )
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
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,
X
1
h3 ð n Þ 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 ð1Þ ¼ h2 ðkÞh2 ð1 k Þ ¼ 1:1 þ 1:1 ¼ 2
k¼0
X
1
h3 ð2Þ 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:
X
4
hðnÞ ¼ h1 ðnÞ∗h3 ðnÞ ¼ h1 ðkÞh3 ðn k Þ
k¼0
X
4
h ð 0Þ ¼ h1 ðkÞh3 ðk Þ ¼ a1 1 ¼ 1
k¼0
) a1 ¼ 1:
X
4
h ð 1Þ ¼ h1 ðkÞh3 ð1 k Þ ¼ a1 1 þ a2 2 ¼ 5
k¼0
) a2 ¼ 3
X
4
h ð 2Þ ¼ h1 ðkÞh3 ð2 k Þ ¼ a1 1 þ a2 2 þ a3 1 ¼ 101
k¼0
) a3 ¼ 3:
X
4
h ð 3Þ ¼ h1 ðkÞh3 ð3 k Þ ¼ a2 1 þ a3 2 þ a4 1 ¼ 11
k¼0
) a4 ¼ 2:
X
4
h ð 4Þ ¼ h1 ðkÞh3 ð4 k Þ ¼ a3 1 þ a4 2 þ a5 1 ¼ 8
k¼0
) a5 ¼ 1:
Thus,
h1 ðnÞ ¼ f1; 3; 3; 2; 1g
j x ð nÞ j β x < 1
we have
296 6 Discrete-Time Signals and Systems
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
X
1
y ð n0 Þ ¼ hðkÞxðn0 kÞ
k¼1
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
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.
X
1
y ð nÞ ¼ bm xð 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
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
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
For an input x(n) ¼ u(n)u(n1), the response of an LTI system is the impulse
response h(n) given by
X
4
y ð nÞ ¼ hðmÞxðn mÞ
m¼0
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
where yc(n) is called the complementary solution and yp(n) is called the particular
solution.
300 6 Discrete-Time Signals and Systems
X
N
ak yð n k Þ ¼ 0 ð6:43bÞ
k¼0
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
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
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 xð n k Þ ð6:48Þ
6.5 Characterization of Discrete-Time Systems 301
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,
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
¼ λn2 ðλ 1Þðλ þ 3Þ ¼ 0
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
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
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
6.5 Characterization of Discrete-Time Systems 303
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
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
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
7 1
y ð nÞ ¼ 1n
þ 2n þ ð1Þn uðnÞ ð6:58Þ
2 2
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:
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
x ( n) xd (n) = x(nM )
M
Sampling period T Sampling period T ' = MT
x ( n) xe (n) = x(n / L)
L
Sampling period T Sampling period T ' = TL
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.
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
X
1 n
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).
where y(n) is the system output and ℧(n) is the system input.
Define the following useful set of state variables:
x1 ð n þ 1Þ ¼ x 2 ð n Þ
x 2 ð n þ 1Þ ¼ x 3 ð n Þ
⋮
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 n Þ ¼ x N ð n þ 1Þ ð6:63bÞ
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 x2 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
xN ð n Þ 1
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
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 n Þ ¼ x 3 ð n þ 1Þ
where
310 6 Discrete-Time Signals and Systems
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
< 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
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:
Determine the response of the system for x(n) ¼ nu(n) and initial conditions
y(-1) ¼ 1 and y(-2) ¼ 0.
312 6 Discrete-Time Signals and Systems
13. Determine the response of the system described by the following difference
equation:
1. Using the function impz, write a MATLAB program to determine the impulse
response of a discrete-time system represented by
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
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
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
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).
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
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 ðnÞ 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
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
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.
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
318 7 Frequency Domain Analysis of Discrete-Time Signals and Systems
F½a1 x1 ðnÞ þ a2 x2 ðnÞ ¼ a1 X 1 ejω þ a2 X 2 ejω ð7:13Þ
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
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
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
Thus,
X1 ðπ
1 jω 2
E¼ jxðnÞj ¼ 2 X e dω ð7:23Þ
1 2π π
320 7 Frequency Domain Analysis of Discrete-Time Signals and Systems
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
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
X1 X1 ∗
F ½ x ∗ ð nÞ ¼ n¼1
x∗ ðnÞejωn ¼ n¼1
xðnÞejωn ¼ X ∗ ejω ð7:24bÞ
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
A complex sequence x(n)can be decomposed into a sum of its real and imaginary
parts as
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.
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
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
7.2 Representation of Discrete-Time Signals and Systems in Frequency Domain 323
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
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
324 7 Frequency Domain Analysis of Discrete-Time Signals and Systems
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:
(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
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
7.2 Representation of Discrete-Time Signals and Systems in Frequency Domain 325
(ii) Now,
X1 X1
n¼1
j hð nÞ j ¼ n¼1
jajn1 < 1 for jaj < 1:
5 1 1
y ð n Þ y ð n 1Þ þ y ð n 2Þ ¼ x ð n 1Þ
6 6 3
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
X 1 X
1 n 1
X 1 ejω ¼ ðaÞn ejωn ¼ aejω ¼
n¼0 n¼0
1 aejω
For m ¼ 2,
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
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
g 2 ð n Þ ¼ g 1 ð n Þ þ g1 ð n 4Þ
4 g1 (n) g2 (n)
3
2
1
0 1 2 3 n 0 1 2 3 4 5 6 7 n
(a) (b)
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
Hence,
1
F1 ½δðω þ 2πkÞ ¼ , ð1 < n < 1Þ
2π
jω
αe
(b) X 2 ðejω Þ ¼ ð1αe jω Þ2
, jαj < 1
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
328 7 Frequency Domain Analysis of Discrete-Time Signals and Systems
· 4
· 3
1 · 1
0 1 5
· ·
-3 -2 -1 2 3 4
· -1
-2 ·
· -3
Then
α
$ ðn þ 1Þαnþ1 uðnÞ
ð1 αejω Þ2
αejω
$ nαn uðn 1Þ
ð1 αejω Þ2
X
1
X ðejω Þ ¼ xðnÞejωn
n¼1
X
1
X ðej0 Þ ¼ x ð nÞ
n¼1
¼ ½3 þ 0 þ 1 2 3 þ 4 þ 1 þ 0 1 ¼ 3
7.2 Representation of Discrete-Time Signals and Systems in Frequency Domain 329
X
1
X ðejπ Þ ¼ xðnÞejπ
n¼1
X
1
X ðejπ Þ ¼ xðnÞ ¼ 3
n¼1
ðπ
(c) X ejω dω
π
Hence,
ðπ
X ejω ejωn dω ¼ 2πxð0Þ ¼ 4π
π
ðπ
jω 2
(d) X e dω
π
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
( )
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
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 the impulse response of the parallel system is shown in
Figure 7.5(b).
332 7 Frequency Domain Analysis of Discrete-Time Signals and 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
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
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
334 7 Frequency Domain Analysis of Discrete-Time Signals and Systems
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 rffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffi
ffi
ð1 0:5 cos ωÞ2 þ ð0:5Þ2 sin 2 ω 1 þ ð0:5Þ2 2ð0:5Þ cos ω
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.
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
7.3 Frequency Response of Discrete-Time Systems 335
(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
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.
336 7 Frequency Domain Analysis of Discrete-Time Signals and Systems
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)
Solution Given
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
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.
7.3 Frequency Response of Discrete-Time Systems 339
Figure 7.9 (a) Magnitude response of h(n) sequence. (b) Phase response of h(n) sequence
340 7 Frequency Domain Analysis of Discrete-Time Signals and Systems
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:
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’);
7.3 Frequency Response of Discrete-Time Systems 341
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:
342 7 Frequency Domain Analysis of Discrete-Time Signals and Systems
(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)
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
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 ðt 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
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
Since
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
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.
346 7 Frequency Domain Analysis of Discrete-Time Signals and Systems
(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
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
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.
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 ðt 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:
348 7 Frequency Domain Analysis of Discrete-Time Signals and Systems
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).
Figure 7.16
1
-W -10000 10000 W
(a)
jWT
1 X (e )
T
·
- -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
350 7 Frequency Domain Analysis of Discrete-Time Signals and Systems
4
4
3 3
2 2
…….
1
1
0 1 2 3 4 5 6 7 8 9 n
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
- 5000p 5000p W
(a)
T1 = 0.0001sec T2 = 0.0001sec
(b)
Figure P7.3 (a) Spectrum of signal. (b) Signal reconstruction
Further Reading
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.
or
x ð n þ 1Þ
jzj > limn!1 ¼ r 1 ðsayÞ ð8:3aÞ
x ð nÞ
or
x ð n þ 1Þ
jzj < limn!1 ¼ r 2 ðsayÞ ð8:3bÞ
x ð nÞ
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:
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
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 ð zÞ ¼ xðnÞzn
n¼N
ð8:9Þ
¼ xðN ÞzN þ þ xð2Þz2 þ xð1Þz
Re(z)
356 8 The z-Transform and Analysis of Discrete Time LTI Systems
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 ð zÞ 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
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.
8.2 Properties of the Region of Convergence for the z-Transform 357
r1 Re
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 ð zÞ 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.
358 8 The z-Transform and Analysis of Discrete Time LTI Systems
r2 Re
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.
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
r2n 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 r1 Þ ðz r 2 Þ
8.2 Properties of the Region of Convergence for the z-Transform 359
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
X
1 X
1 X
1 n
X ð zÞ 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 ð zÞ 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
360 8 The z-Transform and Analysis of Discrete Time LTI Systems
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
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
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
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
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 f xð n k Þ g ¼ n¼1
xðn kÞzn ð8:28Þ
Substituting m ¼ n k,
P1 ð m þ k Þ
P1
Z ½ xðn k Þ ¼ m¼1 xðmÞz ¼ zk m¼1 xðmÞzm
P ð8:29Þ
¼ zk 1m¼1 xðmÞz
m
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 f an x ð n Þ 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,
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
362 8 The z-Transform and Analysis of Discrete Time LTI Systems
dX ðzÞ X1
¼ ðnÞxðnÞ zn1 ð8:34Þ
dz n¼1
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
ra 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
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
Hence,
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
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).
364 8 The z-Transform and Analysis of Discrete Time LTI Systems
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
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
whose ROC is the same as that of X(z).
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
1
Z ½ImfxðnÞg ¼ ½X ðzÞ X ∗ ðz∗ Þ ð8:54Þ
2j
whose ROC is the same as that of X(z).
Proof Now
Thus,
1
ImfxðnÞg ¼ f x 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.
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
8.4 z-Transforms of Some Commonly Used Sequences 367
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
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
Solution X1 X1
X ðzÞ ¼ n¼1
xðnÞzn ¼ n¼0
n2 uðnÞzn
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
z sin ω
Z f sin ωn uðnÞ ¼ for jzj > 1
z2 2z cos 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
where
X1
H ðzÞ ¼ Z fAxðL nÞg ¼ A n¼1
xððn LÞÞzn
P
¼ AzL 1 m¼1 xðmÞz
m
Hence,
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
X
1
X ½z ¼ xðnÞ:zn ¼ xð0Þ þ xð1Þz1 þ xð2Þz2 þ ð8:67Þ
n¼0
0:5z2
X ð zÞ ¼
ðz 1Þðz2 0:85z þ 0:35Þ
0:5z2 0:5z2
xð0Þ ¼ lim X ðzÞ ¼ lim ¼ lim ¼0
z!1 n!1 ðz 1 Þ ðz 2 0:85z þ 0:35Þ z!1 zðz2 Þ
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
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.
372 8 The z-Transform and Analysis of Discrete Time LTI Systems
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
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∗
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
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
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
1 d m1
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
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
x ð 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
1 n1
x ð nÞ ¼ : 0:8 ð0:4Þn1 uðn 1Þ
2:4
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
376 8 The z-Transform and Analysis of Discrete Time LTI Systems
the numerator polynomial N(z) by its denominator polynomial D(z) and expressing X
(z) in the form
X
MN
N 1 ðzÞ
X ð zÞ 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
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 ð zÞ X N
ci
¼ ð8:89Þ
z i¼1
z pi
where
X ðzÞ
c i ¼ ð z pi Þ ð8:90Þ
z z¼v
Hence,
8.6 Methods for Computation of the Inverse z-Transform 377
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 a Þ 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 ð zÞ a 1 b 1
¼
z a bz a a bz b
Hence,
a 1 b 1
X ð zÞ 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 uðnÞ 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 ð zÞ 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
378 8 The z-Transform and Analysis of Discrete Time LTI Systems
Each term in the above expansion is a first-order z-transform and can be recog-
nized easily to evaluate the inverse transform as
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 ð zÞ ¼
ð z 1Þ 3
X ð zÞ zþ1 A B C
¼ ¼ þ þ
z ðz 1Þ3 z 1 ðz 1Þ2 ðz 1Þ3
z 2z
X ð zÞ 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
1 þ 2z1 þ z3
X ð zÞ 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
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
z3 þ 2z2 þ 1 2 8z 13z
X ð zÞ ¼ ¼6þ þ
zðz 1Þðz 0:5Þ z ðz 1Þ ðz 0:5Þ
8.6 Methods for Computation of the Inverse z-Transform 379
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
ð z 1Þ 2
H ðzÞ ¼
ðz2 0:1z 0:56Þ
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
0:0333z 2:7524z
H ðzÞ ¼ 1:7857 þ þ
ðz 0:8Þ ðz þ 0:7Þ
Hence,
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);
380 8 The z-Transform and Analysis of Discrete Time LTI Systems
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
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
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
implying that
u2 u3 u4 u5
logð1 þ uÞ ¼ u þ þ
2 3 4 5
X1
ð1Þnþ1 un
¼ , j uj < 1
n¼1
n
X 1
ð1Þnþ1 bn zn
X ðzÞ ¼ log 1 þ bz1 ¼ , jbj < jzj
n¼1
n
X
1
X ð zÞ 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
z 1
X ð zÞ 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
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:
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
8.6 Methods for Computation of the Inverse z-Transform 383
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
Example 8.26 Determine the impulse response of the system described by the
difference equation:
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
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
Example 8.27 Determine the response y(n), n 0 of the system described by the
second-order difference equation
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 ð zÞ 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 ð zÞ ¼ þ þ
25ðz þ 1Þ 25ðz 4Þ 5ðz 4Þ2
1 6 26
y ð nÞ ¼ ð1Þn uðnÞ þ nð4Þn uðnÞ þ ð4Þn uðnÞ
25 5 25
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
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
Taking z-transform on both sides of the above equation and using the convolution
property, we get
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
386 8 The z-Transform and Analysis of Discrete Time LTI Systems
z-domain, respectively. The transform H(z) is called the transfer function or the
system function and expressed as
Y ðz Þ
H ðzÞ ¼ ð8:98aÞ
X ð zÞ
Or equivalently,
PM
bk zk
H ðzÞ ¼ P
k¼0
N
ð8:98bÞ
1þ k¼1 ak zk
Y ðz Þ
H ðzÞ ¼
X ð zÞ
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
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.
8.7 Analysis of Discrete-Time LTI Systems in the z-Transform Domain 387
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
numerator = [0 1 -1];
denominator = [8 -6 1];
The following MATLAB statement yields the poles and zeros and gain of the
system:
The MATLAB command z-plane (z, p) plots the poles and zeros as shown in
Figure 8.6.
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
8.7 Analysis of Discrete-Time LTI Systems in the z-Transform Domain 389
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.
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 ðzÞ j 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.
390 8 The z-Transform and Analysis of Discrete Time LTI Systems
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.
392 8 The z-Transform and Analysis of Discrete Time LTI Systems
(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
Example 8.33 The rotational motion of a satellite was described by the difference
equation
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.
8.7 Analysis of Discrete-Time LTI Systems in the z-Transform Domain 393
´
0.5
0.5
-0.5
´
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 ðz Þ 1
¼
X ð zÞ 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.
394 8 The z-Transform and Analysis of Discrete Time LTI Systems
(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
(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
Hence,
Y ðzÞ z
H ðzÞ ¼ ¼
X ð zÞ 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
xðnÞ ¼ 10 for n 0,
10z
X ð zÞ ¼
z1
Thus,
z 10
Y ðzÞ ¼ H ðzÞX ðzÞ ¼
zþ 1z 1
or
8.7 Analysis of Discrete-Time LTI Systems in the z-Transform Domain 395
Y ðzÞ 5 5
¼ þ
z zþ1 z1
Therefore,
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 ð zÞ z þ 0:5
Y ðzÞ ð0:4z þ 1Þ ðz þ 0:4Þ
H 3 ðzÞ ¼ ¼ ð8:109Þ
X ð zÞ 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.
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
396 8 The z-Transform and Analysis of Discrete Time LTI Systems
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
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.
8.7 Analysis of Discrete-Time LTI Systems in the z-Transform Domain 397
Solution
Y ð zÞ
Y ðzÞ ¼ X ðzÞ e8α z8 X ðzÞ; ¼ 1 e8α z8
X ðzÞ
1 X ð zÞ
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.
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
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
Thus,
Y ejω ¼ H ejω X ejω : ð8:120Þ
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
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
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
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:
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,
is an all-pass system.
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
Proof By definition,
X1
Z þ ½xðn kÞ ¼ n¼0
xðn kÞ zn
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
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
Thus,
½xð1Þ xð0Þ þ ½xð2Þ xð1Þ þ ½xð3Þ xð2Þ þ ¼ limz!1 ðz 1ÞX ðzÞ xð0Þ
Thus,
or
Hence,
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 ð zÞ ¼
ðz 1 Þ ðz 2 0:85z þ 0:35Þ
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.
404 8 The z-Transform and Analysis of Discrete Time LTI Systems
Kz2
X ð zÞ 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 ð zÞ ¼
z2 12 z þ 14
Since x[n] is right-sided, its ROC is jzj > 12 (note that both poles have
magnitude 12).
þ 1 1 þ
Y ðzÞ ½z Y ðzÞ þ yð1Þ ¼ X þ ðzÞ
2
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
where
2 3
X 1 ðzÞ
6 X 2 ðzÞ 7
6 7
X ð zÞ ¼ 6
6 ⋮ 7
7
4 X N1 ðzÞ 5
X N ðzÞ
h i
z1 ½zI A1 zX ð0Þ ¼ An X ð0Þ ð8:141Þ
Thus,
Xn1
z1 ½X ðzÞ ¼ X ðnÞ ¼ An X ð0Þ þ k¼0
An1k b℧ðkÞ n > 0: ð8:143Þ
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 x1 ð nÞ 0
¼4 1 4 5 þ ℧ðnÞ
x 2 ð n þ 1Þ x2 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
408 8 The z-Transform and Analysis of Discrete Time LTI Systems
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
XN Ak
H a ðsÞ ¼ ð8:145Þ
k¼1 s pk
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
410 8 The z-Transform and Analysis of Discrete Time LTI Systems
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
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.
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
2 ð 1 þ sÞ
Z¼ ð8:157Þ
T ð 1 sÞ
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
412 8 The z-Transform and Analysis of Discrete Time LTI Systems
-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
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
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:
8.11 Problems
n 2 1 0 1 2 3 4
x(n) 1 2 3 4 5 6 7
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 u ½ 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
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
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.
416 8 The z-Transform and Analysis of Discrete Time LTI Systems
y ð n Þ y ð n 1Þ y ð n 2Þ ¼ x ð n 1Þ
1
H ðsÞ ¼
ð s þ 1 Þ ð s 2 þ s þ 1Þ
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
Passband ripple: 2 dB
Stopband ripple: 20 dB
Sampling frequency: 8000 Hz
23. Consider an initially relaxed discrete time with the following state-space repre-
sentation. Find y(n).
" # 2 3
x1 ð n þ 1 Þ 0 1 " x ð nÞ # " 0 #
1
¼ 4 1 35 þ ℧ðnÞ
x 2 ð n þ 1Þ x2 ð nÞ 1
8 4
" #
1 3 x 1 ð nÞ
y ð nÞ þ ℧ðnÞ
8 4 x2 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:
4. Write a MATLAB program using the command freqz to obtain magnitude and
phase responses of the following system:
Further Reading
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
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
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