SYSTEMS
MARTIN SCHETZEN
Northeastern University
IEEE
IEEE PRESS
www.ebook3000.com
Copyright 0 2003 by The Institute of Electrical and Electronics Engineers. All rights reserved.
Published simultaneously in Canada.
No part of this publication may be reproduced, stored in a retrieval system, or transmitted in any
form or by any means, electronic, mechanical, photocopying, recording, scanning, or otherwise,
except as permitted under Section 107 or 108 of the 1976 United States Copyright Act, without
either the prior written permission of the Publisher, or authorization through payment of the
appropriate percopy fee to the Copyright Clearance Center, Inc., 222 Rosewood Drive, Danvers,
MA 01923, 9787508400, fax 9787504740, or on the web at www.copyright.com. Requests to
the Publisher for permission should be addressed to the Permissions Department, John Wiley &
Sons, Inc., 11 1 River Street, Hoboken, NJ 07030, (201) 748601 1, fax (201) 7486008, email:
permreq@wiley.com.
Limit of Liability/Disclaimer of Warranty: While the publisher and author have used their best efforts in
preparing this book, they make no representations or warranties with respect to the accuracy or
completeness of the contents of this book and specifically disclaim any implied warranties of
merchantability or fitness for a particular purpose. No warranty may be created or extended by sales
representatives or written sales materials. The advice and strategies contained herein may not be suitable
for your situation. You should consult with a professional where appropriate. Neither the publisher nor
author shall be liable for any loss of profit or any other commercial damages, including but not limited to
special, incidental, consequential, or other damages.
For general information on our other products and services please contact our Customer Care Department
within the U.S. at 8777622974, outside the U.S. at 3175723993 or fax 3175724002.
Wiley also publishes its books in a variety of electronic formats. Some content that appears in print,
however, may not be available in electronic format.
Schetzen, Martin.
Linear timeinvariant systems/
Martin Schetzen
p. cm.
Includes index.
ISBN 047 1231452 (cloth: alk. paper)
www.ebook3000.com
IEEE Press
445 Hoes Lane, Piscataway, NJ 08855
Technical Reviewers
Rik Pintelon, Yrije Universiteit Brussel
Bing Sheu, Nassda Corporation
www.ebook3000.com
PREFACE
www.ebook3000.com
X PREFACE
not say that the refined model represents reality; rather, one only claims that experi
ments proceed in accordance with the model. In this sense, science is not directly
concerned with “reality.” The question of reality is addressed in philosophy, not
science. However, there are areas of science and philosophy which do influence each
other. Some of these are briefly mentioned in our discussion of system models.
As an illustration, the electron is a model of an object that has not been observed
directly. In an attempt to predict certain experiments, the electron was fist modeled
as a negatively charged body with a given mass which moves about the nucleus of
an atom in certain orbits. This model of the electron helped to predict the results of
many experiments in which the atom is probed with certain inputs such as charged
particles and the output is the observed scattered particles. This model also helped
predict the results of experiments in which the atom is probed with electromagnetic
fields and the output is the observed spectra of the radiation from the atom.
However, to predict the results of other experiments, this model of the electron
had to be modified. The model of the electron has been modified by giving it
spin, a wavelength, and other properties. Does the electron really exist? Science
does not address that question. Science just states that experiments proceed as if
the electron exists.
The modem development of engineering and science requires a deeper under
standing of the basic concepts of system theory. Consequently, rather than an appli
cationsoriented presentation, basic concepts and their system interpretation are
emphasized in this text. The chapter problems are to help the reader gain a better
understanding of the concepts presented. To study this text, the student need not
know mathematics beyond basic calculus. Any additional required mathematical
concepts are logically developed as needed in the text. Even so, all the mathematics
used in the text is used with care. Mathematical rigor is not used; that is the province
of the mathematician. However, mathematics is used with precision. For example,
the impulse is not something with zero width and infinite height. The accurate
development of the impulse presented also lends greater insight to its various appli
cations discussed in the text. The careful discussion and application of mathematics
results in the student having a better appreciation of the role of mathematics and a
more sophisticated understanding of its application in science and engineering.
Linear systems from a functional viewpoint is logically developed in this text.
Each topic discussed lays the basis and motivation for the next topic. In order that
the development be consistent with a systems orientation, many new results and also
new derivations of classic results from a systems viewpoint are included in this text.
Thus, many topics such as the Fourier and Laplace transforms and their inverse are
not just stated. Rather, I have developed new methods to motivate and derive them
from system concepts that had been developed previously.
The text begins with a discussion of systems in general terms followed by a
discussion and development of the various system classifications in order to motivate
the approach taken in their analysis. The timedomain theory of continuoustime
linear timeinvariant (LTI) systems is then developed in some depth. This develop
ment leads naturally into a discussion of the system transfer function, gain, and
phase shift. This lays the basis for a development of the Fourier transform and its
www.ebook3000.com
PREFACE Xi
inverse together with its system theory interpretation and implications such as the
relation between the real and imaginary parts of the system transfer function.
The discussion of the Fourier transform and its inverse motivates the development
of the bilateral Laplace transform and a full discussion of its system interpretation.
One important class of systems which is analyzed is that of passive LTI systems.
Although, as discussed in the text, there is no physical law that requires a system to
be causal, it is shown that a passive LTI system must be causal. Constraints that the
impedance function must satisfy are then obtained and interpreted.
A new approach to the unilateral Laplace transform is presented by which the
bilateral Laplace transform can be used in the transient analysis of LTI systems. The
splane viewpoint is then used to discuss basic filter analysis and design techniques.
The discussion of the splane viewpoint of systems concludes with the analysis of
feedback systems and their stability, interconnected systems, and block diagram
reduction.
Because system theory is the theory of models and the construction of models is
one of the main objectives of science, a discussion of the consistency of models and
some of the paradoxes in LTI system theory that can arise due to improper modeling
is given. The text concludes with an introductory discussion of the statevariable
approach to system analysis and the types of problems for which this approach is
advantageous. Thus the textual material lays a solid foundation for further study of
system modeling, control theory, filter theory, discrete system theory, statevariable
theory, and other subjects requiring a systems viewpoint.
I thank Prof. John Proakis, who was department chairman during the years I spent
writing this text, for his support and assistance. Also, my heartfelt thanks to my wife,
Jeannine, for her encouragement and help in the tedious job of proofreading.
V
Vi CONTENTS
Y= r [XI (1 .I1)
1
2 GENERAL SYSTEM CONCEPTS
The output and the input usually are functions of an independent variable such as
position or time. If they are only functions of time t , Eq. (1.11) is written in the form
m = T [ml (1.12)
This is a symbolic representation of the rule that governs the changes in the diameter
of the eye's pupil due to changes in the light intensity impinging on the eye.
As another example, to study changes in the intensity of light, i(t), from a light
bulb due to changes in the voltage, u(t), impressed across its filament, the system is
one in which the input is the filament voltage, which is a function of time, and the
output is the light intensity, which also is a function of time, so that
4 ( P ) = 1M P ) l (1.15)
In this system, both the input charge density and the output potential field are
functions only of position, p . Another example in which the input and output of a
system are functions only of position is a system used to study the deflection of
beams. In such a system, the input is the force on the beamf(p), which is a function
of the position along the beam at which the force is applied, and the output is the
4SYSTEM
X f
"'
Fig. 1.12 Schematic of a SISO system.
1.1 THE SYSTEM VIEWED AS A MAPPING 3
beam deflection d ( p ) , which also is a function of position along the beam. Thus the
relation can be expressed as
(1.16)
In Eq. (1.12), T is called an operator because the output function y(t) can be
viewed as being produced by an operation on the input function x(t). The statement
that y(t) is the response of a system to the input x(t) means that there exists an
operator 1or, equivalently, a rule by which a given output time function y(t) is
obtained from a given input time function x(t).
Another equivalent way of thinking about a system is to consider the input x ( t )
being mapped into the output y(t). This viewpoint can be conceptualized as shown in
Fig. 1.13. In the figure, the set of all the possible inputs is denoted by X and the set
of all possible outputs is denoted by Y. As illustrated, the input time functions
denoted by x1 and x, are both mapped into the same output time function denoted
by y,. Note that there can be only one time function resulting from a rule being
applied to a given input. Thus if the relation between the system input and output is
y(t) = ~ ? ( t )the , rule is that the output value at any instant of time to, is equal to the
square of the input value at the same time to. Clearly, for this system there is only
one output for any given input. However, note that many different inputs can
produce the same output. For example, consider the input to be a waveform that
jumps back and forth between +2 and 2. Irrespective of the times at which the
jumps take place, the output will be +4 so that for this system, there are an infinite
number of input waveforms that produce the same output waveform. On the other
hand, an example of a relation that cannot be modeled as a system is one in which
y(t) = [ x ( t ) ] ' / * . This cannot be modeled as a system because, in taking the square
root of x(t), there is no general rule by which the correct sign of y(t) can be known
because if x(to) = 4, is y(to) = +2 or is y(to)= 2? However, if the rule is that the
output is the positive square root of the input, then there is only one possible output
for any given input and the relation can be modeled as a system. In terms of the
mapping concept, a system is said to be a manytoone mapping because many
different inputs can result in a particular output, but a given input cannot result in
more than one output. Our discussion in this text will center on SISO systems with
inputs and outputs that are functions of time.
The inverse of a given system is one that undoes the mapping of the given system
as shown in Fig. 1.14. Note that the system inverse is a system, so that its operator
X Y
Fig. 1.13 The mapping of a the operator 1.
4 GENERAL SYSTEM CONCEPTS
must be a manytoone mapping of the set of its inputs, Y , to the set of its outputs,
X.This means that if a system inverse is to exist, the operator of the system must be
a onetoone mapping of the set of its inputs, X,to the set of its outputs, Y . As an
example, if the system mapping were one as shown in Fig. 1.13, there would be no
rule for the system inverse to map its input y , because the correct output, x, or x2,
could not be determined. We thus conclude that a system inverse exists ifand only i f
the system operator is a onetoone mapping. Clearly, the inverse of a squarelaw
device for which y(t) = 2 ( t ) and x(t) is any waveform does not exist. However, if
the set of inputs, X,is restricted to time functions that are never negative so that
x(t) 3 0, then the mapping is onetoone and a system inverse does exist relative to
that restricted class of inputs.
System analysis is the determination of the rule 1of a given system. If nothing is
known a priori about the given system, then one could only perform a series of
experiments on the system from which a list of various inputs and their correspond
ing outputs is made. The difficulty with this is manifold. First, it can be shown that it
is theoretically impossible to make a listing of all possible inputs not because of
human frailty but because there are more possible input time functions than it is
theoretically possible to list. A mathematician would say that the set of possible
inputs is not listable or, equivalently, not countable. To circumvent this problem, one
might attempt to make a list of just a judicious selection of possible inputs and their
corresponding outputs. However, such a listing would not characterize the mapping
because the system output due to an input that is not on the list would not be known.
Even if an input were close to one on the list (that is, they are approximately equal),
it could not be concluded that the two corresponding outputs were close to each
other. For example, it might be that the input, as opposed to the one on the list,
caused a relay in the system to temporarily open and thereby result in an entirely
different output. If it were known for the given system that inputs that are close to
each other result in outputs that are close to each other (such systems are called
continuous systems), then such a listing could be used to obtain the approximate
system response to any input that is close to one on the list. We thus note that,
without more knowledge of the system operator, 1,such a listing is relatively
useless. Aside from these problems, a listing would not result in a comprehensive
knowledge of the rule 1because it would be an onerous, if not impossible, task to
deduce the rule 1just from examining a collection of waveforms.
We thus conclude that to do any meaningful system analysis, some a priori
knowledge of the system operator must be known. It is a common problem in
1.3 TIMEINVARIANT (TI) SYSTEMS 5
This equation states that if the rule does not change with time, then the response to
x(t) shifted by to seconds must be the output y(t) also shifted by the same amount, to.
It is clear that if Eq. (1.31) is satisfied for a given system no matter what the input
x(t) or value of to used, then the system is TI. We thus can state an operational
definition of time invariance: A system is timeinvariant (TI) ifEq. (1.31) is satisfied
for any input, x(t), and any time shift, to.
To illustrate this operational definition, consider the resistor network shown in
Fig. 1.31. As shown, the resistance of each resistor varies with time. The system
input is x(t), which is an applied voltage, and the system output y ( t ) is the voltage
across the resistor rb(t).The relation between the system output y(t) and its input x(t)
is
(1.32)
This is the rule by which the system response is obtained from its input. Clearly, this
system is TI if the resistor values do not vary with time. However, is it possible for
the resistor values to vary with time and yet the system be TI? We shall use the
operational definition of TI to answer this question. For this, we must show that the
system is TI if the system response to x(t  to) is equal to y(t  to) for any x(t) and
for any value of to. This can be viewed schematically as shown in Fig. 1.32. The box
that is labeled ( t  to) represents an ideal delay system for which the output is its
input delayed by to seconds. The output z(t) of the top block diagram is the system
response to the input x(t  to), while the output of the bottom block diagram is
y(t  to). To show that the system is TI, we must show that z(t) = y(t  to) for any
input x(t) and for any delay to.
For our illustrative example, we have from Eq. (1.32) that the system response to
the input x(t  to) is
(1.33)
(1.34)
To understand the difference between these two equations, note that Eq. (1.33) is
obtained by only shifting the input x(t) by to seconds whereas Eq. (1.34) is obtained
by shifting the input x(t) and the system rule 1by to seconds in accordance with Fig.
1.32. The system is TI if z(t) = y ( t  to) for any input x(t) and for any value of to.
From Eqs. (1.33) and (1.34) we observe that this is true for any x ( t ) only if
(1.35)
for any value of to. Our analysis is now made easier by considering the reciprocal of
each side of Eq. (1.35):
(1.37)
Yb(t) =
in which c is a constant. To see this, assume that the value of the ratio in Eq. (1.37)
at the times t = t , and t = t2 differ. If this were so, then Eq. (1.36) would not be
satisfied for t = t , and to = t ,  t2.
We thus note that the system of our example is TI only if Eq. (1.37) is satisfied. If
Eq. (1.37) is satisfied, then, from Eq. (1.32), the relation between the system input
and output can be expressed as
The fact that the system is TI is easily seen because it is clear from Eq. (1.38) that
the rule by which y ( t ) is obtained from x ( t ) does not vary with time. Note that a
system can be TI even though the elements of which the system is composed vary
8 GENERAL SYSTEM CONCEPTS
with time. To be a TI system just means that the rule by which the output is obtained
from the input does not vary with time. A circuit in which one or more elements vary
with time is called a timevarying circuit. However, as we observe from our example,
the circuit can be a timevarying circuit while the system defined from the circuit is
timeinvariant. Care must be taken not to confuse circuit theory and system theory.
Note hrther that if the system output were defined as the current i(t) through the
resistors in Fig. 1.31 instead of the voltage y(t), then the relation between the input
x(t) and output i(t) would be
(1.39)
+
Such a system would be TI only if r,(t) rb(t)= constant. Thus the system with the
output y(t) can be timeinvariant, while the system with the output i(t) is time
varying. We thus observe that whether the defined system is timeinvariant depends
on what is defined as the system input and the system output. This is so because it is
only the rule by which inputs are mapped into outputs that determines whether the
system is TI, and the rule depends on what is called the input and what is called the
output.
We shall be concerned almost exclusively with timeinvariant systems in this text.
(1.41)
in which x is the input amplitude and y is the output amplitude. Equation (1.41) is
called the transfer characteristic of the nomemory system. The function f in Eq.
(1.41) is a rule by which an amplitude x is mapped into an amplitude y . As such it
must be, as discussed in Section 1.1, a manytoone mapping. The procedure for
determining the response y(t) of a nomemory system simply is to determine, at each
time instant, the output amplitude from the input amplitude at that instant in accor
1.4 NOMEMORY SYSTEMS 9
dance with Eq. (1.41). There are several types of nomemory systems of importance
which are discussed below. Also included in this discussion is the definition of some
notation and basic functions of importance for our subsequent discussions.
y=kk (1.43)
The graph of this relation is simply a straight line with a slope of K as shown in
Fig. 1.41.
In system representations, the ideal amplifier is represented by either of the block
diagrams shown in Fig. 1.42.
(1.44)
y = Eocu(x) (1.45)
X ( t 4 Y O
(4 (b)
Fig. 1.42 Block diagrams of an ideal amplifier.
in which
ifx<O
(1.46)
ifx>O
is called the unit step function.’ We shall find the function u(x) to be very useful in
our study of systems.
As an illustration of the halfwave rectifier operation, we determine the output y(t)
when the input x(t) is a sinusoid. The sinusoid is a waveform given by Eq. (1.47)
and illustrated in Fig. 1.44.
Note that a periodic waveform must extend from t = 00 to t = 00 because other
wise no shift ofx(t) will result in the same time function. The positive values of z for
which Eq. (1.48) is satisfied are called periods of x(t); the smallest positive value of
z for which Eq. (1.48) is satisfied is called the fundamental period of the waveform
x(t). In Fig. 144, the fundamental period of the sinusoid is T , while 2T, 3T, . . . are
simply periods of the sinusoid.
The value of T for the sinusoid can be determined by substituting Eq. (1.47) into
Eq. (1.48) to obtain
Y slope = K
0 X
’Some texts define u(0) = 1; others define u(0) = 0. I’ve defined u(0) = 1/2 not to be different but rather
for consistency in system theory. A reason for my choice will be given in Section 2.4.
1.4 NOMEMORY SYSTEMS 11
+
Since sin(+ 2nn) = sin(+) in which n is an integer, this equation can be satisfied
only if, for all values o ft ,
wt + wz = wt + 2nn
or
wz = 2nn, n = f l , f 2 , f 3 , . .. ( 1.410)
For the sinusoid, we thus obtain that the values of z for which Eq. (1.49) is satisfied
are
2nn
z=, n = f l , f 2 , f 3,... (1.41 1)
Q
w = 2nf (1.412)
in which f is the frequency in hertz (which is cycles per second). Note that because
w is the radian frequency in radians per second, the factor 271 must have the dimen
sions of radians per cycle. It is not dimensionless! Substituting Eq. (1.412) in Eq.
(1.41l), we obtain
n
T= n = f l , f 2 , f 3 , . .. (1.413)
f’
The smallest positive value of z is the hndamental period so that the hndamental
period of the sinusoid is
1
T= ( 1.414)
f
Another important relation concerning the sinusoid is that between a time shift
and a phase shift. An expression for the sinusoid delayed by to seconds is, from Eq.
(1.47),
0 = 0to (1.417)
We thus note that for a sinusoid a delay of to seconds is equivalent to a phase shift of
coto radians. Later in this text we shall express certain waveforms, x(t), as a linear
combination of sinusoids:
x(t) =
n
Cn sin(wn + 4,J (1.418)
(1.4 19)
in which
en = (1.420)
A graph of the sinusoidal phase shift versus frequency, w, is thus a straight line
passing through the origin with a slope equal to to, which is the time shift of x(t).
We now determine the output, y(t), of the halfwave rectifier for a sinusoidal
input, x ( t ) . In accordance with Eq. (1.44), it is
(1.42 1)
Thus, for the sinusoidal input given by Eq. (1.47) the output is
Figure 1.45 is a graph of Eq. (1.422). This nomemory system is called a halfwave
rectifier because, as seen from Fig. 1.44, the output is the input with the negative
half eliminated.
1.4 NOMEMORY SYSTEMS 13
(1.423)
Y = Klxl (1.424)
In Eq. (1.424), the vertical bars indicate the absolute value of x so that
x ifx<O
1x1 = (1.425)
x ifxbO
In essence then, the output of the fullwave rectifier is K times the magnitude of the
input. For the sinusoidal input x(t) given by Eq. (1.47), the fullwave rectifier output
is
slope = K
0 X
0 T t
Fig. 1.47 Graph of Eq. 1.426.
1 ifx<O
(1.429)
1 ifx>O
This nomemory system is called a hardlimiter because the output is limited to two
nonzero values. Thus the output y(t) for an input x(r) is
For example, if the input x(t) is the sinusoid given by Eq. (1.47), the corresponding
output is as shown in Fig. 1.49.
Y
Yo
X
YO
1.4 NOMEMORY SYSTEMS 15

 T
Yo
T t
We’ll illustrate the softlimiter operation by determining its output y(t) for the
sinusoidal input x(t) given by Eq. (1.47). First consider the case for which 1A 1 < xo.
Note that then Ix(t)l < xo so that the input amplitude traverses only the central
portion of the softlimiter characteristic for which
(1.432)
= A
Yo sin(ot)
XO
We thus note that if Ix(t)l < xo, then the softlimiter can be modeled as an ideal
amplifier with a gain K =yo/xo. Note that the gain is equal to the slope of the
transfer characteristic.
yo ..............
............ XO
x ? YO
For input amplitudes larger than xo, limiting occurs. To examine its effect, again
consider the input to be the sinusoidal waveform given by Eq. (1.47) but with
IA I > xo. For this input, the resulting output waveform f i t ) is shown in Fig. 1.411.
The output y(t) is equal to yo whenever x(t) > xo so that, for example, we note
that y(t) = yo for t , < t < t2 and y(t) = yo for t3 < t < t4. In between these times
Ix(t)l < xo so that y(t) in these intervals is given by Eq. (1.432). To complete the
determination ofy(t), the times t,, t2, t 3 ,and t4 must be determined. First, the time t ,
is the time at which y(t) first has the value yo so that this must be the time x(t) first
has the value xo. Consequently, x ( t l ) = xo. Substituting the expression for x(t) given
by Eq. (1.47), we then have
(1.434)
A solution of this equation always exists because xo/ IA I < 1. Once t, is known, the
other times are easily determined by using the symmetry of the sinusoid. For exam
ple, note that the time interval between t = 0 and t = t , is equal to the time interval
between t = t2 and the time of the first zero crossing to its right, which is at t = T i
where T is the hndamental period of the sinusoid. Thus
so that
t2 = i T  t , (1.436)
in a similar manner
t3 = i T + t, and t4 = T  t, (1.437)
All the transition times of y(t) can be determined in a similar manner. From this, the
length, L, of a clipped interval can be determined because
L = t*  t , = $2  24 (1.438)
1 1
T =  =  = 20 x lop3 s = 20 ms (1.439)
f 50
Then, from Eq. (1.438) we obtain
t1=?
'['? T  L ] :['2" 8 ] m s = l m s
= (1.440)
The nomemory systems and the functions discussed above are some of the
important ones worth keeping in mind. Other nomemory systems can be analyzed
in a manner similar to the technique illustrated above because a nomemory system
is a system for which the output value at any instant is simply a function of the input
value at the same instant. Remember also that a function,f(.), is just a rule by which
a value x is mapped into the value f (x). This view of a function will be important in
our latter discussions.
We shall examine some simple systems with memory in this section in order to gain
an understanding of some of the basic problems that system memory introduces in
the determination of the system response to a given input. First, a system with
memoly is simply dejined as one that is not a nomemory system. Systems with
memory are ubiquitous in the physical world. As a simple illustration, consider the
pupillary system described by Eq. (1.13). That system clearly is one with memory.
This is made evident by considering what occurs when you walk from a sunny place
into a dimly lit room. In the sunny place, the diameter of your eye's pupil is small to
limit the light intensity at the retina. When you walk into the dimly lit room, a bit of
time is required for the diameter of your pupil to increase sufficiently for there to be
enough light at your retina. That is, the present diameter of your eye pupil is
dependent not just on the present light intensity, but also upon the past light inten
sity. A system with memory is one for which input values at times that are not the
present time affect the present value of the output.
18 GENERAL SYSTEM CONCEPTS
x q m + $ @
y(t)=K f co
x(a) do (1 5 1 )
The output of the ideal integrator, according to Eq. (1.5l), is K times the area under
the infinite past of the input x(t). A block diagram representation is shown in Fig.
1.52.
To illustrate this, we shall determine the output y(t) when the input is the rectan
gular pulse given by Eq. (1.52) and shown in Fig. 1.53.
A ifO<t<T
x(t) = (1 5 2 )
0 otherwise
x(t) = A ( $ ) (1.53)
1 ifO<t<l
if t = O or 1 (1.54)
0 otherwise
This function is useful in expressing a rectangle of any width and location. For
f it').
example, consider the expression r  To determine its graph, we use the rule
O T t
Fig. 1.53 The rectangular input pulse for the example.
(or equivalently, the mapping) given by Eq. (1.54), from which we have that
r ( 7 ) = 1 whenever the value of whatever is in the parentheses is between
zero and one. Consequently we have that r f ito)
~
t  to
= 1 whenever 0 < < 1
T
or, for T > 0, whenever 0 < t  to < T or, equivalently, whenever to < t < T to. +
The graph, shown in Fig. 1.54, is seen to be a pulse with an amplitude of one and a
width of T which starts at t = to.
To determine the output y(t) for the example, we use Eq. (1 Sl), which states that
for any given value o f t , y(t) equals K times the area under the curve of x(a) in the
interval oo < a < t. Note that t is a constant in the integration. First we note that
the graph ofx(a) is as shown in Fig. 1.55. It is identical to the graph of Fig. 1.53
because the same rule x(.) is applied to t and to (T.This means that the same value of
the function is obtained when t and a have the same value because then the value of
the expression in the parentheses is the same. It is important to fully understand the
concept that a function is simply a rule and also how to use the rule to obtain the
graph in a manner similar to that used to obtain the graphs of Fig. 1.54 and Fig. 1.5
5.
We determine y(t) by evaluating Eq. (1.51) for various ranges of t. First, for
t < 0 note that the integral is only over negative values of o.Thus y(t) = 0 for t < 0
because x(a) = 0 for a < 0. Now for the range 0 < t < T , we note from Fig. 1.55
that y(t) which is equal to K times the area under x(a) in the range a< a < t, is
simply K times the area of a rectangle with height A and width t so that y(t) = KAt.
Finally, for the range t > T , note that K times the area of x(a) in the range
oo < a < t is just K times the area of the rectangle with height A and width T
so that y(t) = KAT. In summary, we have obtained
for t < 0
(1.55)
1 '.
Fig. 1.54 f
Graph of r 
;to).
20 GENERAL SYSTEM CONCEPTS
O T cs
Fig. 1.55 Graph of x ( ~ ) .
The expression for y(t) can be written more compactly with the use of the unit
step function defined in Section 1.4B as
With the use of the unitrectangle function, another compact expression for y(t) is
Note that we have expressed y(t) in four different forms [the graph of Fig. 1.56 and
the expressions of Eqs. (1.55), (1.56), and (1.57)]. Normally, there are many forms
available to express a given function. The form of the expression to use is that which
is most convenient. A graph is not less “mathematical” than an equation. Note that
the graph of the input and not its equation was used to obtain y(t) because it was
more convenient to use that form of the input expression for the integration.
The systems discussed to this point can be used as subsystems of a larger system
such as the system shown in Fig. 1.57. In the figure, y(t) is the output of the
summer. The symbol @ represents a summer for which the output is equal to the
sum of its inputs. Note that the arrows represent the direction of signal flow; without
them, which signals are inputs and which are outputs would not be known. It is
important that they be included in every block diagram. From the block diagram we
then have
~ ( t=) 3 J’
oo
~ ( a da
) (1.59)
The output z(t) can be determined using Eq. (15 9 ) once y(t) is determined from
Eq. (1.58). Of course, Eq. (1.58) can be substituted in Eq. (1.59) to obtain an
equation for z(t) in terms of only x(t). At times this is desirable; other times it can
O T t
+
y(t) = u(t)  2u(t  to) u(t  2to) (1 5 1 0 )
With the definition of the unitstep function, this expression is easily seen to be
A graph ofy(t) is shown in Fig 1.58. Now z(t) is easily obtained with the use of Eq.
(1.59) by following the integration procedure discussed in Section 1.5B. A graph of
z(t) is shown in Fig. 1.59.
A few different forms in which z(t) can be expressed analytically are
1O for t < 0
for 0 .= t < to
z(t) = ( 1.5 12)
(6to  3t) for to < t < 2t0
for t r 2t0
i=b 1
and also
(1.514)
It is worth verifying the three different forms given in order to become more adept in
the use and visualization of the unitstep and the unitrectangle hnctions.
We’ll illustrate the use of the concepts developed to this point by developing a
simplified model of a physical situation in which echoing occurs. Every model of
a physical system is, to some extent, a simplified model because there always are
factors that are not included. For example, in developing a model of the Earth’s
motion about the sun, the effect of the gravitational pull of the other planets and the
asteroid belt on the other side of the planet Mars is often ignored. These forces are
small and so only slightly affect the Earth’s motion. If, for example, the effect of
Mars on the Earth’s motion were desired, it could be included by considering its
small effect on the Earth’s motion by a separate perturbation calculation. To
construct a model of a physical system, one must first determine which factors to
include and which to ignore. Very simplified models often are used to understand
some basic phenomenon such as the model we’ll develop to obtain a basic under
standing of echoing. Complicated models that include many factors often obscure
the basic phenomenon but are constructed to obtain better numerical results. Thus,
the determination of the factors to include in a model is governed by the purpose for
which the model is being constructed and the relative size of the effect of each factor.
Our ability to construct simplified models of many phenomena has resulted in our
ability to determine the basic physical laws that govern those phenomena. For
example, the basic laws of gravitation were able to be determined because simplified
models in which only two bodies are interacting could be constructed. The simplified
model of the Earth moving about the sun obtained by ignoring the gravitational field
due to all the other celestial bodies is such an example. If the gravitational forces
were such that the effect of the other celestial bodies could not be ignored, then the
simplest system model would be so complicated that the basic laws of Newton
probably would never have been discerned.
1.6 A MODEL OF ECHOING 23
Yft),
I
) Kp(t)
~ ( t=
= K2r(t  to)
(1.61)
= KIK2dt  to)
= KIK2y(t  2to)
Example 1 As the first example, the output y(t) to the rectangular input given by
Eq. (1.63) will be determined.
(1.63)
Because the system initially is at rest and the input is zero for t < 0, we have that y ( t )
and z(t) are zero for t < 0. Mathematically, as we have discussed, the statement
y(t) = 0 for t < 0 means that whenever the value of whatever is in the parentheses
of the function y(.) is negative, the value of the function is zero. Thus y(t  2t0) = 0
whenever t  2t0 < 0 or when t < 2t0 so that from Eq. (1.62b) we have that
z(t) = 0 for t < 2t0. This result can be viewed physically from Fig. 1.63 by
y(t)
I ’
noting that any change ofy(t) from zero will take 2t0 seconds to propagate through
the delay in the feedback path so that if y(t) is zero for t < 0, then z(t) = 0 for
t < 2to. This is shown in Fig. 1.64. We now can use Eq. (1.62a) to determine y(t) in
the interval 0 < t < 2t0 because we now know x(t) and z(t) in that time interval.
Because z(t) = 0 in that time interval, we then have, as shown in Fig. 1.64, that
y(t) = x ( t ) in the interval 0 < t < 22,.
We now can use Eq. (1.62b) to determine z(t) in the interval 2t0 < t < 4t0.
Because y ( t ) = A in the interval 0 < t < to, we conclude that y(t  2t0) = A when
0 < t  2t0 < to or, equivalently, when 2t0 < t < 3t0 so that, as shown in Fig. 1.64,
z(t) = KA in that interval. Also, because y(t) = 0 for to < t < 2t0, we also have that
y(t  2t0) = 0 when to < t  2t0 < 2t0 or, equivalently, when 3t0 < t < 4t0 so that,
as shown in the figure, z(t) = 0 in that interval. Note that, in accordance with Eq.
(1.62b), z(t) is K times y(t) delayed by 2t0 seconds.
Now that z(t) and x ( t ) are known in the interval 2t0 < t < 4t0, we can use Eq.
(1.62a) to determine y(t) in that range. Then, once y(t) is known, Eq. (1.62b) can be
used as above to determine z(t) in the next 2t0 time interval. In this manner, y(t) can
be determined for all time as shown in Fig. 1.64 which is drawn for K M 0.7. Note
the echo sequence; each succeeding echo is K times the previous one so that the
echo pulses decay exponentially. The essence of echoing thus is contained in our
model.
x(t) = Ar  (1.64)
A K2
0 to 2t0 3to 4to 54, 6to 7to 810 9to f
Because the system initially is at rest and the input is zero for t < 0, we have that
y(t) and z(t) are zero for t < 0. Using the same argument as in the previous example,
we then conclude that z(t) = 0 for t < 2t0. We now can determine y(t) in the interval
0 < t < 2t0 by using Eq. (1.62a) because x(t) and z(t) are known in that time
interval. As shown in Fig. 1.65, y(t) = x(t) in that time interval since z(t) = 0 there.
We now use Eq. (1.62b) to determine z(t) in the interval 2t0 = t < 4t0. Because
y(t) = A in the interval 0 < t < 2t0, we conclude as in the last example that
z(t) = KA = A in the interval 2t0 < t < 4t0. Consequently, we have from Eq.
(1.62a) that y(t) = A  A = 0 in that interval. Then, from Eq. (1.62b), we have
that z(t) = 0 in the interval 4t0 4 t < 6to. Thus, from Eq. (1.62a), we have that
y(t) = 0 in the interval 4t0 < t < 6to. Continuing in this manner, we arrive at the
waveforms shown in Fig. 1.65.
Note that for the system with K =  1, the response to an input pulse that is 4t0
seconds wide is a pulse that is only 2t0 seconds wide. This is pulse compression. In
many applications such as radar, very narrow pulses are required. The principle
illustrated by this example is often used to obtain pulses that are more narrow
than can be achieved by standard electronic circuits.
The input in both examples is a rectangular pulse; the pulse width is the only
difference. Yet the system responses (with K =  1) are significantly different output
waveforms. This occurs in systems with memory because the output at any instant
depends on the past values of the input waveform so that the output at each time
instant depends on the whole input waveform. This is not the case for nomemory
systems. As a result, the response to any given input is not obvious. Furthermore,
although we were able to determine the response for the two examples, the analysis
would become unwieldy if the system were not so simple. A better approach is
needed to analyze systems with memory. We begin the development of a better
approach in the next chapter.
I,Z
"
0
:I I: t
PROBLEMS
11 Each of the following is a description of a nomemory system with the input
x(t) and response y(t). For which does an inverse exist? If an inverse does not
exist, determine whether an inverse would exist if the amplitude of the
allowed inputs was constrained. If so, determine the amplitude range.
(a) Y A t ) = sin[x(t>l
(b) Y d t ) = tan[x(OI
(c) y,(t) = 2(t) 3x(t) 2 +
(d) Y d ( t ) = U[x(t)l
(e) = w  2i3
(t) Y/@)= ex(')
(8) Y&) = arctan[x(t>l
12 The input of the system shown below is the voltage, x(t), and the output is the
voltage, y(t). The two capacitors are initially uncharged. Are there conditions
on the time variation of the two capacitor values for which the system is time
invariant?
13 Sketch each of the following functions for T = 3 and to = 0.5. Make certain
all critical values are labeled.
(a) f ( t >= (i
 I,r(y>
(c) h ( t ) = sin(5nt)r
(" it,>
~
18 +
Let z(t) = x(t) y(t), where x(t) is a periodic waveform with a fundamental
period equal to T, and where y(t) is a periodic waveform with a fundamental
period equal to Ty. What are the conditions for which z(t) is periodic? For
those cases, determine its fundamental period, T,.
19 For each of the hnctions below, determine whether it is periodic and, if so,
determine its fundamental period.
+
(a) x,(t) = sin2nt 3 sin4nt
+
(b) xb(t) = sin t 3 sin 2t
+
(c) x,(t) = 2 sin t sin z/Zt
+
(d) xd(t) = 2 sin 6t 3 cos 7t
110 Sketch the input, x(t), and the output, y(t), of
(a) A halfwave rectifier
(b) A fullwave rectifier for the periodic input with a fimdamental period of
16 and for which
Draw a sketch of the system output, y(t). Label all important amplitudes and
times of y(t).
PROBLEMS 29
Iy
lY
Draw a sketch of the system output, y(t), for 8 < t < 8 to the input, x(t),
shown below. Label all important amplitudes and times of y(t).
114
0 8 4 0 4 8 t
I
17 5 0 5 17 X
Iy
116 A square wave with a fundamental period equal to 12.5 ps and amplitude equal
to 3 is to be generated using a sinusoidal oscillator and a hard limiter. Draw a
block diagram of the system and specify the frequency of the oscillator.
117 It is desired that the output of a soft limiter with the sinusoidal input
x(t) =A sin(o,t+ 4) be a good approximation of a square wave. For this,
it is desired to design the system so that the softlimiter output, y(t), will be as
shown in Fig. 1.410 with Iy(t)l = y o for at least 98 percent of the time.
Determine the required amplitude, A , of the input in terms of xo.
120 Determine and sketch the response, y(t), of the system shown below to the
input x(t) = u(t). Label all important amplitudes and times.
PROBLEMS 31
121 Determine and sketch the response, y(t), of the system shown below to the
input x(t) = r(t). Label all important amplitudes and times.
122 Determine the response, y(t), of the system shown below to the unit
rectangular input x(t) = r(t/2). Sketch and label all important amplitudes
and times. For credit, your work must be shown.
123 Determine the response, y(t), of the system shown below to the input
x(t) = 2r(t). Sketch and label all important amplitudes and times.
YQ),
t2
124 Determine and sketch the response, y(t), of the system ...own below to the
unit step input x(t) = Q). Label all important amplitudes and times.
125 The feedback system shown below is initially at rest. Determine the system
response for x(t) = r(t). Sketch the system response y(t) and label all
important amplitudes and times.
32 GENERAL SYSTEM CONCEPTS
126 The feedback system shown in the first diagram is initially at rest, and the
input is x(t) = v ( t / 4 ) as shown in the second diagram.
z(t)p$+p tI
x(t) 1
0 4 t
Sketch the system response, y(t) and label all important amplitudes and times.
127 The feedback system shown in the first diagram is initially at rest, and the
input is x(t) = Ar(t/4) as shown in the second diagram. Determine and
sketch the waveforms y(t) and z(t). Label all important amplitudes and times.
The basic reasons why systems must be classified was discussed in Section 1.2. This
motivated the classification of systems as being either timeinvariant (TI) or time
varying (TV). As discussed in Section 1.3, this text is concerned mainly with TI
systems. But even this classification is too general to be of much use. So we hrther
classified systems as being either nomemory or with memory. The analysis of no
memory systems was then discussed in Section 1.4. Some structurally simple
systems with memory were discussed in Sections 1.5 and 1.6. There we saw that
the analysis of systems with memory could be rather complicated, especially for
ones which are structurally complex. Thus a more refined approach is needed for
such systems. A problem with the development of a more refined approach is that
even the class of systems with memory which are timeinvariant is too broad. A
further classification is needed to develop the desired refined approach. The further
classification that has been found useful is to classify systems as being either linear
or nonlinear. A nonlinear system is simply a system that is not linear so that we need
to just define a linear system.
Consider a system with the response y ( t ) to the input x(t) as shown in Fig. 2.1 1. r f ;
for the given system, an input x l ( t )produces the output y l ( t ) and an input x2(t)
produces the output y2(t), then the system is a linear system if the input x(t) =
+ +
Clxl(t) C,x,(t) produces the output y ( t ) = C , y l ( t ) C g 2 ( t ) in which C , and
C2 are arbitrary constants. For the system to be linear, this condition must be
true for any two inputs, x l ( t ) and x2(t), and any two constants, C , and C,. Note
33
34 LINEAR TIMEINVARIANT (LTI) SYSTEMS
that the constants need not be real; they can be any two complex numbers. This
definition of a linear system derives from the exactly parallel mathematical definition
of a linear mapping.
A shorthand notation that we shall use is x ( t ) + ~ ( t )which
, means that the input
x(t) produces the output ~ ( t )Using
. this shorthand notation, the definition of linearity
can be expressed as follows: If, for any inputs, x l ( t ) and x2(t),
then
where C, and C, are arbitrary complex constants. In words, a linear system is one
for which the response to a linear combination of inputs is the same linear combina
tion of the responses to each input individually.
As a simple illustration, a squarelaw device is a system for whichy(t) = ??(t). To
apply the definition, we let the system input be
(2.12b)
The righthand side of Eq. (2.12c) is, according to the linearity definition, the
required output for it to be a linear system. To obtain a deeper understanding of
the linearity definition, we consider some special cases of the definition.
Special Case 1 First consider the special case for which C2 = 0. For this case, the
linearity definition states the following:
If
(2.13a)
2.1 LINEAR SYSTEMS 35
then
In words, this states that if the input amplitude of a linear system is changed by a
factor of C, but the shape of the input waveform remains unchanged, then the shape
of the corresponding output waveform remains unchanged and only its amplitude is
changed by the same factor of C , . This property is called the homogeneous property.
Thus, every linear system is homogeneous. The converse, however, is not true. That
is, a system can be homogeneous and not be linear. For example, consider the system
for which the response y(t) to the input x(t) is
(2.14)
This system is homogeneous, but it does not satisfy the linearity definition given by
Eqs. (2.11).
For the special case in which both C, and C2 are zero, we have from Eqs. (2.11)
that if x(t) = 0, then y ( t ) = 0. This can be written in our shorthand notation as
0 + 0. Note that because, for a linear system, the output waveform amplitude
(but not the shape) changes as the input waveform amplitude (but not the shape)
changes, a plot of the maximum output amplitude versus the maximum input ampli
tude can be made. From Eqs. (2.13), this plot would be a straight line that passes
through the origin.
Special Case 2 Now consider the special case for which C, = C2 = 1. For this
special case, Eqs. (2.11) state the following:
If
then
That is, the response of a linear system to a sum of inputs is the sum of the responses
to each individual input. This is called superposition. Thus, every linear system
satisfies superposition. Note that Eqs. (2.11) are a generalized form of Eqs.
(2.15). For this reason a linear system can be defined as a system that satisfies
generalized superposition.
We showed above that if a system is homogeneous, it is not necessarily linear. But
what of superposition? It can be shown that a system can satisfy superposition and
not be linear. However, a system that satisfies superposition is very close to being
linear, and thus examples of systems that satisfy superposition and are not linear are
rather contrived. The reason is that Eqs. (2.15) imply Eqs. (2.11) for the case in
36 LINEAR TIMEINVARIANT (LTI) SYSTEMS
which the constants C, and C, are rational constants. Thus the counterexample
would be one where Eqs. (2.11) are satisfied when C, and C2 are rational constants
but not when C, and C2 are irrational constants. One would not expect to encounter
a physical system with this property. Thus, if a physical system satisfies superposi
tion, there is a very good chance (but not a certainty) that it also satisfies generalized
superposition and thus is linear. However, a system definitely is linear if it is
homogeneous and also satisfies superposition. Thus, to show that a system is
linear, one must show either that it satisfies generalized superposition [Eqs.
(2.1l)] or that it is homogeneous [satisfies Eqs. (2.13)] and also satisfies super
position [Eqs. (2.15)].
To illustrate our discussion, consider the system shown in Fig. 2.12. The system
is a circuit with the input being the voltage x(t) and the output being the voltage y(t).
For the given system,
(2.16)
This system does not satisfy generalized superposition [Eqs. (2.1l)] and thus is not
linear. Of course, it is not necessary to check Eqs. (2.11) for this system to deter
mine whether it is linear because it is easily seen that the system is not homogeneous
since, from Eq. (2.16), y(t) = ~ Ra E when x ( t ) = 0 so that y(t) # 0 when
R, + R A
x ( t ) = 0. Note that the system of k g . i.12 is a linear circuit, but it not a linear
system. In circuit theory, one is concerned with the study of the interactions between
the elements of which the circuit is composed; in system theory, however, one is
concerned with the study of the mapping of inputs to outputs. The system of Fig.
2.12 is a linear circuit because it is composed of linear elements, while it is not a
linear system because the mapping of inputs to outputs does not satisfy generalized
superposition. Care must be taken not to confuse circuit theory and system theory
because the different concerns leads to significant differences of the theories devel
oped for their study.
It should be noted that no physical system is truly linear. The parameter values of
any physical system will change for sufficiently large values of the input amplitude.
For example, the resistor values in the system of Fig. 2.12 will change due to
overheating if the current is sufficiently large. However, the resistor values will
change negligibly for some range of current values so that the system can be
modeled as a linear system if this range encompasses the range of current values of
interest. We similarly model many physical systems as being linear if they satisfy
generalized superposition for input amplitudes over the range of interest. Also note
that no physical system is truly timeinvariant because physical components do age.
However, if the system mapping of inputs to outputs doesn't change measurably over
the time interval of interest, then a good model is a timeinvariant one.
The class of systems with which we are mainly concerned in this text is the class of
systems that are both linear and timeinvariant. The theory of linear timeinvariant
(LTI) systems which we shall develop is of central importance in system theory
because many physical systems can be accurately modeled as LTI systems. Also, the
theory of LTI systems forms the basis of the theory for linear timevarying (LTV)
systems and also for some important classes of nonlinear systems.'
Let the input of an LTI system be composed of a linear combination of waveform
as given by Eq. (2.2la)
(2.2 la)
Then, because the system is linear, it satisfies generalized superposition so that the
system response is
(2.21b)
n
in which w,(t) is the system response to the input u,(t). Now, if each input waveform
is a translation of a particular waveform so that u,(t) = u(t  zn), then because the
system is TI, wn(t)= w(t  7,) where u(t) + w(t). We thus have that if the input of
'This extension to nonlinear systems is developed in M. Schetzen, The blterra and Mener Theories of
Nonlinear Systems, John Wiley & Sons, 1980, updated and reprinted by Krieger Pub. Co., 1989.
38 LINEAR TIMEINVARIANT (LTI) SYSTEMS
(2.22b)
where w(t) is the LTI system response to u(t). This result states that if we know the
system response of an LTI system to a particular input, then as given by Eqs. (2.22),
we can determine the system response to any input which can be expressed as a
linear combination of the particular input and translates of that waveform. Equations
(2.22) are a statement of a fundamentalproperty of LTI systems. This fundamental
property is the basis of the theory we shall develop for the analysis of LTI systems.
However, we consider some illustrative examples first to gain a better appreciation of
the implications of this property.
Assume that an experiment was performed on a particular LTI system from which
it was observed that the response to the input
is the output
as shown in Fig 2.21. The system response to a wide variety of input waveforms can
be determined from this one experimental result. For example, the response to
(2.24a)
which is shown in Fig. 2.22, can be determined by noting that we can express x2(t)
as
Thus, in accordance with Eqs. (2.22), the response of the LTI system to the input
x2(t)is
2.2 LINEAR TIMEINVARIANT (LTI) SYSTEMS 39
With the use of Eq. (2.23b), this can be expressed in the form
+
y2(t) = tr(t) r(t  1) + (3  t)r(t  2) (2.24d)
A graph ofy2(t) is shown in Fig. 2.22. This result should be verified graphically by
using Fig. 2.21.
Now consider the input shown in Fig. 2.23 which can be expressed as
In accordance with Eqs. (2.22) and (2.23), the system response to this input is
A graph of x3(t)andy3(t) is shown in Fig. 2.23. This result also should be verified
graphically with the use of Fig. 2.21.
As a third example, consider the input shown in Fig. 2.24, which is
and substituting the expression from Eq. (2.23b), this can be expressed as
A graph of the input and corresponding output is shown in Fig 2.24. Note in Eq.
i)
(2.26a) that the two waveforms x 1( t ) and x , ( t  overlap in the expression for x4(t).
Even so, the result expressed by Eqs. (2.22) can be applied as in this example.
As a last example, the system response to the input x,(t) = r(2t) will be deter
mined. For this determination, first note that we can express this input in the form
= E(l)nX1
n=O
(t  );
You should verify this result graphically. Consequently, in accordance with Eqs.
(2.22), the system response is
A graph of ys(t) can be obtained by substituting Eq. (2.23b) in Eq. (2.27b) and
manipulating the resulting messy expression. A better procedure is to determine the
waveform by plotting each term of Eq. (2.27b) and graphically adding the straight
lines in each interval. This procedure is particularly simple in this example because
the sum of straight lines is a straight line, and a straight line is determined by just
two points on the line. Clearly, the two points to choose are the ends of each time
segment. Before launching into a calculation, it is good practice to examine the
various available procedures and choose the one that is simplest and lends insight
to the solution. The result for this example is shown in Fig. 2.25.
The basic result used for the development of a general characterization of an LTI
system is given by Eqs. (2.22), which was derived and discussed in Section 2.2.
From that result, we saw that the response of a given LTI system to a large class of
inputs can be determined from knowledge of the system response, w(t), to one
particular input, u(t). The class of inputs is that class which can be expressed as a
linear combination of translates of u(t) as given by Eq. (2.22a). The input, u(t), and
corresponding system response, w(t), are thus said to be a characterization of the
given LTI system for that class of inputs. A difficulty with the characterization as
discussed in Section 2.2 is that it often is difficult to determine the expression of a
given input, x(t), in the form given by Eq. (2.22a). Without such an expression, the
system response, y(t), to the input, x(t), cannot be determined. What is needed is to
choose one particular input, uo(t) for which the expression of any given input, x(t), in
the form of Eq. (2.22a) is easily obtained. With such a choice, the response wo(t)of
a given LTI system to the particular input uo(t) would be a general characterization of
the given system because we then could obtain a general expression for y(t) in terms
of x ( t ) and the response wo(t).
To determine a good choice for u,(t), consider a segment of some arbitrary
waveform for x ( t ) and note that it can be approximated by a piecewiseconstant
curve, x,(t), as shown in Fig. 2.31. The approximation x,(t) can be considered as
a sequence of contiguous rectangles as indicated by the dashed lines. The width of
each rectangle is I: seconds. As shown, the rectangle midpoints are at t = kE for
k = 0, f l , f 2 , 4 3 , . . . . The height of the rectangle with its midpoint at t = nE is
chosen to be ~ ( I z E ) . Thus the height of the rectangle with its center at t = 0 is x(O),
the height of the rectangle to its right is x(E), and the height of the rectangle to its left
is x(E).
: ‘
/&
. I, ’ .
6 0 E iE 3’E 4E t
Note that the difference between x ( t ) and x,(t) can be made as small as desired by
choosing E sufficiently small. This can be expressed mathematically as
To obtain the system response y(t) to the input x(t), we shall obtainy,(t), which is the
system response to x,(t) and then let E + 0. We then shall assume that
(2.32)
That is, we shall assume in our development that, for any input x ( t ) , the difference,
y(t)  y,(t), goes to zero as the input approximation error, x ( t )  x,(t), goes to zero.
Systems for which this is true are called continuous systems. Thus our assumption is
that the LTI system is continuous. The continuity of an LTI system is closely allied
with its stability, and so we shall postpone a discussion of LTI system continuity and
its implications until Section 3.7 after our discussion of LTI system stability.
A function that is useful for our development is d,(t) which is shown in Fig.
2.32. As shown, it is a rectangle with its midpoint at t = 0, a width of E seconds, and
its area is equal to one. In terms of this function, a rectangle with its center at t = nE
and height equal to + E ) in Fig. 2.31 can be expressed as wc(ns)d,(t  n~).The step
approximation of x(t) is the sum of all these rectangles so that it can be expressed as
00
x&) = m(ns)d,(t  ne) (2.33a)
m=m
Note that Eq. (2.33a) is exactly in the form of Eq. (2.22a) with C, = m(nE),
v(t) = s,(t), and z, = nE. Thus, in accordance with Eq. (2.22b), the LTI system
response to x,(t) is
where h,(t) is the response of the given LTI system to S,(t). We now let E + 0 to
obtain y(t) in accordance with our discussion above. In the limit E + 0, Eq. (2.33b)
becomes
y(t)= 1 00
oo
x(o)h(t  0)d o (2.34a)
where
The integral in Eq. (2.34a) is called the convolution integral. Before discussing the
convolution integral, let us examine the limiting process to see that Eq. (2.34a) is
the correct limit of Eq. (2.33b).
First note that lim,+o does not mean that E becomes zero; rather, it means that E
becomes arbitrarily small. For example, consider the unit step function, u(tj, defined
by Eq. (1.46). Note that u(0) = 1/2. However, if E is nonnegative, then
lim6+oU ( E ) = 1 because no matter how small is E (but not zero), the value of U ( E )
is one. The value one is called the righthand limit of the unit step function at zero
since it is the value of the limit as the point zero is approached from the right. A
shorthand expression for this righthand limit is u(O+) = 1. Also note that
lim,+o u(E) = 0 because no matter how small E is (but not zero), the value of
u(E) is zero. Zero is called the lefthand limit of the unit step function at zero
and can be expressed in shorthand notation as u(0) = 0. To further illustrate a
limit, consider the expression, lim,+o 1 / = ~ 00. We do not mean by E + 0 that there
is a number such that the reciprocal of that number is infinite (remember that
division by zero is not an allowed operation).’ It just means that as E becomes
arbitrarily small, the reciprocal 1 / ~becomes arbitrarily large so that 1 / ~ can be
made as large as desired by making E sufficiently small. In summary, lim,+o just
means the limit as E becomes arbitrarily small and not the value when E has the value
of zero.
Recall that h,(t) is the LTI system response to S,(t). Thus in accordance with Eq.
(2.34b), h(t) is the limit as E + 0 of the LTI system response to a rectangular pulse
* If division by zero were allowed, then we could show that any two numbers are equal. For example, since
1 . 0 = 2 . 0 we could divide both sides of the equation by zero to obtain 1 = 2. Division by zero is the
basis of many mathematical puzzles; the cleverness of those puzzles is in the method by which the division
by zero is concealed.
44 LINEAR TIMEINVARIANT (LTI) SYSTEMS
d,(t) shown in Fig. 2.32. Physically, as E + 0, h,(t) keeps changing because the LTI
system response is different for different values of the rectangular width, E . However,
as E + 0, h,(t) approaches some waveform that we call h(t). A useful shorthand
expression is to say that h(t) is the LTI system response to a rectangular pulse with a
width E = O+. Note that h(t) is not the LTI system response to a rectangular pulse
with zero width and area equal to one because such a pulse is mathematically
meaningless. Rather, as discussed in the next section, it is the LTI system response
to a rectangular pulse with infinitesimal width and area equal to one.
We denote a pulse with an infinitesimal width, E = 0+, and area equal to one by
d(t) and call it the unit i m p u l ~ eThe
. ~ adjective “unit” refers to the fact that the area
of d(t) is one and ‘‘impulse’’refers to the fact its width is E = O+. The unit impulse,
d(t), is represented as shown in Fig. 2.33. The area of the impulse is indicated by a
value in the parentheses next to the arrowhead (which is one in this case). Note that
the impulse, &t), only has meaning in terms of a limit as E + 0. It is important to
keep this in mind in all your applications of the impulse. This is discussed in more
detail in Section 2.4. In terms of this definition of the unit impulse, we say that h(t) is
the LTI system response to d(t). We thus call h(t) the system unitimpulse response.
Again note that this means that h(t) is the limit as E + 0 of the LTI system response
to d,(t).4
It now is easy to see that the integral, Eq. (2.34a), is the correct limit of the
summation, Eq. (2.33b). To see this, first consider an integral of the form
(2.35)
Its value can be determined graphically for any desired value of t, say t = to, by
plottingf(o, to) versus a and determining the area under the curve in the interval
from CI to b. An approximate value of the area can be determined by approximating
f ( o , to)as depicted in Fig. 2.34. The area of the approximation shown is the sum the
areas of the rectangles, which is
(2.36)
0 t
Fig. 2.33 Depiction of the unitimpulse, 6(t).
Although we shall use s(t) for the unit impulse function in this text in keeping with the fashion of the
day, another notation used is u,(t).
Some problems that arise by considering the impulse width to be zero are discussed and illustrated in
Section 10.5.
2.4 THE UNITIMPULSE SIFTING PROPERTY 45
To better understand the unit impulse, consider the limit as E + 0 of Eq. (2.33a). In
accordance with Eq. (2.3l), the limit of the lefthand side is x(t). Thus, using our
discussion in Section 2.3, the limit as E + 0 of Eq. (2.33a) is
00
(2.41)
46 LINEAR TIMEINVARIANT (LTI) SYSTEMS
Note that this result states that the convolution of any function with a unit impulse is
equal to that function. We will verify the validity of this result by actually performing
the integration using the concepts discussed in Section 2.3.
Now as discussed in Section 2.3, the width of S(t) is 0+, which means that Eq.
(2.41) is just a shorthand notation for the limit
(2.42)
We thus must perform the integration given in Eq. (2.42) and then let E + 0 to
obtain the value of the integral in Eq. (2.41). For this, first note that Eq. (2.42)
states that for any given value oft, say t = to, x(to) is the limit as E + 0 of the area
under the product of x(a) and S,(to  a).
To illustrate the integration procedure, consider the waveform x(t) to be that
shown in Fig. 2.31. Then the graph of x(a) versus a is as shown in Fig. 2.41.
Note that the graph of x(a) versus a in Fig. 2.41 is identical with the graph of x(t)
versus t in Fig. 2.31. The reason, as discussed in Section 1.5, is that a function is
simply a rule by which the value of the number in the parentheses is mapped into a
value. Thus, in accordance with our discussion in Chapter 1, a function is a manyto
one mapping of numerical values into numerical values. For our present example,
x(t) is the value obtained by applying the rulewhich is denoted by x(.)to the
number t. The graph of x(t) versus t in Fig. 2.31 is simply a graph of this rule.
Consequently, if a. = to,then x(ao) = x(to) so that the graph of x(a) is as shown in
Fig. 2.41.
We now require a graph of S,(to  a). From Fig. 2.32, this is a rectangle with a
width of E and with its midpoint at the value of a for which to  a = 0. Because we
shall be taking the limit as E + 0, we shall consider E to be infinitesimal so that
S,(to  a) is an infinitesimally wide rectangle with its midpoint at a = to. A graph of
S,(to  a) thus is as shown in Fig. 2.42.
Using Figs. 2.41 and 2.42, the graph of the product x(o)S,(to  a) drawn for
to > 0 is as shown in Fig. 2.43. Now, if x(t) is differentiable about t = to,the graph
of x(a) is a straight line in the &wideregion about a = to. The reason is that E is
infinitesimal and the first approximation of a differentiable h c t i o n x(a) is a straight
line with a slope equal to the derivative of the function at the point a = to. Conse
quently, the graph of the product x(o)S,(to  a) is a trapezoid as shown in Fig. 2.43.
Q
t0
Fig. 2.42 Graph of S,(to  0).
The value of an integral is the area under the graph of the function being integrated.
For our case, the integral in Eq. (2.42) is just the area of the trapezoid in Fig. 2.43
which is equal to the trapezoidal width, E , times its height at its midpoint, x(to)/&.
Thus the value of the integral in Eq. (2.42) is x(to). Because this value does not
change as E becomes smaller, we note that the limit as E + 0 of the integral in Eq.
(2.42) is x(to). This is in accordance with the result given by Eq. (2.4l), which is
the shorthand notation for Eq. (2.42).
Our evaluation of Eq. (2.41) was for the case in which the function x(t) is
differentiable at t = to. What is the result if x ( t ) is discontinuous at t = to? For
this, consider x ( t ) to be discontinuous at t = to with a lefthand limit at to equal
to A and a righthand limit at to equal to B. As in Section 2.3, we express this as
x(to) = A and x(to+) = B. Using the procedure discussed above, the graph of
x(a)d,(to  c) is as shown in Fig. 2.44. As in Fig. 2.43, the width of the figure,
which is infinitesimal, has been enlarged greatly for ease of viewing. Again, because
x ( t ) is differentiable in the infinitesimal interval to the right and also to the left of
CJ = to, the graph is a straight line in each of these regions as shown. However, note
that the straight lines do not necessarily have the same slope because the derivative
of x ( t ) to the left and right of the discontinuity is not necessarily the same. The area
under the curve of Fig. 2.44 is the value of the integral in Eq. (2.42). The area
under the curve is the sum of the areas of the two trapezoids, which is seen to be
[(
equal to  x to  
2 2+ ( + 21
x to  . We obtain the value of the integral in Eq.
(2.41) by taking the limit as E + 0. Thus the value of the integral in Eq. (2.41) is
(2.43)
We thus note that, at a discontinuity, the value of the convolution of a function with a
unit impulse is equal the average of the leftand righthand limits of the function at
.........................................
B l s ............
to; t0 to+; 0
Fig. 2.44 Graph of x(o)b(to a) for the case in which x(t) is discontinuous at t = to.
2.5 CONVOLUTION
The response, y(t), to the input, x(t), of an LTI system was shown in Section 2.3 to
be given by the convolution integral, Eq. (2.34a):
00
where h(t) is the LTI system unitimpulse response. Note that the output for any
given input can be determined from Eq. (2.51) once the unitimpulse response, h(t),
A formal mathematical theory of a class of symbolic functions that have the same properties as those
derived above for the unit impulse has been developed by the French mathematician Laurent Schwartz and
published by him in Thiorie des Distributions, Vols. 1 and 2, Actualitk Scientifique et Industrielles,
Hermann & Cie., Paris, 1950 and 1951. I mention this because some texts on system theory include a
short outline of the Schwartz theory of distributions. This is my only mention of it because it really is not
needed for a discussion of the unit impulse and its properties. My discussion in this section is
mathematically accurate and, because it is physically based, it also lends a better physical understanding
for system theory than the symbolic theory of Schwartz.
2.5 CONVOLUTION 49
of a given LTI system is known. In this sense, the unitimpulse response of a given
LTI system completely characterizes the system mapping of inputs into outputs. For
this reason, the unitimpulse response will play a central role in our discussion of LTI
system theory. However, before beginning this discussion, we shall evaluate the
convolution integral for some cases in order to illustrate some techniques that can
be used for its evaluation and to gain some insight for interpreting the integral.
Example 1 For our first example, consider an LTI system with the unitimpulse
response
h , ( t ) = A6(t  t o ) (2.52)
y(t) = 1
00
m
x(a)AG(t  to  CJ)da (2.53)
Using the sifting property of the unit impulse developed in the last section, the value
of the integral is equal to the value of the function multiplying the unit impulse
evaluated at the location of the unit impulse. The function multiplying the unit
impulse is Ax(o). The unit impulse is located at the value oft^ for which its argument
is zero; this is the value of CJ for which t  to  CJ = 0 so that the unit impulse is
located at CJ = t  to. Thus we obtain
We observe that the output of an LTI system with a unitimpulse response given by
Eq. (2.52) is its input multiplied by a factor of A and delayed by an amount of to
seconds. Using the block diagram representation developed in Section 1.5, this
system can be represented as shown in Fig. 2.51. Note that the system is simply
an ideal amplifier with a gain of A in tandem with a delay of to seconds. It is called
an ideal amplifier because the input waveform is amplified without any distortion.
Example 2 For our second example, consider an LTI system with the unitimpulse
response
(t  to)
Fig. 2.51 Block diagram representation of the system with the unitimpulse response h , ( t ) .
50 LINEAR TIMEINVARIANT (LTI) SYSTEMS
Using the convolution integral, Eq. (2.5l), the response, y(t), of the given system to
an input, x(t), is
y(t) = / 00
m
x(o)Ku(t  a) do (2.56)
In accordance with the definition of the unit step, Eq. (1.46), it has the value zero
when its argument is negative and has the value one when its argument is positive.
Thus
0 for a > t
u(t  a) = (2.57)
1 foro < t
Consequently, the function being integrated is zero for a > t so that the integral can
be expressed as
y(t) = f
oo
x(a)K do
(2.58)
The value of the unit step at a = t is 1/2. However, it has no effect on the value
of the integral because the area under a point is zero. Equations (2.58) and (15 1 )
are identical. Thus, a block diagram of this system is as shown in Fig. 2.52, which is
the same as Fig. 1.52. Note that the system is simply an ideal integrator with a gain
of K .
Example 3 For our third example, consider an LTI system with the unitimpulse
response
K forO<t<T (2.59)
0 fort<Oort>T
In this equation, r(t) is the unitrectangle function defined in Section 1.5. To deter
mine the system response to an input, x(t), we again use the convolution integral, Eq.
(2.51). For this we require h3(t  a) as a function of a for a given value of t. From
Eq. (2.59) above we have that h3(t) = K only when 0 < t < T. Consequently, from
our discussion of a function as a mapping, we have that h3(t  a) = K only when
Fig. 2.52 Block diagram representation of the system with the unitimpulse response h2(t).
2.5 CONVOLUTION 51
0 < t  o < T . Solving for the range of o, we have that h,(t  o) = K only when
t  T < o < t. Keep in mind that t is a constant in the integration so that we are
determining the output, y(t), for a particular value oft. Thus
0 foroctT
K for t  T < o < t (2.5 10)
0 foro > T
Now, from the convolution integral, Eq. (2.5l), for a given value oft, y(t) is equal to
the area under the curve x(o)h(t  6).For our case, this product is
for o < t  T
for t  T < 0 < t (2.51 1)
for o > t
so that
(2.5 12)
We note for this case that the LTI system response at the time t is K times the integral
of the input over the previous T seconds.
To illustrate the evaluation of this integral, consider the case in which the input is
the rectangular pulse x(t) = Ar(t/T):
y(t) = KA fIT
r(:) do (2.5 13)
To perform this integration, it is best to draw a sketch such as the one shown in Fig.
2.53. With such a sketch, the value of the integral for each value o f t can easily be
determined. For example, if t < 0, then the integral is over only negative values of 0.
The value of the integral is zero because, from the sketch, r ( o / T )= 0 for o < 0.
Thus y(t) = 0 for t < 0. Now consider the value o f t to be in the range 0 < t < T .
For t in this range, note that t  T < 0 and t > 0. Thus the value of the integral is the
shaded area shown in Fig. 2.54a. This area is equal to t so that y(t) = KAt for
0 < t < T . Now consider the value o f t to be in the range T < t < 2T. For t in this
range, note that 0 < t  T < T and t > T . Thus the value of the integral is the
shaded area shown in Fig. 2.54b. This area is T  (t  T) = 2T  t so that
40 17)
1
lJ
0 T
Fig. 2.53 Graph of Y(cT/T).
52 LINEAR TIMEINVARIANT (LTI) SYSTEMS
y(t) = KA(2T  t) for T < t < 2T. Lastly, consider the value o f t to be greater than
2T. For t in this range, note that t  T > T so that the integral is over values of (T
larger than T . The value of the integral is zero because, from the sketch, Y((T/T)= 0
for (T > T. Thus y(t) = 0 for t > 2T. Combining the expressions for y(t) in the
various ranges, we have
for t < 0
forO<t<T
(2.5 14)
t) for T < t < 2T
fort > 2T
A graph of the output, y(t), for the example is shown in Fig. 2.55.
Example 4 For our fourth example, consider an LTI system with the unitimpulse
response
0 t
Fig. 2.56 Graph of x(t) and h4(t) for example 4.
For this example, I have chosen x(t) = h4(t). A graph of this function is shown in
Fig. 2.56.
As in the previous examples, the system response, y(t), will be determined using
the convolution integral, Eq. (2.51). The variable of integration is a so that t is a
constant in the integral. Thus the value of the integral, y(t), is determined by first
choosing ranges of values of t to use for the determination of y(t). The ranges of
values of t to choose is determined, as was done in Example 3, by examining a
sketch of x(a)h4(t a) for various values of t . For this, a sketch of h4(t  0 ) is
required.
We first note from Fig. 2.56 that h4(t) = 0 for t < 0. Thus, in accordance with
our discussion of a function as a rule, we have that h4(t  a) = 0 when the value
of the argument ( t  a) < 0. For a given value o f t , this is for a > t as seen in Fig.
2.57. To determine the graph for a < t, we evaluate h4(t  a) for a = t  a, with
a > 0. This point on the a axis is shown in Fig. 2.57. For this value of a we have
that ( t  a) = [t  ( t  a)] = a so that h4(t  a) = h4(a). From Eq. (2.515), this is
equal to e'' as seen in Fig. 2.57. Note that the graph of h4(t  a) versus a can be
obtained by first folding the graph of h4(t) about t = 0 and placing the origin at the
point t on the a axis. It can be seen that for any h(t) the graph of h(t  a) versus a is
obtained by folding and shifting as described above. For this reason, mathematicians
often use the descriptive German word Faltung for convolution because Faltung in
German means a folding over. Now, from the convolution integral, Eq. (2.5l), for a
given value oft, y ( t ) is equal to the area under the curve x(a)h(t  a). This product is
obtained by multiplying the graph of x(a) shown in Fig. 2.58 and the graph of
h4(t  a) shown in Fig. 2.57 for a particular value o f t . As t increases, it is seen
from Fig. 2.57 that the graph of h4(t  o) moves to the right. Thus we begin by
considering the product curve x(a)h4(t  a) for a large negative value of t and
observe any changes of the product curve as the value o f t is increased. For our
I
ta t 6'
0 d
example, we note that the product curve is zero for t < 0. This is noted because then,
from Fig. 2.58, x((T) = 0 for (T < 0 so the product curve is zero for (T < 0; also, from
Fig. 2.57, h4(t  (T)= 0 for (T > 0 so the product curve is zero for 0 > 0. Because
the product curve is zero for all (T, the area under the product curve, which is y(t), is
zero for t < 0. We now consider positive values oft. If t > 0, then the product curve
is as shown in Fig. 2.59. The curve is zero for 0 < 0 because x(a) = 0 for (T < 0;
also the product curve is zero for 0 > t because h4(t  0 ) = 0 for 0 > t. For the
interval 0 < (T < t, the product curve is x(a)h4(t (T)= e@"e@('")  e@',which

is a constant because we are evaluating the convolution integral for a particular value
oft. Figure 2.59 is a graph of x(cr)h4(t  0)for t > 0. Now y(t) is equal to the area
under the product curve. From Fig. 2.59, this is the area of the rectangle, tepa'.
Collecting our results for y(t), we have
(2.517)
The above four examples illustrate some of the important techniques that can be
used to evaluate the convolution integral. In all the examples, the function h(t) was
folded and shifted to evaluate the convolution integral. However, the same result
0.36 
aY(0
Fig. 2.510 Graph of y(t).
would have been obtained if x ( t ) were folded and shifted instead of h(t). To show
this, we begin with the convolution integral, Eq. (2.5I), which is repeated below:
y(t) = 1
00
00
x(o)h(t  0 ) do (2.5 1)
y(t) = 100
oo
h(z)x(t  z) dz (2.5 19)
Note that the difference between Eq. (2.51) and Eq. (2.519) is that the roles of x(t)
and h(t) have been interchanged. Thus, the same result, y(t), would be obtained if
x(t), instead of h(t), were folded and shifted. This is an important result that can, at
times, be used to advantage in evaluating the convolution integral. To appreciate the
computational difference, you should evaluate y(t) in the four examples given above
by folding and shifting x ( t ) instead of h(t).
Note that the LTI system response to any input can be determined by use of the
convolution integral once the system unitimpulse response is known. It is in this
sense that the unitimpulse response, h(t), completely characterizes the LTI system
mapping of inputs to outputs. Consequently, h(t) is the fundamental function we
shall use to determine and study properties of LTI systems. An understanding of the
techniques and concepts discussed in this and the last section is important for this
development.
56 LINEAR TIMEINVARIANT (LTI) SYSTEMS
PROBLEMS
21 Use Eq. (2.11) to show that the response of a linear system to the input
22 A system with the input x(t) and corresponding response y(t) is composed of
an ideal amplifier with a gain equal to A which is connected in tandem with a
soft limiter with a characteristic shown in Fig. 1.410 in which xo = 5 and
yo = 10. The system input is a class of waveforms for which Ix(t)l < B. For
what values of A and B can the system be modeled as a linear system
irrespective of the order of the tandem connection? What is the model of the
linear system?
23 Show that the system described by Eq. (2.14) is homogeneous but is not
linear.
24 (a) For the input, x(t), and corresponding response, y(t), show that the
system shown in Fig. 2.12 is not a linear system if E # 0.
(b) To circumvent this difficulty, define the system input to be h ( t ) =
x l ( t )  x 2 ( t ) and the corresponding output to be Ay(t) = y l ( t )  y 2 ( t ) in
which x,(t) + y ,(t) and x2(t) + y 2 ( t ) . Show that the mapping of
h ( t ) + Ay(t) is a linear mapping. Thus note that we can analyze the
given system using linear analysis by changing the definition of the input
and the output.
(c) Now choose x2(t) = 0. Determine the expression for Ay(t) in terms of
xl(t) and the parameters of the given system. The analysis of linear
electronic circuits in which the dc supply voltages contribute to the circuit
output can be linearized by this procedure.
25 A digital computer can only store sequences of numbers. Thus, only sample
values of a continuous waveform can be stored on a digital computer. This is
accomplished by a sampler that samples the waveform every T seconds. For
the input waveform,f(t), the sampler output isf(nT), which are the values of
the waveform, f ( t ) , at the times t = nT for n = 0, f l , f 2 , . . . .
Show that a sampler is a linear, timevarying (LTV) system.
PROBLEMS 57
26 The input, x ( t ) , and response, y(t), of a given system are related by the
constant coefficient differential equation
28 In an experiment, the response of an LTI system to the input xl(t)= r(t) is
y l ( t )= sin(nt)u(t). Determine the system response, y2(t), to the input
x 2 ( t ) = r(t/2).
Use the basic properties of linearity and time invariance to determine and
sketch the system response, yh(t),to the input x,,(t) = sin(nt)r(t).
58 LINEAR TIMEINVARIANT (LTI) SYSTEMS
21 2 For the input xl(t) = u(t) of an LTI system, it is observed that the
corresponding output is y l ( t ) = r(t/2). The waveform x2(t) =
+
u(t) 2 CgP=l(l)"u(t n) is now used as the LTI system input. Sketch
x2(t) and determine the corresponding system response, y2(t).
216 For this problem, instead of defining 6,(t) as a rectangle, define it as the
triangle,
1[
h,(t) = E 1  ?Iu( 1 Y)
(a) Show that the resulting approximation of x(t), x,(t), connects the values
x(m) by straight lines. Thus show that Eq. (2.31) is valid so that d,(t)
defined in this problem also can be used as the basis for the definition of
an impulse.
(b) Note that d,(t) defined in this problem is once differentiable. Sketch &(t).
(c) Use the result of part b to show that, with this definition of S,(t), we have
that if x ( t ) is differentiable, then lim,+o$(t) = x'(t).
PROBLEMS 59
(d) Use the result of part c to show that, equivalent to Eq. (2.42), we have
This result is used in mechanics and field theory where $ ( t ) is called the
doublet.
218 An LTI system with the input x ( t ) and corresponding output y(t) is shown
below. Determine and sketch the unitimpulse response, h(t), of the given
system. Label all important amplitudes and times of h(t).
219 An LTI system with the input x ( t ) and corresponding output y(t) is shown
below. Determine and sketch the unitimpulse response, h(t), of the given
system. Label all important amplitudes and times of h(t).
60 LINEAR TIMEINVARIANT (LTI) SYSTEMS
220 An LTI system with the input x(t) and corresponding output y(t) is shown
below. Determine and sketch the unitimpulse response, h(t), of the given
system. Label all important amplitudes and times of h(t).
221 The unitimpulse response of an LTI system is h(t) = Ar(t), where A > 0.
Use convolution to determine the system response, y(t), to the input
0. Sketch y(t) and label the values and times of
224 The unit impulse response of an LTI system is h(t) = Ae"u(t). Use
convolution to determine the system response, y(t), to the input
x ( t ) = Be"u(t). Sketch y(t) and label the values and times of all maxima
and minima.
230 The unit impulse response of an LTI system is h(t) = B( 2 21' (>; .
r
Determine the system response, y(t), to the input x(t) = Ar(t).
231 Determine the unitimpulse response of the feedback system shown below.
232 Use convolution to determine the response of the feedback system of problem
231 for the following inputs:
CHAPTER 3
One way to determine system properties is to study the effect of connecting the
system in various ways. For this, we shall study the tandem connection of LTI
systems in this section. A tandem connection of two systems is one in which the
output of the first system is the input of the second system as shown in Fig. 3.11.
The two systems are also said to be connected in cascade. In system theory, the
connection of systems is considered not to affect the characteristics of the individual
systems. Thus, in the tandem connection shown in Fig. 3.11, the inputoutput
relation of system S, is not affected by the connection of system S,. Note that the
tandem connection of circuits may not satisfy this condition. For example, consider
the tandem connection of two resistor circuits as shown in Fig. 3.12. The output of
the first circuit is y(t) for the input x ( t ) , and the output of the second circuit is z(t) for
the input y(t). If the second circuit were not connected in tandem, then
= ~ R2 x(t) (3.11)
R , +R2
However, with the second circuit connected in tandem, the input resistance (R, R4) +
of the second circuit is in parallel with the resistor R2 of the first circuit so that the
value R2 in Eq. (3.11) must be replaced by
(3.12)
Thus the output of the first circuit, y(t), is affected by the tandem connection of the
second circuit. We discussed differences between circuit theory and system theory
concepts in Sections 1.3 and 2.1. This is another essential difference between circuit
63
64 PROPERTIES OF LTI SYSTEMS
S
.............................................................................
Sb
theory and system theory concepts. However, the two circuits of Fig. 3.12 could be
made to satisfy the system definition of a tandem connection by connecting a unity
gain isolation amplifier between the two circuits. This is sometimes done in circuit
design. Generally, before attempting to apply system concepts to a circuit, it is
important to first determine whether all the system theory definitions are valid for
the given circuit.
For our study of the tandem connection of systems, the first observation to make
concerning Fig. 3.11 is that the operation S contained within the dotted lines is a
system because it is a manytoone mapping of x ( t ) to z(t). The reason is that because
Sa is a system, it is a manytoone mapping of x(t) to y(t) and because &, is a system,
it also is a manytoone mapping of y(t) to z(t). Now, a manytoone mapping of a
manytoone mapping is a manytoone mapping, so that the tandem connection, S ,
is a manytoone mapping of x(t) to z(t). Thus we have shown that S is a system.
We now note that if Sa and Sb are timeinvariant (TI) systems (linear or
nonlinear), then S is a TI system. From our discussion in Section 1.2, we can
show that S is a TI system by showing that if for any input we have x(t) + z(t),
then for any time shift to we obtain x(t  to)+ z(t  to).The arrow is the shorthand
notation introduced in Section 2.1. To show this, we note in Fig. 3.11 that
x ( t  to)+ y(t  to) because we are given that Sa is a TI system and also
y(t  to) + z(t  to) because we are given that Sb is a TI system. Thus we have
shown that x(t  to) + y(t  to) + z(t  to), so that S is a TI system. Observe that
the converse is not necessarily true. That is, if S is TI, it is not necessarily true that Sa
and Sb are TI. As a simple example, consider the case for which Sa is a timevarying
ideal amplifier with the output y(t) = a(t)x(t) and Sb also is a timevarying ideal
amplifier with the output z(t) = b(t)y(t)in which a(t)b(t)= A , where A is a constant.
The output of the system S then is z(t) = b(t)a(t)x(t)= Ax(t) so that S then is an
ideal amplifier with constant gain.
Next, we show that if Sa and Sb are linear systems (timeinvariant or timevary
ing), then S is a linear system. In accordance with our discussion in Section 2.1, we
.................................... ............................................
show that S is a linear system by showing that if x,( t ) + z1( t )and x 2 ( t ) + z2(t) for
any inputs x, ( t ) and x,(t), then
for any complex constants C, and C,. This is shown by first noting that
because s
h is a linear system. Thus we have shown that S is linear because
Observe that the converse is not necessarily true. That is, if S is linear, it is not
necessarily true that S, and s b are linear. As a simple example, consider any case for
which Sa is nonlinear system for which an inverse exists (so that S, is a onetoone
mapping of x ( t ) to y ( t ) in accordance with our discussion in Section 1.1) and s b is
the inverse of S,. Then z(t) = x(t), so that S is a linear system while S, and S, are
nonlinear systems. A specific example is the case for which the output of S, is
y(t) = x3(t) and the output of s b is z(t) = b(t)]1’3, which is the principal cube root of
its input.
Because the tandem connection of two linear systems is a linear system and also
the tandem connection of two TI systems is a TI system, we conclude that the
tandem connection of two LTI systems is an LTI system. Our concern in this chapter
is only with LTI systems. Thus we shall only consider the case for which both Sa and
s b are LTI systems, so that S in Fig. 3.11 is an LTI system. In accordance with our
discussion in Section 2.3, the output of the system S, z(t), can then be expressed as
the convolution of its input, x(t), with its impulse response, h(t), as
y(t) = 1
00
w
x(o)h(t  ). do (3.13)
For convenience, we shall express the convolution integral, Eq. (3.13), using the
shorthand notation
The star indicates the convolution integral of the two functions. In this notation, note
that it is the second function, h(t), which is folded and shifted in the integration.
The tandem connection of two LTI systems with unitimpulse responses h,(t) and
h h ( t ) is shown in Fig. 3.13. In accordance with our discussion in Sections 2.3 and
66 PROPERTIES OF LTI SYSTEMS
h(t) ....._______._____.__.
,.......... ...................._._.____............... .
x(t) I
~ 4 hdt) h&)
izt
2.5, h,(t) and hb(t) completely characterize the two tandemconnected LTI systems.
Thus we should be able to express the unitimpulse response, h(t), of the tandem
connection in terms of only h,(t) and hb(t).To determine this relation, we’ll use the
result developed in Section 2.3 that z(t) = h(t) when x(t) = d ( t ) and the system is
initially at rest. Now, if x ( t ) = d(t), then y,(t) = h,(t). Because z(t) = y,(t) * hb(t),
we then have z(t) = h,(t) * hb(t) when x(t) = d(t). Consequently,
(3.15)
It was shown in Section 2.5 [see Eq. (2.519)] that the value of the convolution is not
changed if the roles of the two functions being convolved are interchanged. Thus the
convolution integral in Eq. (3.15) also can be written in the form
(3.16)
Equation (3.17) is a statement that the convolution operation denoted by the asterisk
is commutative. Now, the expression for h(t) in Eq. (3.16) would have been
obtained if the order of connecting the LTI systems in Fig. 3.13 were reversed as
shown in Fig. 3.14. Because the unitimpulse response of the tandem system in Fig.
3.13 is the same as that in Fig. 3.14, we conclude that the two systems have the
same inputoutput relation. That is, for the same input, x(t), they both have the same
output, z(t). We shall call this the commutativeproperty ofLTI systems because it is a
consequence of the commutative property of the convolution operation. Note,
’@’hb(t) 1 ha@
Zt
however, that the waveforms y,(t) and yb(t) are not the same because
y,(t) = x(t) * h,(t) while yb(t) = x(t) * hb(t). While the systems of Figs 3.13 and
3.14 theoretically have the same inputoutput relation, it may not be so in practice.
The reason is, as discussed in Section 2.1, that a physical system can, be modeled as
a linear system only if the amplitude of the system input is less than a certain value.
Thus, in determining the order to use in connecting two physical systems in tandem,
it is important to make certain that, for the order chosen, the maximum amplitude of
the waveform between the systems is within the range for which the second system
can be considered to be linear.
The commutative property of LTI systems described by Eq. (3.17) leads to some
important properties of LTI systems. The important property we shall show and
discuss in this section is that if x(t) + y(t) for a given LTI system with the unit
impulse response h(t), then x(t) * h,(t) + y ( ~*)h,(t). Also, if the LTI system is
modified so that its unitimpulse response is changed to h(t) * h,(t), then the
response of the modified system to the input x(t) is y(t)* h,(t).
To show these properties, consider the two systems shown in Fig. 3.21. Both
systems have the same input, x(t). By the commutative property of LTI systems, the
output, z(t), is the same for both systems. First, from Fig. 3.2la, the response of the
system with the unitimpulse response h(t) is y (t ) to the input x(t). Now the output,
z(t), in Fig. 3.2la is z(t) = y(t) * h,(t). Thus the output of the system with the unit
impulse response h(t) in Fig. 3.2lb also is z(t) = y(t) * h,(t); its input, however, is
noted to be y,(t) = x(t) * h,(t). We thus note for the LTI system with the unit
impulse response h(t) in Fig. 3.2lb that x(t) * h,(t) + y(t) * h,(t).
Figure 3.22 is a summary illustration of the results we have obtained. The basic
relation obtained from Fig. 3.2la is illustrated in Fig. 3.22a. Using Fig. 3.2lb, we
then obtained the relation shown in Fig. 3.22b. Also from Fig. 3.21, we obtain the
relation shown in Fig. 3.22c.
The results we just obtained imply some fundamental relations that will be
developed and examined in this and subsequent sections. We begin by considering
(a) ...............................................................................
I ...............................................................................
...............................................................................
i
I I
ic)
the case for which h,(t) = u(t). Then, in accordance with the result obtained in
Example 2 of Section 2.5 we have
and
Consequently, from the result summarized in Fig. 3.22b we have the result that if
+ y(t) for a given LTI system with the unitimpulse response h(t), then
x(t)
As a specific example of this result, consider the case for which the input is
The system response then is the unitimpulse response so that
x ( t ) = d(t).
y(t) = h(t). NOW,
d(o) do = [ 0
1{2
ift<O
if t = 0
ift>O
= u(t) (3.25)
3.2 A CONSEQUENCE OF THE COMMUTATIVE PROPERTY 69
In accordance with our discussion in Section 2.4, Eq. (3.25) was obtained by using
8 J t ) in the integral and then taking the limit of the result as E + 0. Thus, we have
from Eq. (3.24) that
~ ( t+
) f cc
h(a) d o (3.26)
For convenience, we call a system response to a unit step, s(t). We then have from
Eq. (3.26) that, for an LTI system,
(3.27)
d
s’(t) =  s ( t ) = h(t) (3.28)
dt
This result suggests a practical method for experimentally determining the unit
impulse response, h(t), of an LTI system. Normally, it is not practical to determine
h(t) directly. To make such a measurement, the input would have to be a very narrow
pulse. In accordance with our discussion in Section 2.4, if the input were
x(t) = A6,(t), then y(t) M Ah(t) at those values of t for which h(t) can be well
approximated by a straight line within the &region about the value of t. To be
specific, consider a case in which the graph of h(t) contains a sinusoidal wiggle
in which the frequency of the sinusoid is 10 kHz. As discussed in Section 1.4, the
sinusoid fundamental period is 100 ps. For y(t) x Ah(t), we would have to choose E
to be much less than 100 ps. Let us choose to be 10 ps. Thus the amplitude of the
input pulse, Ad,(t), is A / &= 105A. As we’ve discussed, no physical system is truly
linear; it can be modeled as a linear system only if the maximum input amplitude is
less than a certain value. Let that value be one for our example. We then would
require A / & d 1. The largest possible value of A we then could choose is A = 1 Ow5.
With such a choice of values for x(t), the system response would be y(t) x 10w5h(t).
Note that the maximum amplitude of y(t) would be rather small. The values I’ve
used are really not unreasonable at all. Now, in practice, noise (which is just some
random fluctuation) is everpresent. Its presence limits what can be done in practice.
(For example, why will a pencil not stand on its point for very long?) It is noise that
prevents the practical implementation of many seemingly reasonable ideas. We’ll not
discuss noise in this text except to note some of the limitations it imposes. For our
example, the presence of noise would make it difficult to observe y(t) because of its
small amplitude. Consequently, except in special cases, the determination of h(t) by
applying a narrow pulse to the system is not a practical method. However, a practical
method suggested by Eq. (3.28) is to apply a unit step to the LTI system to obtain
s ( t ) experimentally. Then h(t) can be determined by differentiating s(t), which is
accomplished experimentally by plotting the slope of s(t) versus t. There are other
70 PROPERTIES OF LTI SYSTEMS
Fig. 3.23
(3.29b)
(3.29d)
and
These equations could be combined into one large equation for the output, ~ ( t )in,
terms of the input, x(t). Even though such an equation might appear impressive, it is
more difficult to work with than the five simple component relations in Eq. (3.29).
We thus will work with the component equations directly.
First, to determine h(t), we make use of the fact that ~ ( t=) h(t) when x(t) = S(t).
Then, from Eq. (3.29a) we have
Y I ( ~ )= f
co
zl(u) do = f
co
[6(o to) 6(a  2t0)] do (3.2 1Ob)
3.2 A CONSEQUENCE OF THE COMMUTATIVE PROPERTY 71
+(v
Fig. 3.24
The value of y l ( t ) is the area under zl(o) in the interval co < o d t. From the
graph of zl(o) in Fig. 3.24, we observe that the area is zero if t < to. For
to < t < 2t0 we observe that the area is equal to that of the impulse, which is
one. For t > 2t0, the value of the integral is equal to the sum of the areas of the
two impulses, which is 1  1 = 0. We thus have
y,(t) =
I 0 for t < to
1 for to < t < 2t0
0 fort > 2t0
(3.21 la)
Using the notation established in Section 1.5, this result can be expressed more
compactly as
(3.21 lb)
We now determine y2(t). For this determination, we have from Eq. (3.29c) that
y2(t) = f
03
z2(o) do = [6(o  3t0)  h(o  2to)l do (3.212b)
The value of y2(t) is the area under z2(o) is the interval oo < o d t. From the
graph of z2(rJ) in Fig. 3.25, we observe that the area is zero if t < 2t0. For
2t0 < t < 3t0, we observe that the area is equal to that of the impulse which is
Fig. 3.25
72 PROPERTIES OF LTI SYSTEMS
minus one. For t > 3t0,the value of the integral is equal to the sum of the areas of the
two impulses which is 1 + 1 = 0. Thus we have
y2(t)=
I 0
1
0
fort < 2t0
for 2t0 < t < 3t0
fort > 3t0
(3.2 13a)
Using the notation established in Section 1.5, this result can be expressed more
compactly as
(3.2 13b)
Finally, y(t) = h(t) because x(t) = s(t), so that from Eq. (3.29e) we have
(3.2 14)
(0 for t d to
for to Q t Q 2t0
(3.215)
to  (t  2t0) = 3t0  t for 2t0 d t d 3t0
for t 2 3t0
Fig. 3.26
3.2 A CONSEQUENCE OF THE COMMUTATIVE PROPERTY 73
Fig. 3.27
to use the basic property of linear systems, which is generalized superposition. For
this, observe that x ( t ) can be expressed as the linear combination of two functions as
(e)
x ( t ) = Y  = u(t)  u(t  to) (3.216)
From generalized superposition, the system response, y(t), to the given input can be
expressed as the system response to u(t) minus the system response to u(t  to).The
system response to u(t) is s(t) given above. Using the fact that the given system is
timeinvariant, the system response to u(t  to)is s(t  to).Thus the system response
to the given input is
With the use of the graph of s(t) given in Fig. 3.27, we obtain the graph of y(t)
shown in Fig. 3.28 with the equation given in Eq. (3.218).
I
0 for t d to
t  to for to < t d 2t0
y(t) = 5t0  2t for 2t0 d t d 3t0 (3.218)
t  4t0 for 3t0 d t d 4t0
0 for t 2 4t0
Note how much insight was obtained concerning the system response to various
inputs and how much effort was saved by making use of the basic properties of LTI
systems we have discussed.
Fig. 3.28
74 PROPERTIES OF LTI SYSTEMS
Our development of the unit impulse in Sections 2.3 and 2.4 has been in terms of a
rectangular pulse with a width equal to E and an area equal to one which we called
d,(t). Using the results obtained in the last section, we can show that d,(t) really
could be defined as any nonnegative pulse with an area equal to one which is
symmetric about t = 0 and with a width that goes to zero as E goes to zero. We
used the rectangle because its use simplified our discussions in the previous sections.
Other forms of the pulse shape are usehl (e.g., Problem 216) and so we shall
develop this generalization in this section.
To begin, the hdamental defining property of the unit impulse is Eq. (2.34):
where
In accordance with Eq. (2.4l), note that this must be true for any waveform, not just
for h(t). Now define u,(t) to be the integral of s,(t),
(3.33)
(3.34)
Then, in accordance with the result obtained in the last section, we obtain
Now, instead of just a rectangle, let d,(t) be defined as any nonnegative pulse with an
area equal to one which is symmetric about t = 0 and a width that goes to zero as E
goes to zero. Then
42 o r2 t
For example, if d,(t) were the rectangular pulse as defined in Sections 2.3 and 2.4,
then
E
fort < 
2
u,(t) =
,s' 6,(0) do =
E + 2t
2E
E
for   < t < 
2
for t > 
E
E
2
(3.37)
A graph of u,(t) as given by Eq. (3.37) is shown in Fig. 3.31. It is clear from Fig.
3.31 that Eq. (3.36) is satisfied for this example.
Note that, in general, the total rise of u,(t) is equal to the area of s,(t). If this area
is one, then the total rise is one as shown for our example. Also, if d,(t) is a positive
pulse, then uc(t) increases monotonically from a value of zero to a value of one.
Also, u,(O) = 1/2 because the pulse is symmetric about t = 0. Furthermore, the rise
time of uB(t)is equal to the width of d,(t). Because this width goes to zero as E goes
to zero, we have that the rise time of uE(t)goes to zero as E goes to zero. We thus note
that Eq. (3.36) is satisfied if s,(t) is any nonnegative pulse with an area equal to one
which is symmetric about t = 0 and with a width that goes to zero as E goes to zero.
Now, from Section 3.2, the LTI system response to u(t) is s(t). As in Section 2.3,
we assume the LTI system is continuous.' Then, from Eqs. (3.35) and (3.36) we
have
where s,(t) is given by Eq. (3.34) and s(t), from Eq. (3.27), is
s(t) =
SI, h(O) do (3.39)
By differentiation, we then have from Eqs. (3.38), (3.39), and (3.34) that
But this is Eq. (3.3l), which is the fundamental defining property of the unit
impulse. We thus observe that 6,(t) can be any symmetric nonnegative pulse with
an area equal to one and a width that goes to zero as E goes to zero.
An important example of such a pulse is the normal pulse, which is defined as
(3.31 1)
It is a bellshaped pulse with an area equal to one whose width goes to zero as E goes
to zero. Thus it satisfies all the requirements discussed above so that it can be used in
place of the rectangular pulse for 6,(t).
One other useful result that follows from our discussion above is obtained from
Eq. (3.36). Although the derivative of u(t) does not formally exist, we can define it
by differentiation from Eq. (3.36) and Eq. (3.33):
But the limit on the righthand side of this equation is the unit impulse. Thus we can
define
With this result, we can define the derivative of a discontinuous function. For
example, consider the function shown in Fig. 3.32. The function f ( t ) shown is
discontinuous at t = to. Because it makes a jump of ( B  A ) there, we could express
it as the sum of a continuous function, g(t), and a step function as
where g(t) isf(t) with the jump at t = to removed; the jump at t = to is expressed by
the step function. Now, differentiating Eq. (3.314) and using the result given by Eq.
(3.313), we have
>f ,
Fig. 3.32 Example of a discontinuous function.
3.4 CONVOLUTION REVISITED 77
The results obtained in the last two sections can be used to simplify many calcula
tions. Also, the simplifications obtained often can be used to gain a better under
standing of the equations involved. Some of these simplifications will be illustrated
in this section.
Let us consider the following convolution:
(3.42)
where
(3.43a)
and
~ ( t=) J’
cc
do
~’(0) (3.43b)
We thus note that, instead of performing the convolution of h(t) and x ( t ) to obtain
y ( t ) , we could convolve h(t) and x’(t) to obtain j ( t ) and then integrate j ( t ) in
accordance with Eq. (3.43a) to obtain y(t). In some cases, this differentiation
procedure leads to a great simplification of the calculation.
As an illustration, consider the system with the input x(t) and unitimpulse
response h(t) shown in Fig. 3.41. The output, y(t), could be calculated directly
with the use of Eq. (3.42). For this example, however, it is simpler to determine
y(t) by the differentiation procedure described above. For this we differentiate x(t)
using the result given by Eq. (3.313).
Observe that x(t) can be expressed in the form
Now, fiom the result of Example 1 of Section 2.5 and using the commutative
property obtained in Section 3.1, we have
A graph of y’(t) shown in Fig. 3.43 is now easily obtained using the graph of h(t)
given in Fig. 3.41.
We now integrate y’(t) in accordance with Eq. (3.43a) to obtain y(t). This is
easily done graphically because y(t) is just the area under y’(o) in the interval
m < r s < t . A graph of y(t) is shown in Fig. 3.44.
3.4 CONVOLUTION REVISITED 79
To realize how much effort was saved in this example, it would be worthwhile for
you to determine y(t) by actually convolving h(t) with x ( t ) . The advantage of this
technique is that convolution was reduced to the convolution with impulses, which is
particularly simple. When differentiation does provide such a reduction, the proce
dure just illustrated is worth considering. Also note that we convolved x‘(t) and h(t)
in our example. Notice, however, that we would have obtained the same result if we
had convolved h’(t) and x ( t ) instead. Thus the function to choose to differentiate is,
of course, the one that results in the simplest calculation.
The differentiation technique also can be used in some cases to obtain a differ
ential equation that relates the LTI system response, y(t), and the system input, x ( t ) .
To illustrate this technique, consider an LTI system with the unitimpulse response
To obtain this result, note that h(t) is discontinuous at t = 0 with h(0) = 0 and
h(O+) = A . Now, by substituting Eq. (3.49) into Eq. (3.41 l), we obtain
We thus have obtained a differential equation which h(t) must satisfy. Now by
substituting Eq. (3.412a) in Eq. (3.4lo), we obtain
Because x ( t ) = x(t) * d ( t ) and y(t) = h(t) * x(t), we obtain from Eq. (3.413) the
differential equation
or, by rearranging terms, the desired differential equation relating the output, y(t),
and input, x(t), is
It should be noted that in the solution of Eq. (3.414b), the following condition must
be used: If x(t) = 0 for t < to , then y(t) = 0 for t < to. This condition follows from
the fact that h(t) = 0 for t < 0, as seen from Eq. (3.49).
If the system input, x(t), is the unit impulse, d(t), then the system response, y(t), is
the unitimpulse response, h(t). If this is substituted in Eq. (3.414b), we obtain Eq.
(3.412b). The unitimpulse response, h(t), is called the fundamental solution of Eq.
(3.414b) because it is the solution of the differential equation when the input is a
unit impulse. Note that h(t) is not the homogeneous solution because y(t) is defined
to be the homogeneous solution when the input, x(t), is zero for all time.
3.5 CAUSALITY
Causality is a concept that there is relation between a cause and an effect in which
the cause precedes the effect. It is presently believed that all physical systems are
causal. This seemingly simple concept is not always obvious. For example, a
problem with which the ancient Greeks grappled was the following: Achilles ran
a foot race and received a prize for winning the race. Now, it was argued, it was the
future event of receiving the prize which caused Achilles to run and win the race,
and so this is an example of noncausality. Those who argued that this is an example
of noncausality missed the crucial point that Achilles only ran and won the race
because he was told before the race that there would be a prize waiting for the winner
at the finish line and Achilles believed it. It is possible that a prize was not really to
be awarded and that Achilles was misled. So the real cause of Achilles running and
winning the race was not the prize but rather the belief he held before running the
race that a prize would be awarded the winner of the race. This example, although
3.5 CAUSALITY 81
simple, is sufficient to show that the causal relation can be subtle. The subtleties of
causality have been explored by many philosophers. A good classic discussion of
this topic is contained in Treatise of Human Nature by David Hume.2
For systems in which the system input and output are functions of time, a system
is dejined to be causal iJ; at any time, the output does not depend on future values of
the input.There is nothing in physics that requires all physical systems to be causal.
That is, there is no concept in basic physics that requires the relation between the
input and output of every physical system to be causal. Rather, it suits our social
philosophy to believe that all physical systems are causal, and this belief is rein
forced by the fact that every system observed to date can be modeled as a causal
system. If a noncausal system could be constructed, then the output at any time
would contain information about the future of the input so that the system could be
used to predict at least some of the future of its input. Thus, if the input were your
speech waveform, then the system could be used to predict some of what you will
say in the future. But that would imply you do not have the free will to say whatever
you want and whenever you want in the future! Thus, the belief in noncausal
physical systems leads to fundamental questions about the existence of free will.
Without free will, such desirable social concepts as ethics and morality become
questionable. We thus believe in free will, and this belief leads us to assume that
all physical systems are causal. As I stated above, every system observed to date can
be modeled as a causal system so that a counterexample to our belief does not seem
to exist. This reinforces our belief that all physical systems are causal. Although we
cannot prove that all physical systems are causal, we shall prove in our discussion of
passive systems in Section 8.4 that all passive linear systems must be causal.
Causality is important in system theory for two main reasons. The first stems
from the desire to know the constraints causality imposes on a given system so that
one can know if a given theoretical model can be the model of a physical system.
The second reason stems from a desire to know the theoretical best that can be done
in certain situations by a noncausal system. There are limitations of the best that can
be done in many problems in communication theory such as filtering noise from a
signal and in control theory such as controlling a given system by designing a
system to be placed in feedback. One of the sources of the limitation is due to
the requirement that all the designed systems be causal. It is then of interest to
know whether the performance could be improved significantly if the causality
constraint were removed and, if so, in what manner was the future of the input
used to obtain the improvement. With the understanding that derives from such
knowledge, new strategies can sometimes be devised to improve performance
with the use of causal systems.
We have seen that the unitimpulse response, h(t), completely characterizes the
inputoutput mapping of an LTI system. Thus whatever constraints causality
imposes on an LTI system must be reflected in a constraint on its unitimpulse
response. For this determination we require a more formal definition of causality.
We defined a system to be causal if, at any time, the output does not depend on
'David Hume was a Scottish philosopher who lived from 171 1 to 1776.
82 PROPERTIES OF LTI SYSTEMS
future values of the input. Let to be some arbitrary time. Our definition of causality
then can be translated into the statement that a system is causal iJ; for any value of to,
y(to)does not depend on x(t) for t > to.
To determine the constraint causality imposes on h(f), we begin with the expres
sion for y(to) obtained from the convolution integral:
(3.5 1a)
The integral over all a in Eq. (3.5la) has been expressed in Eq. (3.5lb) as an
integral for a < to plus an integral for a > to.The reason for expressing the integral
in this manner is that x(a) for a > to are future values of the input as discussed
above, while x(a) for a < to are past and present values of the input. Thus the first
integral in Eq. (3.5lb) involves only past and present values of the input, while the
second integral involves only fbture values of the input. If the output is not to depend
on future values of the input, then the value of the second integral must be zero for
any input. We thus note that the LTI system is causal if and only if
(3.52)
for any input, x(t). This can be satisfied if and only if h(t,  a) = 0 for a > to. First,
the restriction that a > to is because the integration is only over that range of a.
Now, it is clear that if h(t,  a) = 0 for a > to, then I , the value of the integral in
Eq. (3.52), is zero for any input, x(t). To see that I is zero only if h(to  a) = 0 for
a > to,assume that this condition is not satisfied. Then because the integral must be
zero for any input, we choose the input to be
x ( 4 = sgn[Wo  4 1
1 if h(to  a) > 0 (3.53)
1 if h(to  a) < 0
where sgn[.] is the signum function defined by Eq. (1.429). With this choice of the
input, we note that x(a)h(to a) = Ih(to  a)[ so that
(3.54)
The value of an integral is just the area under the function being integrated. Because
the function being integrated in Eq. (3.54) is never negative, the value of the
integral, I , is zero only if the function being integrated is zero. We thus have
3.5 CAUSALITY 83
shown that Z = 0 for any input, x(t), if and only if h(to  a) = 0 for rs > to. Note that
this condition is that the unitimpulse response be zero if its argument is negative
because to  a < 0 for (T > to. Thus we have shown the following:
Remember that h(t) is the LTI system response to the unit impulse, d(t),so that t = 0
for h(t) is the instant the unit impulse is applied. In a sense, the causality condition
for an LTI system is that the system cannot scream before it is kicked. It is clear that
this condition is necessary even for systems that are not LTI because if the system
started to scream, say one second before it is kicked, then it is predicting that it will
be kicked one second later. This, as I discussed above, implies that one does not have
the free will to kick it whenever desired. We have shown that this condition also is
sufficient to ensure that an LTI system is causal. If a system is not LTI, then this
condition is not sufficient. To see this, consider the nonlinear noncausal system
shown in Fig. 3.51. As seen from the diagram, the system output is
so that the system is noncausal if T > 0. However, we observe from Eq. (3.55) that
y(t) = 0 for t < to if x(t) = 0 for t < to, so that the noncausal system depicted will
not scream before it is kicked. Thus we note that the condition that a system not
scream before being kicked is a necessary condition for any system (linear or
nonlinear, timeinvariant or timevarying) to be causal. However, if the system is
LTI, then it is both a necessary and sufficient condition for the LTI system to be
causal. Another useful expression of this condition is as follows: An LTZ system is
causal if and only if for any value of to and any input that is zero for t < to (that is,
x(t) = Ofor t < to), we obtain an output, y(t),for which y(t) = 0 for t < to. Note
again that this statement holds only for LTI systems.
In terms of our discussion above, we can give a more physical interpretation of
the unitimpulse response. Consider first the system described in Example 3 of
Section 2.5. The LTI system unitimpulse response given by Eq. (2.59) is
h(t) = Kr(t/T), and the system response given by Eq. (2.512) is
The output at any time, t, is K times the integral of the input over the last T seconds.
If K = 1/T, then the output at a time t is the average of the input over the last T
seconds. For our example, note that values of the input more than T seconds in the
0 1
past have no effect on the output. In a sense, we can say that the system only
remembers the last T seconds of the input.
In general, for a causal LTI system, the unitimpulse response is simply a graph of
the weighting over the past of the input used to produce the output. To see this more
clearly, consider the causal LTI system with the unitimpulse response
and shown in Fig. 3.52. This is the unitimpulse response of the LTI system of
Example 4 in Section 2.5. The output at a time, to, is given by
(3.58)
A graph of h(to  a) versus c is shown in Fig. 3.53. From Eq. (3.58),this is seen to
be the weighting of the past of the input used to produce the output at the time to. We
see from Fig. 3.53 that, as an input value recedes into the past, its influence on the
output decreases exponentially. In a sense, we can say that the system has a memory
that decays exponentially in time. In this sense, we can interpret the unitimpulse
response of an LTI system as a graph of the system memory. With this view, an LTI
system with the unitimpulse response h(t) = u(t) has infinite memory.
3.6 STABILITY
There is no one definition of system stability. The reason is that the stability of a
system is considered relative to a particular concern about the system. Thus the
definition of stability used is one that is meaningful relative to the particular concern.
Thus different concerns dictate different definitions. For example, consider a ball
that is at rest at the bottom inside a bowl. If the ball is hit with a small force, the ball
will just roll up the side of the bowl a bit, roll about the bowl, and finally settle back
at the bottom of the bowl. On the other hand, if the ball is hit hard, the ball will roll
up the side and over the edge of the bowl, never to return. For this situation, a
meaningful definition of stability is a local one in which the system is considered to
be stable if the ball will eventually return to the bottom of the bowl if it is initially
perturbed less than a certain amount. This system is then said to be locally asymp
totically stable because it eventually will return to its initial quiescent state. With this
definition, the system would be unstable if the bowl were turned upside down and
the ball placed on top of the bowl. The mathematical theory for the study of
asymptotic stability is called stability in the sense of Lyapunov (often abbreviated
as stability i . ~ . i . )It. ~is used in discussing many autonomous systems such as the
stability of an electron in orbit about an atomic nucleus and also the stability of a
planet in orbit about the sun. The theory for i.s.1. stability is developed using the
statespace description of a system discussed in Section 10.6. However, stability i.s.1.
is not very useful in the discussion of nonautonomous systems which is our major
concern in this text. For a definition of stability to be useful, it must not only be
meaningful relative to our concerns, but it also must be useful. This requires that the
definition leads to analytical techniques that can be used without undue effort. Of the
various possible definitions for nonautonomous systems, the definition of stability
that we shall use is the BIBO stability criterion:
A system is defined to be stable in accordance with the BIBO stability criterion if the
system response to any bounded input waveform is a bounded output waveform (hence
the abbreviation BIBO).
+
1. The waveform f i ( t ) = A cos(ot 4) is a bounded waveform because
[AI < f i ( t ) < IAl or, equivalently, Ifi(t)l d IAl.
2. For a # 0, the waveform h(t)= d' is not a bounded waveform because
f;(t) + 00 as t + 00 if a > 0 andh(t) + 00 as t + 00 if a < 0.
Named in honor of the Russian mathematician A. M. Lyapunov, who first published his studies in 1892.
Since then, this definition of stability has been extensively studied. Many texts and articles in the
cngincering and mathematical journals have been published on Lyapunov stability.
86 PROPERTIES OF LTI SYSTEMS
Observe that the BIBO stability criterion only requires that the system response to
any bounded input waveform be a bounded output waveform. The system is not
BIBOstable if only one bounded input waveform results in an output that is
unbounded at just one instant of time. Thus a system is or is not BIBOstable; a
system cannot be conditionally BIBOstable. Also note that there is no specification
of the system response to an unbounded input waveform. There is only a specifica
tion of the system response to a bounded input waveform. In a sense, a system is
BIBOstable if it is not explosive. That is, an unbounded response is not obtained for
a bounded input as occurs in an explosion.
Because the unitimpulse response, h(t), of an LTI system completely determines
its inputoutput mapping, we expect that the required conditions for the BIBO
stability of an LTI system can be specified in terms of required conditions on
h(t). This indeed is the case, and we shall show that an LTI system is BIBOstable
ifand only i f
(3.61)
For this proof, we need the basic inequality that the magnitude of the area under a
curve is not greater than the area under its magnitudethat is,
(3.62)
We also need the basic identity shown in Appendix A, Eq. (A26), that the magni
tude of a product of two quantities is equal to the product of their magnitudes:
For our proof, we begin with the basic inputoutput relation for an LTI system:
y(t) = 100
00
h(z)x(t Z) dz (3.64)
3.6 STABILITY 87
We first show that if Eq. (3.61) is satisfied, then the response to any bounded
input is a bounded output. If the input is bounded so that Ix(t)l d M , then from
Eqs. (3.62), (3.63), and (3.64) we obtain
(3.65)
Thus, if Eq. (3.61) is satisfied, then ~ ( tis) a bounded waveform so that the response
to every bounded input waveform is a bounded waveform.
We now must show that the LTI system is BIBOstable only if Eq. (3.61) is
satisfied. That is, we must show that if Eq. (3.61) is not satisfied, then the LTI
system is not BIBOstable. To show this, we need to show that if Eq. (3.61) is not
satisfied, then there exists at least one input waveform for which the magnitude of
the output waveform becomes infinite at least at one instant of time so that it is
not a bounded waveform. For this, we assume that Eq. (3.61) is not satisfied
and we choose a bounded input for which y(t) becomes infinite at t = 0. First,
from Eq. (3.64), the output at t = 0 is
y(0) = 1 00
'x
h(z)x(r) dz (3.67)
x(t) = sgn[h(t)]
x(t) =
I 1
0
1
if h(t) > 0
if h(t) = 0
if h(t) < 0
The input chosen clearly is a bounded waveform because its magnitude never
(3.68)
exceeds one. With this choice of input, we have from Eq. (3.67)
(3.69)
88 PROPERTIES OF LTI SYSTEMS
so that the output at t = 0 would be infinite if Eq. (3.61)is not satisfied. We have
thus shown that an LTI system is BIBOstable if and only if Eq. (3.61)is satisfied.
Let us apply Eq. (3.61)to determine the BIBOstability of some LTI systems.
For the first example, consider an LTI system with the unitimpulse response
For this system, we have Ihl(t)l = IA6(t  to)[= IA16(t  to) because, from our
definition of the unit impulse, we have 6(t) 3 0. Thus, from Eq. (3.61)we have
100
cc
00
Ihl(t)l dt = IAl1
cc
6(t  to)dt = IAl < 00 (3.611)
so that the LTI system is BIBOstable. Note that the BIB0 criterion does not require
h(t) to be a bounded waveform; it only requires that the area under its magnitude be
jnite. The BIBOstability of the LTI system should, of course, have been obvious
because, from Example 1 of Section 2.5, the response, y l ( t ) of the system is
so that
Thus the LTI system with the unitimpulse response h , ( t ) given by Eq. (3.610)is
BIBOstable and causal if to 3 0; it is BIBOstable and noncausal if to < 0.
As a second example, consider a causal LTI system with the unitimpulse
response
where a is a real number. It would be difficult to apply the criterion, Eq. (3.6l),
because the resulting integral is not easily evaluated. The criterion, however, does
not require the evaluation of the integral. Rather, we need only show that its value is
less than or equal to some positive number. For this, we note that because
Icos(8)I < 1, we have
3.6 STABILITY 89
The last inequality is obtained by noting that the exponential and the unit step are
never negative so that each is equal to its magnitude. Thus
00
Ih2(t)ldtd [AI lo
00
e@'dt (3.6 1 6)
This integral is finite only if a > 0, for which the value of the integral is IAl/a. We
thus note that this LTI system is BIBOstable if a > 0.
The stability proof given above cannot be used for a < 0. The reason is that Eq.
(3.616) is just an upper bound to the value of Eq. (3.61). Thus, if the upper bound
in Eq. (3.616) is infinite, the value of Eq. (3.61) could be finite or infinite. To
examine the case for which a < 0, we need to obtain a lower bound of h2(t).For this,
first consider the case for which a = 0. For this case,
so that
M
Ih2(t)l dt = IAl loI
00
The value of the integral is infinite because the cosine is a periodic waveform so that
its magnitude is a periodic waveform that is never negative. Thus the value of the
+
integral is the area under the curve, I cos(oOt 4)1, over one period times the
number of periods of the waveform. The area under the curve over one period is
a finite positive number, but there are an infinite number of periods in the interval
0 < t < 00 so that the value of the integral is infinite. The LTI system thus is not
BIBOstable if a = 0.
The case for which c1 < 0 can now be resolved because for this case a lower
bound is easily obtained. Note that for a < 0
Thus we have shown that the LTI system with the unitimpulse response h2(t) given
by Eq. (3.614) is BIBOstable only if a > 0. The frequency ooand the phase 4 do
not aftect the BIBOstability of the system. For the special case in which oo= 0 and
4 = 0, we have that the causal LTI system with the unitimpulse response
h(t) = Ae@'u(t) (3.621)
90 PROPERTIES OF LTI SYSTEMS
is BIBOstable only if CI > 0. It is not stable, for example, if a = 0. For the case
a = 0, the unitimpulse response is a step function so that the causal LTI system with
the unitimpulse response h(t) = Au(t) is not a BIBOstable system. Note that this
system has, in terms of our discussion in Section 3.5, infinite memory. Observe that
a BIBOstable system cannot have infinite memory because then the stability condi
tion given by Eq. (3.61) would not be satisfied.
As we have seen, the BIBOstability of an LTI system can be determined by use
of Eq. (3.61) if the unitimpulse response, h(t), is known. If the unitimpulse
response is not known or if the system is not LTI, then one must resort to the
basic definition of BIBOstability to determine whether the system is BIBO
stable. In this latter case, one must show either that the system is BIBOstable by
showing that every bounded input results in a bounded output or show that the
system is not BIBOstable by showing that there is at least one bounded input for
which the output is unbounded. For example, consider the LTI system for which
d
y ( t ) = x(t). In accordance with the result obtained in Section 3.3, Eq. (3.313), the
dt
response of this system to the bounded input x(t) = Au(t) is the unbounded output
y ( t ) = Ad(t) so that the ideal differentiator is not a BIBOstable system.
The derivation in Section 2.3 of the convolution integral, Eq. (2.34), and also our
discussion in Section 3.3 required that the LTI system be a continuous system. That
is, we had to require that if xl(t) + y l ( t ) and x2(t)+ y 2 ( t ) , then the difference
between the two outputs, b2(t) y l ( t ) ] ,goes to zero as the difference between the
two inputs, [x2(t) x l ( t ) ] ,goes to zero. We shall show in this section that an LTZ
system is a continuous system if and only if it is BIBOstable.
We begin by defining the waveform differences:
(3.71)
and
Then, by the superposition property of linear systems, we have that AJt) is the
system response to the input Ax(t). That is,
(3.73)
3.7 SYSTEM CONTINUITY 91
Because we are considering the case in which A,(?) goes to zero, we let it be a
bounded waveform with IAx(t)l < M, in which M, goes to zero as A,(t) goes to zero.
Then, similar to Eq. (3.66), we have that
(3.74)
(3.75)
so that
We thus note that 1AJt)I goes to zero as M, goes to zero. That is, we have shown that
the difference in the outputs, AJt), goes to zero as the difference in the inputs, Ax(t),
goes to zero if the LTI system is BIBOstable.
To show that this is true only if the LTI system is BIBOstable, we must show that
if the LTI system is not BIBOstable, then there exists some input x2(t) such that the
difference A,(?) goes to zero and yet the output difference, AJt), does not go to zero
at least at one instant of time. For this we choose the time instant to be t = 0 so that
00
(3.77)
(3.78)
1 if h(t) > 0
g(t) = sgn[h(t)] = (3.79)
1 if h(t) < 0
(3.71 1)
92 PROPERTIES OF LTI SYSTEMS
If the LTI system is not BIBOstable, the integral in Eq. (3.711) diverges and so
AJO) is infinite no matter how small is E so that AJO) does not go to zero as A,(t)
goes to zero. Thus we have shown that an LTI system is continuous if and only if it is
BIBOstable.
As an illustration, consider the ideal integrator, which was discussed as Example
2 in Section 2.5. The unitimpulse response of this LTI system is h(t) = Ku(t); and
from Eq. (2.58), its response, y(t), to the input x ( t ) is
We showed near the end of Section 3.6 that this system is not BIBOstable so that, in
accordance with our result in this section, it is not a continuous system. This can be
illustrated for this system by considering the input
lo otherwise
f,
til 1
e" dt = [eato 11 < 00
CI
(3.714)
Normally for causal physical systems, as in the example above, we would have that
for any finite value of to
(3.715)
3.8 THE POTENTIAL INTEGRAL 93
Now, in accordance with the convolution integral, the output at the time t = to of a
causal physical system due to an input, x(t), which is zero for t < 0, is
y(to) = 100
00
h(z)x(to z) d z = (3.7 16)
The lower limit of the integral is zero because h(t) = 0 for t < 0 and the upper limit
of the integral is to because, for the class of inputs we are considering, x(to  z) = 0
for z > to. Thus the system response at t = to involves h(t) only for t d to. Thus, if
we define
= { h(t) for t d to
0 for t > to
(3.7 17)
(3.7 18)
and the system with the unit impulse response ho(t)is BIBOstable because
100
J oo
Iho(t)ldt =
J
t0
[
aJ
Ih(t)l dt = [
t0
Jo
Ih(t)l dt < 00 (3.7 19)
In this manner the response at any finite time, to, of an unstable causal physical
system to any input that is zero for t < 0 can be considered to be the response of a
causal and stable LTI system, and so the convolution integral is valid for any finite
value o f t .
Our specific interest in this text is LTI systems. However, with a slight change of
viewpoint, the theory of LTI systems can be applied to many different fields of study.
Often, the concepts and theories developed for a given field of study are found, with
some small modifications, useful for other fields of study. This transference of
concepts and theories from one scientific field to another is very powerful because
it unifies many fields of scientific study and the different viewpoint often results in
new insights. This is one of the reasons why an individual should attempt to be
educated in several fields of study. An individual with such an education can then
use the concepts and approaches in one field of study to develop new ways of
thinking in another field of study. However, this can be accomplished only with a
basic understanding of the concepts and theories developed in the given field. It is
not sufficient to just know how to solve problems in the given field using the derived
94 PROPERTIES OF LTI SYSTEMS
formulae. This is one of the reasons why a basic discussion of the concepts and
theories of LTI systems are presented in this text.
As an illustration, we shall, with a slight change of viewpoint, use the concepts
and the theory we have developed for the study of LTI systems to analyze the
potential distribution in free space due to a given charge density distribution. Free
space is space with nothing else present. In electrostatics, a charge density distribu
tion in space, p ( p ) , results in a potential, 4 ( p ) , to exist in space which varies with
position, p . The basic equation governing this relation, called the Poisson equation,
is
(3.8 1b)
Various methods for solving this linear differential equation are presented in texts
that discuss electrostatics. By one method, the potential is obtained by an integration
of the charge density called the potential integral. By using the concepts we have
developed for LTI systems, we shall obtain this integral without solving the Poisson
equation.
For our development, we view the problem as a system with the input being the
charge density distribution, p ( p ) , and the output being the potential, 4 ( p ) . Note that
the input and the output in our defined system are functions of position, p , and not
time. We first show that the system is linear. This is shown by noting that if a certain
charge, q l , results in the potential distribution 41,and some other charge, q2,results
+
in the potential distribution &, then the charge q = clql c2q2 will result in the
+
potential distribution 4 = clg51 ~ ~That4is, the ~ mapping
. of inputs to outputs is
a linear mapping.
Now let the potential distribution in free space due to a charge located at p = p 1
+
be 4 ( p ) . If the charge is moved from p , to a new position, p 2 = pI d, then the
+
potential distribution due to the charge in its new location will be 4 ( p d). That is,
the potential distribution will remain the same except for being translated in space by
the same amount and direction as was the charge. Thus we observe that the mapping
of inputs to outputs is shiftinvariant. Note that because the input and output of our
defined system are functions of position only, we have shift invariance instead of
time invariance. Thus our system is a linear shiftinvariant (LSI) system.
The expression for the output of an LTI system is the convolution of the input
with the system unitimpulse response. Thus, by replacing position for time we have
that the output potential distribution of our LSI system will be equal to the spatial
convolution of the input charge density distribution with the LSI system unit
impulse response.
3.8 THE POTENTIAL INTEGRAL 95
(3.82)
1
q5o(r) = (3.83a)
where E is the dielectric permittivity of the space and r is the distance from the unit
point charge. In rectangular coordinates, this equation is
4(x, y , z ) = 111
p(x’, y’, l ) 4 0 ( x  x’, y  y’, z  z’) dx’dy’dz‘
where the integration is over all space. Clearly, if the charge density is nonzero in
only some region in space, then the only nonzero contribution of the integration is
over that region in which the charge density is nonzero. The integral is called the
potential integral. Note that we were able to obtain this result without solving the
Poisson equation by using the concepts of LTI we have developed.
Clearly, causality is not a meaningful concept because we are concerned with
position and not with time in our chargepotential system. We will not discuss the
96 PROPERTIES OF LTI SYSTEMS
PROBLEMS
31 System A is an LTI system with the unit impulse response h,(t) =
r(t)  r(t  1). System B is the tandem connection of two systems A as
shown below.
1
B
............................... _......__.___.
...._
Use the results of Section 3.1 to show the following LTI system relations.
Also draw the equivalent block diagrams.
(a) [ha(t) * hb(t)l * = ha(t) * [hb(t) * hc(t)l
(b) C[ha(t) * hb(t)l = [Cha(t)l * hb(t) = h a ( t ) * [Chb(t)l
(c) ha(t) * + hc(t)l = * + [ha(t) * hc(t)l
(dl + * hc(t) = [ha(t) * hc(t)l [ h b ( t ) * +
(e) h a w * W )= h a w
(f) ha(t) * = * ha(t)
In mathematics, these six properties define a commutative algebra with a
unity element. Properties (a) and (b) are called associative properties.
Properties (c) and (d) are called distributive properties. Property (e) is the
statement that there is a unity element, h(t) = S(t). Property (f) is the
commutative property.
The importance of this result is that the whole mathematical theory of
commutative algebras with a unity element can be directly applied to LTI
system theory. One direct application of the above six properties is block
diagram reduction, which is discussed in Chapter 10.
(a) Sketch s(t). Label all important time and amplitude values.
(b) Sketch h(t), the system unitimpulse response.
(c) Is the given system causal? Your reasoning must be given.
(d) Is the given system stable (BIBO)? Your reasoning must be given.
34 System A is an LTI system with the unitimpulse response h(t). System A is
connected in tandem with an ideal delay system with a delay equal to T
seconds.
(a) Show that the ideal delay system is an LTI system.
(b) Use the discussion in Section 3.2 to prove for system A that if
x ( t ) + y(t), then x(t  T ) +y(t  T ) .
(c) Also show that if the unitimpulse response of system A is changed to
h(t  T ) , then x(t) + y(t  T).
35 The unitimpulse response of an LTI system is h(t) = Ar(t/2). Determine the
following:
(a) Its unitstep response, s ( t ) .
(b) Its response to the input x ( t ) = r(t).
(c) Its response to the input x ( t ) = r(t/2).
Note that the solution to this problem can be obtained without using
convolution.
36 Let h,(t) = d(t  to) in Fig. 3.22, and thus show
(a) y(t  to)= x(t  to)* h(t).
(b) y(t  to) = ~ ( t* )h(t  to).
37 (a) Show that d(t) * d(t) = d(t) by evaluating lim,+o d,(t) * d,(t).
(b) Show that d(t) * d(t) * . ' . * d(t) = d(t).
Suggestion: Letf,(t, E ) = d,(t) * d,(t) * . . . * d,(t) and show that:
1. .f,(t, E ) is a positive pulse which is zero outside the interval
m < t < nc.
2. The area underf,(t, E ) is equal to one. That is, JyWf,(t, E ) dt = 1 so
that, from the results of Section 3.4, limc+Ofn(t,E ) = d(t).
38 +
The unitstep response of an LTI system is s(t) = [I  (t 1)e']u(t). Use
the results developed in Sections 3.2 and 3.3 to determine the system unit
impulse response, h(t).
98 PROPERTIES OF LTI SYSTEMS
312 *
Let y l ( t ) = x(t) x(t) and y2(t)= x(t) * x(t). Determine y l ( t ) and y2(t)for
x(t) = AeP'u(t), where a > 0.
313 (a) Determine the unitimpulse response of the LTI feedback system shown
below.
(b) Use the result of part a to determine the system unitstep response, s(t).
(c) Use the result of part b to determine the system response to the input
x(t) = r(t/4).
314 Let the unitimpulse response of an LTI system be h(t) = Ar(t) and let the
system input be the triangle, x(t) = (1  Itl)u(l  It[).Determine the system
response, y(t), by
(a) Convolution.
(b) Differentiating x(t).
(c) Differentiating h(t).
315 Let the unitimpulse response of an LTI system be h(t) = Ar(t/2), and let the
system input be the exponential, x(t) = Be3'u(t). Determine the system
response, y(t), by
(a) Convolution.
(b) Differentiating h(t).
PROBLEMS 99
316 Let the unitimpulse response of an LTI system be h(t) = Ar(t/T),and let the
system input be the sinusoid x(t) = B sin(w,t)u(t), where coo = 2n/T. Deter
mine the system response, y(t), by
(a) Convolution.
(b) Differentiating h(t).
317 Two LTI systems, A and B, are connected in tandem to form the system C
shown in the diagram below.
..............................C
................. ......
h,(t) =
l 1
1
0
forl<ttO
for 0 < t < 1
otherwise
318 Consider two causal LTI systems, A and B, with unitimpulse responses ha([)
and hb(t),respectively. The unitstep responses of systems A and B are sa([)
and s,(t), respectively. It is observed that s,(t) = sb(t)for 0 < t < 1.
(a) Is it necessary that s,(t) = s,(t) for t > l? Explain.
(b) Is it necessary that the responses of systems A and B to the input
x ( t ) = r(t) be equal? Explain.
319 Use convolution to show that, for a causal system, y(t) = 0 for t < to if
for t < to.
x(t) = 0
320 In this problem, you are asked to prove the inequality, Eq. (3.62), used to
prove the necessary and sufficient condition that an LTI. system be stable.
(a) Show that
iff(t) > 0 over some portion of the integration interval andf(t) < 0 over
the other portion of the integration interval.
321 (a) The unitimpulse response of an LTI system is h(t). How is the point
t = 0 determined experimentally?
For each statement given below, state whether it is true or false and give a
short statement of your reasoning.
(b) The unitimpulse response of a stable LTI system can be a periodic
function.
(c) The input of a causal and stable LTI system must be zero for t < 0.
(d) If the input, x(t), of an LTI system is periodic, then depending on h(t), the
output, y(t), may or may not be periodic.
322 Consider two tandem connected LTI systems as shown in Fig. 3.13.
Use the basic definition of causality to show that the tandem connected
system is causal if systems A and B are causal. Thus show that h(t) = 0
for t < 0 if both h,(t) and hb(t)= 0 for t < 0. From this, we have the
general result that the convolution of two functions that are zero for t < 0
is a function that is zero for t < 0.
Use the basic definition of stability to show that the tandem connected
system is stable if systems A and B are stable. Thus show that the area
under Ih(t)l is finite if the area under Ih,(t)l and Ihb(t)l are finite. From
this we have the general result that if the area under the magnitude of two
functions is finite, then the area under the magnitude their convolution is
also finite.
323 For each of the systems with the response y(t) to the input x ( t ) described
below, determine whether it is (1) linear, (2) timeinvariant, (3) stable, and (4)
causal.
PROBLEMS 101
~ ( t=) u + h ( t )+ C X ( ~
 T ) + dx’(t)
Determine the values of the constants, a, 6, c, d, and T for which the system is
(a) Linear.
(b) Timeinvariant.
(c) Causal.
(d) Stable.
325 Show that an LTI system with the unitimpulse response h(t) = u(t) is not
stable by determining the system response to the input x(t) = u(t).
326 Show that the causal LTI system with the unitimpulse response
+
h(t) = (1/1 t)u(t) is not BIBOstable so that it is necessary but not
sufficient that limt+m h(t) = 0 for an LTI system to be BIBOstable.
327 The response, y(t), for the input, x(t), of an LTI system is
329 The response, y(t), to the input, x(t), of a certain class of systems is
102 PROPERTIES OF LTI SYSTEMS
where P(t) 2 a(t). Determine restrictions, if any, on a(t) and P(t) are required
in order that the system be
(a) BIBOstable.
(b) Causal.
(c) Timeinvariant.
( 4 L'inear.
330 The response of a given system to the input x(t) is y(t) = x(2  t ) . Determine
whether the given system is
(a) Linear.
(b) Timeinvariant.
(c) Causal.
(d) BIBOstable.
+
(b) Show that y(t  to) = h+(t) * x(t) h(t) * x(t) = y+(t)+y(t), where
y+(t) depends only on present and past values of x(t) and y(t) depends
only on future values of x(t).
The system with the unitimpulse response h(t) is not a physical one
and thus cannot be realized. We thus construct a system with the unit
impulse response h+(t) for which the output isy+(t). The error incurred is
PROBLEMS 103
'Characteristic vectors also are called eigenvectors, and characteristic values are also called eigenvalues.
Eigen means characteristic in German. The two alternate names are thus a mixture of German and English.
English, after all, is rather eclectic, so that it is not uncommon to see words in the English language which
have been borrowed in whole or part from other languages. For clarity, I am using the wholly English form
of the names.
105
106 THE FREQUENCY DOMAIN VIEWPOINT
i= 1
Thus the characteristic vectors are a natural coordinate system to use for describing
the mapping of vectors by the given matrix.
In general, the objects that are invariant under a given linear mapping are impor
tant objects to determine and study because they can be used to characterize the
mapping. In the theory of LTI systems, an input function is a characteristic function
if the corresponding output is equal to a constant times the input. The constant is the
characteristic value associated with the characteristic function. We shall study the
characteristic functions and characteristic values of an LTI system and their signifi
cance in this chapter.
In accordance with our discussion in Section 3.7, the stability condition is required
so that the output for a given input can be discussed without ambiguity. For our
present discussion, please note that we are not requiring the system to be causal. We
shall show that the characteristic function of a stable LTI system is the phasor
Note that the phasor is not zero over any time interval. Also, the magnitude of the
phasor is one. This is seen by noting that the phasor is the polar form of a complex
number with the magnitude of one and an angle equal to wt. That the magnitude is
one for all values o f t also can be obtained by expressing the phasor in trigonometric
form as [see Appendix A, Eq. (A12)]
so that the magnitude of the phasor is one for all values oft. We thus note that the
phasor is a bounded waveform. This is a good time to study Appendix A for a review
of complex algebra if you are a bit uncertain about it because we shall have continual
need of complex algebra from this point on.
Because the input phasor is a bounded waveform, the corresponding system
output also must be a bounded waveform because we are considering only BIBO
stable systems. It is important to keep in mind that the only waveforms that can be
4.1 THE CHARACTERISTIC FUNCTION OF A STABLE LTI SYSTEM 107
used physically as a system input are real functions. Because the phasor is a complex
function, it cannot be used physically as a system input. However, we certainly can
consider it theoretically. Later, we shall construct real waveforms as linear combina
tions of phasors to obtain the system response to real waveforms.
We have shown that generally, for a stable LTI system, the output, y(t), for the
input, x(t), is
y(t) = 1
00
00
h(z)x(t  z) dz (4.14)
y(t) = 1
00
00
h(z)dw('')dz (4.15)
(4.16)
we obtain
00
(4.17)
The phasor, eJCot, is factored out from under the integral because the integration is
over 5, and so it is a constant during the integration. Now the value of the integral is a
complex number that depends on the value of w and the function h(t). We thus
express it as
H(jw)= 1 00
00
h(z)eJWT
dz (4.18)
The capital H indicates its dependence on h(t). With the use of Eq. (4.18), we can
express y(t) as
We observe that the response of a BIBOstable LTI system to a phasor is the same
phasor multiplied by the constant, H ( j w ) . Thus the phasor, eJwf,is a characteristic
function of the stable LTI system, and the constant H( J w ) ,is the characteristic value
associated with the characteristic function. In LTI system theory, the characteristic
value, H( j w ) , is called the transfer function for reasons we shall discuss in the next
section.
108 THE FREQUENCY DOMAIN VIEWPOINT
(4.110 )
In obtaining Eq. (4.110) we have used the result developed in Appendix A that the
magnitude of the product of two complex numbers is equal to the product of their
magnitudes and, as shown above, that the magnitude of a phasor is one. As discussed
above, y(t) is a bounded waveform because we are considering only BIBOstable LTI
systems. Consequently, from Eqs. (4.110) and (4.18) we have that
(4.11 1)
That is, the value of the integral must be finite for any value of o because the system
is BIBOstable. The necessary and sufficient condition for the LTI system to be
BIBOstable was shown to be
(4.1 12)
Thus we note that a sufficient condition for the integral given by Eq. (4.18) to
converge (that is, for the value of the integral to be finite) is that the area under the
magnitude of h(t) be finite. Thus, the value of the transfer function, H(jo), of any
BIBOstable LTI system is not injnite for any value of o.
Before proceeding, let us consider a simple specific example to illustrate the
results obtained. We consider an LTI system with the unitimpulse response
(4. I  14)
The lower limit of the second integral is zero because, for our example, h(t) = 0 for
t < 0. Also, for t > 0 we have Ih(t)l = IAle%(t).
The transfer function is obtained by using Eq. (4.18). For that integration, the
function to be integrated is
4.2 SINUSOIDAL RESPONSE 109
so that
(4.1  16)
Jo

A
 , U > O
a+jo
(4.117a)
L H(jo)= L A  tan
' (3
where, as discussed in Appendix A, we have
0 ifA>O
(4.1 17b)
.n ifA<O
Note that, as expected, H ( j o ) is not infinite for any value of w. Thus we have for our
example that the response of the given stable LTI system to the input x(t) = eJwtis
y ( t ) = [A/(u+jw)].'"'.
We showed in the last section that the response of a BIBOstable LTI system to the
input x(r) = e'"'' is y ( t ) = H ( jw)ejwt in which the transfer function, H( jo),is given
by Eq. (4.18). From generalized superposition, Eq. (2.1l), the response of a linear
system to a linear combination of functions is the same linear combination of the
system response to each individual function. Consequently, the LTI system response
to the input
(4.21)
is
(4.22)
110 THE FREQUENCY DOMAIN VIEWPOINT
Thus, once the transfer function is known, we can determine, without convolution,
the response of the stable LTI system2 to any input, which can be expressed as a
linear combination of phasors. This is a principal reason why characteristic functions
are important in the theory of stable LTI systems. That is, if the input can be
expressed as a linear combination of characteristic functions, then the output can
be expressed as the same linear combination of the mapping of each individual
characteristic function (which is just the characteristic function times its character
istic value). This result leads to a different and important view of LTI systems. Our
previous analysis using convolution considers a system in the time domain. The
analysis we shall develop using the above result considers a system in thefiequency
domain. Note from Eq. (4.21) that the analysis we shall develop is restricted to
inputs that can be expressed as the linear combination of phasors. However, even
with this restriction, the frequency domain analysis we shall develop results in an
important and useful view of LTI systems.
We begin our development by determining the response, y(t), of a stable LTI
system to the constant input
x(t) = E (4.23)
This equation states that the input is a constant equal to E for GO < t < 00. Such
an input is often referred to as a dc input3 with a dc value of E. This input can be
expressed in phasor form as E times a phasor with a frequency o = 0. That is
x(t) = E d o (4.24)
We thus note that the system response also is a constant with the dc value of EH(0).
For this reason, IH(O)( is called the dc gain of the LTI system. Note from Eq. (4.18)
that
00
(4.26)
The dc gain is seen to be equal to the magnitude of the area under h(t). In conse
quence, the dc gain is zero if and only if the area under h(t) is zero.
2For conciseness from now on, stable will mean BIBOstable because that is the type of stability with
which we are mainly concerned in this text.
The notation dc stands for direct current. This is a carryover from an abbreviation used in the early days
of electrical engineering. Today, it is used to refer to any constant waveform, not just direct current.
4.2 SINUSOIDAL RESPONSE 111
We now determine the response, y(t), of a stable LTI system to the sinusoidal
input
Note that the input, x(t), is a sinusoid for 00 < t < 00. This input can be
expressed as the linear combination of phasors. From Eq. (A14) in Appendix A,
we can express x ( t ) as
(4.28)
(4.210)
(4.21 1)
J 00
112 THE FREQUENCY DOMAIN VIEWPOINT
Equations (4.21 1) and (4.212) are identical if h(t) = h*(t). But, from Appendix A,
this is true if and only if h(t) is real. We thus have shown that H(jo) = H * ( j o ) if
and onZy if h(t) is real. This is the case for physical systems. It is sometimes
convenient for theoretical reasons to define an LTI system (which is not a physical
system) with a unitimpulse response that is a complex function. All results obtained
for the case in which h(t) is a real function must be reexamined when studying such
theoretically contrived systems.
As expected for physical systems, we have with this result that the two terms in
Eq. (4.29) are conjugates, so that from Eq. (A19) in Appendix A we have
(4.213)
This equation can be put in a much better form by expressing the transfer function in
polar form:
(4.215)
The real part of this expression is easily obtained with the use of Euler's formula [Eq.
(A12) of Appendix A]. The result is
By use of Eq. (4.22) we have been able to obtain the system response to the
sinusoidal input given by Eq. (4.27) without convolution.
There are a few important observations to make concerning our result:
1. The first observation is that the response of a stable LTI system to a sinusoid is
a sinusoid. Note from our previous examples in Section 2.5 that this is not
generally true for other waveforms. Also, it is not generally true for nonlinear
or even linear timevarying systems; it is only for LTI systems that it is
generally true.
2. The second observation to make is that the frequency of the output sinusoid is
the same as that of the input sinusoid. This is not generally true for nonlinear
or even linear timevarying systems. It is only for LTI systems that it is
generally true.
4.2 SINUSOIDAL RESPONSE 113
3. The third observation to make is that the ratio of the amplitude of the output
sinusoid to that of the input sinusoid is equal to the magnitude of the transfer
function, IH( jo)l. For this reason, IH( jo)l is called the system gain. The
system gain is a function of the frequency, o.The value that must be used in
Eq. (4.216) is the value of the system gain, IH(jo)l, evaluated at the
frequency of the input sinusoid. Note that the output amplitude must be
finite because the system is stable and the input amplitude is fiqite. In
consequence we conclude from Eq. (4.216) that IH(jw)l < 00. This is just
another way of obtaining the result given by Eq. (4.11 1).
4. The fourth observation to make is that the phase angle of the output sinusoid is
equal to that of the input sinusoid plus e(o) = i H ( j w ) . That is, the input
sinusoid has been phaseshifted by L H( jw). For this reason, L H ( j o ) is
called the system phaseshift. Like the system gain, the system phase shift is a
fimction of the frequency, o.The value that must be used in Eq. (4.216) is the
value of the phase shift, Q(o) = L H( jo), evaluated at the frequency of the
input sinusoid.
Note that the same output, y(t), given by Eq. (4.216) would be obtained if the
phase shift were increased or decreased by any integer number of 2n radians (or,
equivalently, an integer number of 360"). For example, one cannot differentiate
between a phase shift of 400" and one of 40" or between a phase shift of 300"
and one of 60". For this reason, when reporting the phase shift, an integer number
of 360" (or equivalently, an integer number of 2n radians) often is added or
subtracted from a calculated value of phase shift so that the reported value of
Q ( w ) lies in the range 180" < O(o) 5 180" (or, equivalently, in the range
n < d(w) 5 n radians).
As an illustration, consider an LTI system with the unitimpulse response
h(t) = AeP'u(t) where a > 0. The gain and the phase shift of this system was
obtained in Section 4.1, Eq. (4.117). From that result we have that the system
response to the input
~ ~I ~ ( o O t
y(t) = d 7EIA
a2 + o
+ 4o+ L A  tan' 9) U
(4.218)
Observe for this example that, for a given input amplitude, the higher the frequency
oo,the smaller the amplitude of the output. This is so because the gain for the given
system is a decreasing function of frequency. In consequence, such a system is called
a lowpass filter because lowfrequency sinusoids are "passed" with a larger gain
than are highfrequency ones.
114 THE FREQUENCY DOMAIN VIEWPOINT
A
= [1 e j wT
]
The gain and the phase shift of this system can be obtained easily by first expressing
H ( j o ) in polar form. This can be done by noting that
(4.223)
(4.224)
(4.225a)
4.2 SINUSOIDAL RESPONSE 115
Because the angle of a positive real number is zero and the angle of a negative real
number is n radians, we have
0 ifA>O
IAT= (4.225b)
n ifA<O
and also
0 if
sin (9)
>o
2
n if
sin (F)
wT <o
(4.225~)

2
Thus, to determine and sketch the gain and the phase shift versus w, we need to
sketch sin(wT/2)/wT/2 versus w. For this, we first sketchf (8) = sin(8)/8 versus 8,
which is shown in Fig. 4.21. Note thatf(8) =f(8). A function for which this is
true is called an even function. Becausef(8) is an even function, we really only need
to determine its graph for 8 3 0 because the graph for negative values of 0 is then
easily obtained by reflecting the curve for positive values of 8 about the ordinate.
The sketch is obtained by first obtaining a simple expression for the curve for small
values of 8. This is obtained by using the power series of sin 8 about 8 = 0, which,
from Appendix A, is
sin8=88 1
3!
3 +... (4.2 26a)
(4.226b)
From this approximation, we observe thatf(0) = 1 and that the graph off(@ about
8 = 0 is a parabola. The rest of the graph off(8) shown in Fig. 4.21 is obtained by
noting that
I
1 n
 when sin8 = 1 or 8 =+2nn
8 2
.f(Q= !when sin8=1
371
or 8 =  + 2 n n
(4.227)
8 2
0 when sin8 = 0 or 8 = nn
I,
00
y(t) = x(z)h(t  Z) dz
(4.228)
The output, y(t), is thus A times the area under the last T seconds of x(t). Now,
the gain and the phase shift is defined only for a sinusoidal input as given by Eq.
(4.217). For this input, the system response for our example is
If the frequency of the input sinusoid is f = n/T Hz, then from our discussion of
sinusoids in Section 1.4, Eq. (1.414), the output, y(t), is the area under the last n
cycles of the input sinusoid. This area is zero because the area under each sinusoidal
cycle is zero. Thusy(t) = 0 for f = n/T Hz. From Eq. (4.216), this means that the
gain is zero at these frequencies. Remember that the gain and the phase shift are
defined only for a sinusoidal input.
The LTI system response to any input that can be expressed as the sum of
sinusoids now can be immediately determined with the use of generalized super
1
0.9
0.0
0.7
0.6
0.2
0.1
0
0.1
0.2
0.3I
5 4 3 2 1 0 1 2 ' 3 4 5
e
2n
271
4.2 SINUSOIDAL RESPONSE 117
position and our present result, Eq. (4.216). The stable LTI system response to the
input
(4.230)
is
The transfer fimction thus allows us to determine, without convolution, the response
of any stable LTI system to any input that can be expressed as the sum of phasors,
Eq. (4.2l), or, equivalently, as the sum of sinusoids, Eq. (4.230).
The input, x(t), given by Eq. (4.230) is for 00 < t < 00. If, for example, the
input were zero for t < 0, then the system response would not be given by Eq.
(4.231). However, as t + 00, the system output of any stable LTI system will
tend to the output given by Eq. (4.231). Consequently, it is called the steady
state response.
This last result can be shown using our discussion at the end of Section 3.5. For
the example given there it is shown that, as an input value recedes into the past, its
influence on the system output decreases exponentially because the impulse
response, h(t),of the given system decays exponentially. Consequently, in the convo
lution integral given by Eq. (3.58), the influence of the input values for t 5 0
decreases exponentially as to increases. This means that the output of the given
system will tend to the same waveform as to increases irrespective of the input
waveform for t < 0. Thus, for an input given by
the output will tend to that given by Eq. (4.23 1) as t + 00. The output given by Eq.
(4.23 1) is thus called the steadystate system response because it is the response to
which the given system output tends as t + 00. The difference between the actual
response and the steadystate response is called the transient response. Note that the
transient response tends to zero as t + 00.
We now can generalize our discussion above to any stable system. The necessary
and sufficient condition for stability of an LTI system is given by Eq. (3.61). The
requirement, as discussed in Section 3.6, is that the area under Ih(t)l be finite. Note
that this requires that h(t) + 0 as t + 00 because, if not, then the area under Ih(t)l
would be infinite. The system discussed above is a special case of a stable LTI
system because its impulse response decreases to zero exponentially as t increases.
In terms of our discussion in Section 3.5, we thus observe that any stable LTI system
has a memory that decays as time increases so that as an input value recedes into the
past, its influence on the output decreases. Thus, as time increases, the influence of
the input values for t 5 0 decreases so that, for an input given by Eq. (4.232) above,
118 THE FREQUENCY DOMAIN VIEWPOINT
the output will tend to the steadystate response given by Eq. (4.231). The rate of
this approach, which is the same as the rate at which the transient tends to zero,
depends on the rate at which Ih(t)l approaches zero as t increases; equivalently it
depends on the rate at which the system memory decays as time increases.
These results lead us to the question as to which input waveforms can be
expressed as the sum of phasors, Eq. (4.2l), or, equivalently, as the sum of sinu
soids, Eq. (4.230), and if so, how to determine the expression. The representation
of a waveform as the linear combination of phasors is called Fourier analysis. Our
results in this section is the reason Fourier analysis is important in the theory of
stable LTI systems. It is a consequence of the result that the phasor is a characteristic
function of a stable LTI system. If some waveform other than the phasor were a
characteristic function of an LTI system, we would be interested in expressing the
input as a linear combination of the other waveform. It is important to appreciate this
because the phasor is not a characteristic function of, for example, linear time
varying (LTV) systems. Consequently, Fourier analysis is of limited value in the
study of LTV systems.
Before considering Fourier analysis, we shall examine the gain and the phase shift
of tandem connected systems. The results we shall obtain are basic because they
form the basis for our analysis of LTI systems in the frequency domain. An inter
esting observation to make as you study the following sections is that the results
obtained can be viewed as mathematical results without any regard to their system
theory source. These results are the fimdamental ones required for the development
of Fourier theory, which we shall discuss in the next chapter.
In Section 3.1, the tandem connection of two LTI systems as shown in Fig. 4.31 was
studied in the time domain. We showed there that the tandem connection of two LTI
systems, systems A and B, is an LTI system. Consequently, it was shown that the
tandemconnected system can be represented in terms of an impulse response, h(t),
given by Eq. (3.15). Using the notation established in that section, we showed
(4.31)
.
........................................................................
...
Fig. 4.31 Two LTI systems connected in tandem.
is true for the tandem connection of any two systems whether linear or nonlinear,
timeinvariant or timevarying.
We showed in Section 3.5 that an LTI system with a unitimpulse response h(t) is
causal if and only if h(t) = 0 for t < 0. We thus conclude the following
If
h,(t) = 0 fort < 0 and hb(t) = 0 fort < 0
then
h(t) = h,(t) * hb(t) = 0 for t < 0
Note that this result can be viewed purely as a mathematical result without any
regard to system theory. That is, if two functions that are zero for t < 0 are
convolved, then the result is a function that is zero for t < 0. The obtaining of
mathematical results from system theory results is an important and useful technique
that we shall use.
For the second property, we show that if systems a and b are stable systems, then
the tandem connected system is a stable ~ y s t e m .This
~ result follows from the
definition of BIBO stability. If x(t) is a bounded waveform, then y,(t) is a bounded
waveform because system A is a stable system. Ify,(t) is a bounded waveform, then
z(t) is a bounded waveform because system B is a stable system. Thus the tandem
system is BIBOstable because every bounded input, x(t), produces a bounded
output, z(t).Thus, if systems A and B are stable systems, then the tandem connected
system is a stable system. Again note that the type of system was not used in this
argument, so that this result is true for the tandem connection of any two systems
whether linear or nonlinear, timeinvariant or timevarying.
We showed in Section 3.6 that an LTI system with the unitimpulse response h(t)
is BIBOstable if and only if h(t) is absolutely integrable; that is, if and only if
(4.32)
Remember that, for conciseness, we are using “stability” to refer to “BIBOstability” because that is the
type of stability with which we are mainly concerned in this text.
The L is used to honor Henri Leon Lebesgue (18751941), who made major contributions to the theory
of integration. The sub1 denotes that it is the integral of the first power of the magnitude of the function
that is finite.
120 THE FREQUENCY DOMAIN VIEWPOINT
If
then
Note that we have not assumed either system A or system B to be causal in obtaining
this result. Thus this result is valid even if the component systems are not causal.
Again, note that this result can be viewed purely as a mathematical result without
any regard to system theory. That is, if two L , functions are convolved, then the
result is a L , function. This is another example of obtaining a mathematical result
from a system theory result.
We now determine the transfer function of a tandemconnected system. Because,
from Eq. (3.15), h(t) = h,(t)*hb(t), we should be able to express H ( j w ) in terms of
H,(jw) and H b ( j 0 ) . This could be done by substituting the convolution integral,
Eq. (3.15), into the equation for the transfer function, Eq. (4.18), and manipulating
the resulting equation to obtain the desired result. However, a simpler and more
insightful method is to use the fact that the phasor is a characteristic function of a
stable LTI system.
The method uses the result expressed by Eq. (4.19), which is that
e@' + H ( jw)ej"' for a stable LTI system. For this we consider the tandem connec
tion shown in Fig. 4.31 in which systems A and B are stable. The tandemconnected
system then is stable in accordance with our result above. Now let x ( t ) = eJwt.The
response of the tandemconnected system to the input phasor is z(t) = H(jw)ej"'.
Another expression for z(t) is obtained by following the phasor through the tandem
connection. For this we have that the response of system A to the input phasor is
y,(t) = H,(jo)ej"'. Thus y,(t) is equal to a constant, H,(jw), times the phasor, eJwt.
Hence using the homogeneous property of linear systems, Eq. (2. 13), the response
of system B is z(t) = H u ( j w ) H b ( j w ) e j WFrom
'. above, z(t) = H(jw)ejw'. Because
the two expressions for z(t) must be equal for all t , we immediately have the result
Note that we have not assumed either system A or system B to be causal in obtaining
this result. Thus this result is valid even if the component systems are stable but not
causal.
This result also can be viewed purely as a mathematical result without any regard
to system theory. That is, if h,(t) and h b ( t ) are L , functions, then h(t) = h,(t)*hb(t)
is an L , function in accordance with our previous result and also
4.3 TANDEMCONNECTED LTI SYSTEMS 121
The gain of the tandemconnected system is thus the product of the gains of the
twocomponent system because, from Eq. (4.33) and with the use of Eq. (A26) in
Appendix A, we obtain
Also from Eq. (4.33), the phase shift of the tandemconnected system is the sum of
the phase shifts of the twocomponent systems because, with the use of Eq. (A27)
in Appendix A, we have
The last two relations also can be obtained from physical considerations. For this,
let the input, x ( t ) , be a sinusoid with a magnitude /El and a frequency o,rad/s.
Because the gain of system A is IH,(jwo)l, the output of system A , y,(t), is a
sinusoid with a magnitude IEIIH,(joo)l. The sinusoid vu([)is the input of system
B. Because the gain of system B is IHb(joo)l, the output of system B, z(t), is a
sinusoid with a magnitude IEIIH,(jwo)llHb(joo)l. Thus we see that the gain of the
tandemconnected system is that given by Eq. (4.35). The phase shift also can be
obtained from physical considerations. System A shifts the input sinusoid by an
amount equal to L H,(joo), and system B shifts the sinusoid y,(t) by an amount
equal to L Hb( jo,).Consequently, the difference in phase between the output sinu
soid, z(t), and the input sinusoid, x(t), is the sum of the two phase shifts as given by
Eq. (4.36).
If a third stable LTI system with the transfer function H,( jo)were connected in
tandem with the other two stable LTI systems, it is clear from our previous work that
the tandenconnected system is a stable LTI system with the transfer fknction
Clearly, if several stable LTI systems are connected in tandem, the transfer function
of the tandemconnected system is simply the product of the individual transfer
hnctions.
To illustrate the results we've obtained, consider the system formed by connecting
P identical stable LTI systems in tandem, each with the unitimpulse response:
A
=
Hu(jo) (4.3 9)
a +jo
122 THE FREQUENCY DOMAIN VIEWPOINT
(4.310)
(4.31 1)
L H ( jo)= P [ L A  tan'
%I
U
(4.312a)
where
LA= { 0 ifA>O
71 i f A < O
(4.312b)
(4.313a)
(4.313b)
This is a lowpass filter because the higher the sinusoidal frequency, the smaller the
output amplitude. The bandwidth of such a filter is usually defined as the frequency
at which the gain has decreased by a factor 4 of its maximum. For our filter, the
maximum gain is at coo = 0. At this frequency, the gain is IAIp/d'. Thus the
bandwidth is the frequency at which the gain is IAIP/&aP. To determine this
frequency, w, we must determine the frequency oo= w at which
(4.315)
4.4 CONTINUOUS FREQUENCY REPRESENTATION OF A WAVEFORM 123
This requires
Raising both sides of the equation to the power 2/P and solving for w, we obtain
w = a m (4.317)
Notice that the larger the number of identical systems which are connected in
tandem, the smaller the bandwidth of the tandemconnected system. The bandwidth
for a few values of P are:
P: 1 2 3 4
w: a 0 . 6 4 ~ 0 . 5 1 ~ 0 . 4 3 ~
Note that with two identical systems in tandem, the bandwidth has been reduced to
64% of the bandwidth of a single system. The bandwidth with three identical
systems in tandem is only about half of the bandwidth obtained with a single system.
Of course, the gain of P identical systems in tandem is the gain of a single system
raised to the power P so that, in design, one can trade bandwidth for gain in this
manner. In our discussion of filters later in this text, we shall examine methods of
improving the filter characteristics by not using identical systems in tandem.
We showed in Section 4.2 that if the input, x ( t ) , of BIBOstable LTI system can be
expressed as the linear combination of phasors,
(4.41)
then the output, y(t), can be expressed as the same linear combination of the system
response to each individual phasor:
(4.42)
As long as the sum in Eq. (4.4.1) converges, the number of frequencies in the sum
can be arbitrarily large. In fact, instead of summing over a discrete set of frequencies,
the sum can be extended over all frequencies from ato +co. Now, an integral can
124 THE FREQUENCY DOMAIN VIEWPOINT
be viewed simply as a sum over a continuous variable so that we also can express the
input given by Eq. (4.41) as an integral over all frequencies:
x(t) = 103
03
C ( o ) d w fd o (4.43)
y(t) = 103
03
C(o)H(jo)ejufd o (4.44)
In the representation of the input given by Eq. (4.43), the phasor amplitudes are
zero but they have an amplitude density. To understand this concept better, first
consider this textbook. It has a certain mass. However, there is zero mass at any
point within the book because the volume of a point is zero. If the mass at any point
within this book is zero, how then can this book have nonzero mass? For a descrip
tion of the mass of this text, it is not fruitful to discuss the mass at a point because it
always will be zero. Rather we define a mass density at a given point. This is
obtained by determining the mass in a small sphere centered on the given point.
We then determine the ratio of the mass within the sphere divided by the volume of
the sphere. This ratio tends to some limiting value as the sphere's radius goes to zero.
The limiting value of the ratio is called the muss density at that point. The mass
density generally will vary from point to point. With this definition, the mass of this
book is just the integral of its mass density over all points within the book.
Let us now consider an integral of the form
I =
1: f ( t ) dt (4.45)
Note that the area under any point off(t) is zero. Thus, how can there be area under
the curve if the area under every point is zero? Similar to our discussion in Section
2.3, this problem is circumvented by first forming narrow rectangles as shown in Fig.
4.41. The sum of the areas of the rectangles is an approximation of the value of the
integral, I . This sum tends to some limiting value as the widths of the rectangles go
to zero. The value of the integral, I , is this limiting value.6 Now consider the shaded
rectangle with its center at t = z shown in Fig. 4.4 1. Let the width of the rectangle
be dt, an infinitesimal. The area is then du, also an infinitesimal. From the figure, the
infinitesimal area is du = f ( z ) dt. Thus, we can expressf(z) asf(z) = du/dt. Note
that this is the area density at t = z. That is, from the point of view of Riemann
integration, the graph off(t) versus t is a graph of its area density versus t ! From this
Integration by this procedure is called Riemann integration in honor of the German mathematician Georg
Friedrich Bemhard Riemann (18261 866), who made major contributions to many areas of mathematics,
including the theory of integration.
4.4 CONTINUOUS FREQUENCY REPRESENTATION OF A WAVEFORM 125
viewpoint, note that the impulse, d(t  z), which has unit area in the infinitesimal
interval about the point t = z, plays the same role as the unit point mass in
mechanics or the unit point charge in electromagnetics.
With this understanding of integration, we can view the representation of x ( t ) in
Eq. (4.43) as the sum of phasors with an amplitude density of C(o).To obtain a
representation in the form of Eq. (4.43) as a linear combination of phasors with
nonzero amplitude as in Eq. (4.4l), C(w) must contain impulses. For example,
consider the case for which C(o)is
~ ( t=)
n
C,, 1
00
oo
d ( o  on)dW'
do
(4.47)
= cnejwnt
n
The value of the integral is obtained using the sifting property of the impulse derived
in Section 2.4. Equation (4.47) is exactly Eq. (4.4l), which is the linear combina
tion of phasors with frequencies onand nonzero amplitudes C,. Thus we note that if
x ( t ) can be represented in the form given by Eq. (4.43), then the response, y(t), of
the BIBOstable LTI system is given by Eq. (4.44).
The transfer function, H ( jo),is obtained from the unitimpulse response, h(t), by
means of Eq. (4.18). But can the unitimpulse response be obtained from knowl
edge of the transfer function? This is a very important question. We have discussed
the fact that h(t) completely characterizes the inputoutput mapping of a given stable
LTI system because once it is known, the output, y(t), due to any input, x(t), can be
determined by convolving x ( t ) with h(t). If h(t) could be determined from H ( j w ) ,
then H(jo)would also completely characterize a stable LTI system because, once
known, h(t) could be determined. Now the system gain is equal to the magnitude of
H ( j o ) and the phase shift is equal to the angle of H ( j w ) so that once the gain and
the phaseshift are known, Eq. (4.214) can be used to determineH(jw). This means
that the gain and the phase shift would also completely characterize a stable LTI
system.
From a mapping viewpoint, we can view Eq. (4.18) as a mapping of h(t) into
H(jo).The obtaining of h(t) from H ( j w ) is then the inverse mapping. As we
126 THE FREQUENCY DOMAIN VIEWPOINT
discussed in Section 1.1, the inverse of a mapping exists if and only if the mapping is
onetoone. This means that if h(t) can be determined from H ( jo),then the mapping
is onetoone so that no two different unitimpulse responses can have the same
transfer function. Consequently, the system gain and phase shift would uniquely
determine the system unitimpulse response. Note that this would imply that we
cannot arbitrarily specify a desired system gain and phase shift of a causal LTI
system because the corresponding system unitimpulse response may not be zero
for t < 0. Before examining this and other consequences, we shall show that the
mapping is indeed onetoone by showing that h(t) can be determined from H( j w ) .
The results we shall derive are often obtained by reference to mathematics texts
because they are part of the mathematical theory of Fourier transforms. However, to
gain a better appreciation of system concepts, I have developed the following deri
vation using only the system theory results we have obtained.
Equations (4.43) and (4.44) now will be used not only to show that h(t) can be
determined from H ( jo), but also to determine an equation by which it can be
determined. For this, consider the case for which
Then
To evaluate this integral, we first complete the square to express the parenthetical
expression in the exponent in the form
(4.4 10)
(4.41 1)
The integral in Eq. (44 11) is now in a standard form that is listed in many tables of
integrals. Its value is ~ , / Z / EWith
. this value of the integral, we then have
(4.412)
4.4 CONTINUOUS FREQUENCY REPRESENTATION OF A WAVEFORM 127
Observe from Eq. (3.31 1) that this is h,(t). Thus we have shown that if
1
C(o)= e(c2'4)w2 then x ( t ) = h,(t) (4.413)
271
We also showed in Section 3.3 that if the input is x(t) = S,(t), then the LTI system
response is y ( t ) = h,(t). Thus we have from Eq. (4.44) that
(4.414)
We now make use of Eq. (3.3l), which states that h(t) = lim,+o h,(t). For this, first
note that lims+o e(E2/4)w2= 1 for all values of w . Consequently, by taking the limit
as E f 0 in Eq. (4.414), we obtain'
Thus we have shown that the unit impulse response, h(t), of a BIBOstable LTI
system can be determined from its transfer function by use of Eq. (4.4 15). Accord
ingly, as we discussed above, no two different stable LTI systems can have the same
transfer function because the mapping is onetoone. The transfer function thus
completely characterizes the inputoutput mapping of a stable LTI system because,
once known, the system output for any input can be determined. One way this can be
done is to first use Eq. (4.4 15) to determine h(t) and then convolve the system input
with h(t) to determine the system response. Later, we shall discuss other methods.
Because the system gain and phase shift are the magnitude and angle of the
transfer function, the gain and phase shift also completely characterized the input
output mapping of a stable LTI system. The gain and phase shift can easily be
determined experimentally by using a sinusoid as the input of the stable LTI
system. As we discussed, the output is a sinusoid with the same frequency as the
input sinusoid. The ratio of the amplitude of the output sinusoid to that of the input
sinusoid is the system gain at the frequency of the sinusod, and the phase difference
between the output and input sinusoids is the system phase shift at the frequency of
the sinusoid. The gain and the phase shift can be measured and plotted as a function
of the sinusoidal frequency.
'Note from Eq. (4.414) that we should integrate first with respect to w and then take the limit as E goes to
zero. By writing Eq. (4.415), we have really taken the limit as E goes to zero before integrating with
respect to w . Mathematically, this requires that H ( j w ) go to zero as o becomes arbitrarily large. Now, the
faster a waveform wiggles, the higher the frequency content of the waveform. Thus, in order that H ( j w )
go to zero as w becomes arbitrarily large, we require that h(t) not be infinitely wiggly in any interval. Such
a iknction is called by mathematicians a function of bounded variation, so that the requirement is that h(t)
be a function of bounded variation. Physically, this means that we require that the system gain must
approach zero as w tends toward infinity This condition is satisfied by all physical systems.
128 THE FREQUENCY DOMAIN VIEWPOINT
The result we have just obtained can be viewed purely as a mathematical result
without any regard to its system theory source. For the statement of the mathematical
result, I'll call the h c t i onf(t) instead of h(t). The mathematical statement that we
have obtained in this section is as follows:
Iff(t) is an L, function of bounded variation so that
(4.416)
(4.41 7)
converges for all values of o andf(t) can be retrieved from F( j o ) by evaluating the
integral
f ( t )=
271
1
bo
bo
F( jo)d"' d o (4.41 8 )
PROBLEMS
41 Let
F( j o ) =
3 +jo
3 3 9
Determine IF( jo)l and i F(jo)for w = 0,   , and 3.
4' 2' 4
42 Let
for w = 0, 1, 2, 3, and 4.
44 The unitimpulse response of a given LTI system is h(t) = r(t/3). Determine
the system transfer function, H ( j w ) , and determine the system response, y(t),
to the input
x(t)
K 3
=Acos  t +  +Bsm 2nt+
. [ 3
45 The unitimpulse response of a given LTI system is h(t) = d ( t )  2eP2'u(t).
Determine the system transfer function, H ( j w ) , and determine the system
+
response to the input x ( t ) = A Bcos(2t).
46 The unit impulse response of a given LTI system is h(t) = [4eC2'  2e']u(t).
Determine the system transfer function, H ( j w ) . What is an approximate
expression for the gain and phase shift for w >> 2?
2
47 The transfer function of a given LTI system is H ( j w ) = 
(a) Determine the system gain.
3 +jw'
(b) Determine the system phase shift.
+ +
(c) Determine the system response to x ( t ) = A B sin(3t n/4).
410 The response of a given LTI system to the input x ( t ) is y(t) = Ax(t  to).
Determine the system gain and phase shift.
x ( t )=A+Bc os t  
( t) +Csm 3t+
. ( 3
Determine the response, y(t), of the LTI system with the unitimpulse
response
412 The unitimpulse response of an LTI system is h(t) = e" cos(oot 4)u(t), +
where CI > 0. Determine the system transfer hnction H(jw). Hint: Use
Eq. (A14) to express h(t) as the sum of exponentials so that you can
evaluate the integral. Two special cases are h,(t) = ePatu(t) and hb(t) =
e? sin(w,t)u(t). Does your result agree with these two special cases?
413 Use Eq. (4.216) to obtain the result given by Eq. (4.25).
414 (a) Use convolution to show that the response of a stable LTI system to the
input x(t) = E is y(t) = EH(O), where H(0) is equal to the area under the
system unitimpulse response, h(t).
(b) Show that this same result is obtained from Eq. (4.216).
416 Each component system in the diagram below is a stable LTI system.
0
....... ........................................hi.........................................................,
j_.
PROBLEMS 131
(a) Show that overall system with the unit impulse h(t) is stable.
(b) Determine h(t) in terms of the unitimpulse responses of the component
systems.
(c) Determine the transfer function of the overall system in terms of the
transfer functions of the component systems. For this determination, use
the technique used to obtain Eq. (4.33).
417 Show that Eqs. (4.313a) and (4.313b) are equivalent expressions for y(t).
3a2
H(jw)=
(a +jw)(3a + j w )
419 A lowpass stable LTI system with the transfer function H,(jo)= a / ( a + j w )
is connected in tandem with a lowpass LTI system with the transfer function
H ( j w ) = b/(b + j w ) . As discussed in Section 4.3, the bandwidth of the
tandem connection is 0 . 6 4 ~if b = a. What positive value of b is required for
the bandwidth of the tandem connection to be 0.8a?
H ( j w )= 1
00
00
h(t)e'"' dt.
(b) Thus show that if the two conditions stated in the problem are satisfied,
then limw+ooH ( j w ) = 0.
422 Use Eq. (4.418) to determine the time function,f(t), whose Fourier trans
form is F( j w ) = e + .
CHAPTER 5
133
134 THE FOURIER TRANSFORM
Later in this text we shall develop and discuss the bilateral Laplace transform in
some detail. The Fourier transform will be shown to be a special case of the bilateral
Laplace transform. Thus, in this chapter we shall only develop and discuss some
aspects of Fourier transform theory that will help enhance our understanding of the
frequency domain view of LTI systems.
(5.11)
F(jw)= 1 03
00
f(t)ejw' dt (5.12)
converges for all values of w and alsof(t) can be retrieved from F ( j w ) by evaluating
the integral
f ( t )=
2n
1
l o o
F(jw)ejw' do (5.13)
(5.14)
Admittedly, this integral is not easy to evaluate without the use of a special integra
tion technique called contour integration. However, we need not evaluate this inte
gral because the fact that Eqs. (5.12) and (5.13) are a Fourier transform pair means
5.2 AN EXAMPLE OF A FOURIER TRANSFORM CALCULATION 135
that we know that the value of the integral is as given. Because of the difficulty often
incurred in evaluating the integral in Eq. (5.13), tables of Fourier transforms have
been developed by calculating F( jo)using. Eq. (5.12) and knowing that the value
of the integral in Eq. (5.13) will be the same function, f(t).
Note that the requirement thatf(t) be an L , function is not a necessary condition
but only a sufficient condition. We only showed in Section 4.4 that iff ( t ) is an L ,
function of bounded variation, then Eqs. (5.12) and (5.13) definitely are a Fourier
transform pair. However, there are some functions of bounded variation which are
not L , functions for which Eqs. (5.12) and (5.13) are a Fourier transform pair.
To illustrate our discussion in the previous section and to present some basic aspects
of the Fourier transform, consider the case for which f(t) is the rectangle:
E if  T < t t T
(5.21)
0 otherwise
In this equation, r(.) is the rectangular function defined by Eq. (1.54). Using Eq.
(5.12), the Fourier transform of this function is
t+T .
F( j w ) = dt
02
(5.22)
2j sin(oT) 2 sin(wT)
=E =E
.io w
sin(wT)
F( jo)= 2TE ~ (5.23)
0T
The equation for F(jo) is similar to that of the transfer function determined in
Section 4.2, Eq. (4.223), where a graph of (sin8)/8 is given and discussed.
Observe from Eq. (5.12) that the value of the Fourier transform at o = 0 is
In words, the value of F(0) is equal to the area underf(t). This usually is an easy
check on your calculation of a Fourier transform. That is, your expression for F( jo)
is wrong if Eq. (5.24) is not satisfied. Unfortunately, it is possible that your expres
sion for F(jo)is incorrect and yet Eq. (5.24) is satisfied so that you only can say
that your expression is wrong if Eq. (5.24) is not satisfied. For our case from Eq.
(5.23), we have F ( 0 ) = 2TE, which is equal to the area of the rectangle,f(t), given
by Eq. (5.21).
Note for our example thatf(t) = S,(t) if E = 1 / and ~ T = ~ / 2 .Thus, from Eq.
(5.23), the Fourier transform of S,(t) is
(5.25)
We obtain the unit impulse for the case where E = O+ so that, from Eq. (5.25) we
obtain
The impulse width is infinitesimal, so that the interpretation of Eq. (5.26) is really
that it is Eq. (5.25) with E = O+. Observe that this result also can be obtained
directly by using the sifting property of the impulse developed in Section 2.4 in
Eq. (5.12). From our discussion in Section 5.1, S(t) and 1 are a Fourier transform
pair. This result is consistent with our discussion of S,(t) in Section 4.4.3
In accordance with our discussion in Section 5.1, the inverse Fourier transform of
F ( j w ) given by Eq. (5.13) should bef(t) given by Eq. (5.21). As an illustration,
we shall verify this in Section 5.4 for our example by substituting the expression for
F ( j o ) given by Eq. (5.23) in the expression for the inverse Fourier transform, Eq.
(5.13), and evaluating the resulting integral. For this calculation, we shall make use
of some basic properties of even and odd functions. Thus we shall discuss a few of
the basic properties of these functions in the next section.
To be mathematically precise, Eq. (5.26)really cannot be one for all values of w but must go to zero in
accordance with Eq. (5.25) because E = O+ # 0. However, in accordance with OUT discussion in the
footnote in Section 4.4 following Eq. (4.414). we require that all transforms tend to zero as w tends to
infinity as a sufficient condition to obtain Eq. (4.415) from (4.414). This is equivalent to using Eq.
(5.26).
5.4 AN EXAMPLE OF AN INVERSE FOURIER TRANSFORM CALCULATION 137
are those depicted in Figs. 1.46 and 1.47, and other examples of odd functions are
those depicted in Figs. 1.41, 1.44, 1.48, 1.49, 1.410, and 1.41 1. Note that the
functions depicted in Figs. 1.43 and 1.45 are neither even nor odd. However, a
function that is neither even nor odd always can be expressed as the sum of an even
and an odd function. First note that, for any function,f(t),f(t) + f (  t ) is an even
function o f t whilef(t)  f (  t ) is an odd function oft. Thus we can expressf(t) as
the sum of an even function, f ,(t), and an odd function, fo(t):
f ( t ) =f,(t>+h@) (5.31)
where
= $ [ f ( t >+f(Ol (5.32a)
and
The decomposition of functions into their even and odd components as above leads
to some important properties of Fourier transforms and LTI systems which we shall
discuss later in this chapter. Even and odd functions also are very useful in integra
tion. One important property that we shall use in the next section is that the area
under an odd function from co to +co is zero. This is easily seen because the area
under an odd function from co to 0 is the negative of the area from 0 to +co. That
is,
w
Also, the area under an even function from co to +co is twice the area under it
from 0 to +oo since the area under an even function from co to 0 is equal to that
from 0 to +co. That is,
(5.33b)
In accordance with our discussion in Section 5.1, the inverse Fourier transform of
F ( j w ) given by Eq. (5.13) should bef(t) given by Eq. (5.21). We shall verify this
for our example in Section 5.2 by substituting the expression given by Eq. (5.23) in
the expression for the inverse Fourier transform, Eq. (5.13), and evaluating the
resulting integral. For this calculation, we shall make use of the properties of even
and odd functions given by Eqs. (5.33).
138 THE FOURIER TRANSFORM
Substituting the expression for F(jo),Eq. (5.23), in the expression for the
inverse Fourier transform, Eq. (5.13) we obtain
O0 sin(wT) .
e'"' do (5.41)
From Appendix A, e'"' = cos(ot) + j sin(wt) so that we can express this integral in
the form
cos(ot) do +j"
n
1O0
oo
sin(wT) .
~
oT
sin(wt) d o (5.42a)
cos(ot) do (5.42b)
Equation (5.4213) was obtained by noting that the value of the second integral in Eq.
(5.42a) is zero. To see this, note that the hnction being integrated is an odd function
of w. Thus, with the use of the result given in Eq. (5.33a), we have that the value of
the integral is zero. Now, to evaluate the integral in Eq. (5.42b), we first note that the
hnction to be integrated is an even hnction of o so that, from the result given in Eq.
(5.33b), we have
do (5.43)
This form of the integral allows us to use a table of integrals in which can be found
the definite integral
(5.46)
1 ifcl<O
(5.47)
1 ifct>O
5.5 SOME PROPERTIES OF THE FOURIER TRANSFORM 139
(5.48)
Note that this is the same as the functionf(t) with which we started. Observe that
f(t) is discontinuous at the points t = &T and the value of the inverse Fourier
transform is equal to the average of the left and righthand limits off(t). In general
at points of discontinuity of a functionf(t), the inverse Fourier transform of any
function will be equal to the average of the left and righthand limits of the function,
f (t). This is in keeping with our discussion in Section 2.4 where we discussed left
and righthand limits and our definition of functions at a discontinuity following Eq.
(2.43).
The computation of the inverse Fourier transform for this example was not
simple. This is generally true in the computation of the inverse Fourier transform
using Eq. (5.13). However, as discussed in Section 5.1, Eqs. (5.12) and (5.13) are
a Fourier transform pair iff(t) is an L , function. We can consider Eq. (5.12) as
mapping f ( t ) into F( j w ) and consider Eq. (5.13) as mapping F( jo)into f ( t ) . The
two equations being a Fourier transform pair means that the mapping is a onetoone
mapping. Thus once a given F( jo)is computed from a givenf(t), we know that the
use of Eq. (5.13) will result in the samef(t) with which we started. It is for this
reason that tables of Fourier transforms are so useful. However, a few properties of
Fourier transforms are required to use them effectively. These required properties are
derived in the next section. They then will be illustrated in the following sections by
applying them to the analysis of LTI systems.
Properties of the Fourier transform greatly assist in their physical interpretation and
use as well as in their determination. Several important properties are derived in this
section. Then, some applications of these properties are illustrated in the next
section. For our discussion in this section,fi(t) andf,(t) are L , functions of bounded
variation, and their Fourier transforms are F , ( j w ) and F 2 ( j o ) ,respectively. Also, c1
and c2 are constants.
140 THE FOURIER TRANSFORM
(5.5 1a)
then
(5.5 1b)
We show this by replacing o with o and taking the conjugate of Eq. (5.12) to
obtain
F*(jo) = [ oo
f(t)e+'"' dt]* (5.53)
F*(jo) = / 00
oo
f*(t)e'"' dt
(5.54)
if and only iff(t) is a real function oft. Another form of Eq. (5.55a) is obtained by
taking its conjugate to obtain
Observe from this result that iff(t) is a real function o f t , then the magnitude of
F( jo)is an even function of o and the angle of F( jo)is an odd function of o.
B. For the case in whichf(t) is a real function oft , there is a second important
symmetry property that we need. For this we expressf(t) as the sum of its even and
odd parts as discussed in Section 5.3. Then, with the use of Eq. (5.31) and the
linearity property, Eq. (5.5lb), we can write the Fourier transform off(t) as
wheref,(t) andf,(t) are real functions o f t becausef(t) is a real function oft. Now,
00 00
f,(t)[cos(wt)  j sin(ot)] dt
00 00
(5.57)
00
00
=
J, f , ( t )cos(ot) dt
The imaginary part of g{f,(t)} in Eq. (5.57) is zero in accordance with Eq. (5.33a)
becausef,(t) sin(@ is an odd function of t. Note from this result that the Fourier
transform off,(t), S(f , ( t ) } ,is a real function of o and that it is an even function of o
because cos(ot) is an even function of o.
We now examine S ( f , ( t ) )the , second term of Eq. (5.56).
00 00
g { f , ( t ) }=
00
f,(t)e""' dt = J'f,(t)[cos(wt)
,
 j sin(ot)] dt
00 00
= J'
00
f , ( t ) cos(ot) dt  j (5.58)
= j 1 00
03
f,(t) sin(wt) dt
The real part of 5(f;,(t)) in Eq. (5.58) is zero in accordance with Eq. (5.33a)
because f , ( t )cos(wt) is an odd function of t. Note from this result that the Fourier
transform ofJL(t), S ( f , ( t ) ) ,is an imaginary function of o and that it is an odd
function of w because sin(wt) is an odd function of w.
142 THE FOURIER TRANSFORM
where F,( jo)is the real part and Fi(j w ) is the imaginary part of F( jo).Then from
Eqs. (5.56), (5.57), and (5.58) we observe that iff(t) is a real function oft, then
00
and
F , ( j w ) = j g [ f , ( t ) }= 1
00
00
f,(t)sin(ot) dt = 2 f , ( t ) sin(ot) dt (5.5lob)
from which we note that F,( jo)is an even function of o and Fi( j w ) is an odd
function of o.
(5.51 la)
then
(5.51 lb)
where IC( is the absolute value of the constant [see Eq. (1.425)].
To prove this property, first consider the case for which c > 0. Substituting Eq.
(5.51 la) in Eq. (5.12), we have
F( j w ) = 1 fi
00
00
(ct)ejwtdt (5.5 12)
To put this equation in the form of the Fourier transform offi (t),we make the change
of variable z = ct to obtain
(5.5 13)
5.5 SOME PROPERTIES OF THE FOURIER TRANSFORM 143
This is Eq. (5.51 lb) for c > 0. For the case in which c < 0, the change of variable
z = ct in Eq. (5.58) results in
(5.5 14)
= F,(
1 j:)
C
then
then
F(jo)= 1 fi
00
mJ
(t  to)ejwfdt (5.517)
This equation can be put in the form of a Fourier transform offi ( t ) with the change
of varible z = t  to to obtain
(5.518)
144 THE FOURIER TRANSFORM
This is Eq. (5.516b) and so we have proven the timeshift property. Note that to can
be positive or negative.
then
Note that the frequencyshift property is the dual of the timeshift property. This
duality is to be expected since the equation for the Fourier transform, Eq. (5.12) and
the equation for the inverse Fourier transform, Eq. (5.13), are almost the same. They
differ only in a factor of 271 and in a minus sign in the exponent of e. Consequently,
we expect every Fourier transform theorem to have a dual. This stimulates us to
attempt to determine and prove dual theorems. Looking for and exploiting simila
rities of the form of equations is an important way that new theoretical results are
obtained.
The frequencyshift property is easily proven by use of Eq. (5.12) because
00
F( jo)=
S, f i (t)ejwO'ejw' dt
J ,
then
The convolution property was already obtained and proved in Section 4.3, Eq.
(4.34), where we examined the transfer function of the tandem connection of two
stable LTI systems. We proved there that the convolution of two L , functions is an L ,
function. Thus the Fourier transform off(t) exists and, in accordance with the result
obtained in Section 4.3, Eq. (4.34), is given by Eq. (5.520b).
5.5 SOME PROPERTIES OF THE FOURIER TRANSFORM 145
(5.52 1a)
To prove this property, we again begin with Eq. (5.12) and obtain
This integral can be expressed in the form of the Fourier transform offi ( t )by using
integration by parts, which is
(5.523)
for which
du = joeJ''' dt and fi
u = (t)
Now the first term is zero becausefi(t) is an L , function. To see this, first note that
Ifi(t)e'"'I = I,fi(t)Ilej"'I = Ifi(t)l because leJw'I = 1. Now theareaunder Ifi(t)l
is finite because it is an L , function. Thus limt+*OOIfi(t)l = 0 otherwise the area
under Ifi(t)l would be infinite. We then have from Eq. (5.524)
We showed in Section 4.3 that if two stable LTI systems with the unitimpulse
responses h,(t) and hb(t) are connected in tandem as shown in Fig. 4.31, then
the tandem connected system is a stable LTI system with the unitimpulse response
(5.61)
A problem that occurs is the following. We are given a stable LTI system with the
unitimpulse response h,(t). We desire to connect in tandem with it a stable LTI
system with the unitimpulse response hb(t) to form a stable LTI system with a
certain desired unitimpulse response, h(t). What is the required unitimpulse
response hb(t)? Here we are given h(t) and h,(t), and the function to be determined
is hb(t) in Eq. (5.61). Such an equation is called an integral equation because the
function to be determined is part of the function being integrated. Integral equations
often are difficult to solve. However, an integral equation of the convolution type is
not difficult to solve because, by use of the convolution property, Eqs. (5.520), we
have
(5.63)
1 1
H(jo)=  and H,(jo)= (5.65)
B +.io a+jw
a +jw
Hb( jo)=  (5.66)
B +jw
5.7 AN APPLICATION OF THE TIME AND FREQUENCYSHIFT PROPERTIES 147
Hb( j w ) = 1 .P
+ (5.67)
P +ju
Thus, by use of the linearity property, Eqs. (5.51), together with the Fourier trans
forms given by Eqs. (5.26) and (4.116), we have
You should convince yourself of the correctness of this result by convolving the
given h,(t) with the hb(t) we have just determined to show that the result is the given
h(t). We’ll discuss this important technique in more detail as part of our discussion
of the bilateral Laplace transform.
Let the unitimpulse response of a desired stable LTI system be h(t) and let its
transfer function be H ( j w ) . Sometimes, h(t) of the desired system is not zero in
the interval to < t < 0, so that the desired system is not causal. Thus, instead of
constructing the desired system, we construct one with the unitimpulse response
hd(t) = h(t  to), which is zero for t < 0. How is the gain and phase shift affected?
To answer this, we use the timeshift property, Eqs. (5.516), from which we have
Thus we note that the gain is unaffected. However, the difference between the phase
shift of the constructed system and that of the desired system is (uto).A graph of
this phase difference versus w is a straight line with a slope of to. Thus we note in
accordance with our results in Section 1.4 that a phase shift that is proportional to
frequency corresponds to a time shift that is equal to the slope of the straight line. A
negative slope corresponds to a delay, and a positive slope corresponds to an
148 THE FOURIER TRANSFORM
advance. It is for this reason that, to eliminate phase distortion of a filter, one
attempts to make the filter phase shift proportional to w within the system pass band.
The dual of the timeshift property is the frequencyshift property given by Eqs.
(5.519). Consider an L , waveformf(t) with the Fourier transform F ( j o ) . For
example, the waveform could be that of a musical composition for which the spec
trum is in the audio band (which is less than about 25 kHz). In accordance with our
discussion of symmetry properties in Section 5.5, F ( j w ) will extend from about
25 to +25 kHz. Such a waveform cannot be transmitted efficiently by radio. One
reason is that, for efficient transmission, the length of the transmitting antenna
should be on the order of a wavelength of the waveform being transmitted (for
example, the length of an efficient short dipole is 1/2 wavelength). The relation
between wavelength, 1, and frequency,f,is 1,= c, where c is the velocity of light
(approximately 3 x 10' m/s in the atmosphere). Thus the wavelength corresponding
to a frequency of 100 MHz is 3 m, (100 MHz is in the middle of the FM band). To
utilize an FM antenna, we move the spectrum center off(t) to 100 MHz. For this, we
form the waveform g(t) by multiplyingf(t) by a 100MHz sinusoid as
in which, for our example, oo= (271) x 10' rad/s. The waveform g(t) then can be
transmitted efficiently with an antenna of reasonable size. This is a form of ampli
tude modulation. It is called amplitude modulation because the amplitude of the
sinusoid is being modulated (Le. altered) byf(t). The Fourier transform of g(t) is
easily obtained with the use of the frequencyshift property. For this, we first express
the cosine in exponential form, which, from Eq. (A14), is
The expression for g(t) is now in the form required to directly apply the frequency
shift property, Eqs. (5.519), from which we obtain
The first term of this equation is the spectrum off(t) centered at w = coo, and the
second term is the spectrum off(t) centered at o = ao.The second term is due to
the second term of Eq. (5.79, which is required because cos(oOt)is a real function
of t . You should show that G( jo) satisfies the symmetry properties, Eq. (5.55),
because g(t) is a real function oft. The result given by Eq. (5.77) is often called the
modulation theorem because it relates to the Fourier transform of an amplitude
modulated waveform.
5.8 AN APPLICATION OF THE TIMEDIFFERENTIATION PROPERTY 149
E
f ” ( t ) =  (d[t
2a
+ ( T + 2a)]  d [ t + TI  d[t  T ] + d[t  (T + 2~()]) (5.81)
Now, from Eq. 5.26, g(d(t)}= 1 so that, with the use of the timeshift property, we
have that the Fourier transform of a unit impulse centered at t = to is
Thus we have
This expression can be put into a nicer form by grouping the terms as follows:
(5.84)
E
=  (2jsin(cm)){2jsin[w(T
2a
+ a)])
150 THE FOURIER TRANSFORM
2E .
( j o ) ’ ~ ( j w=
)  sin[oa] sin[o(T
a
+ a)] (5.85)
The algebraic manipulations have been presented in some detail so that you can
observe how an expression can be manipulated to put it in a nice form. As a check
on our work, note that F(0) = 2E(T + a). This is equal to the area under f ( t ) in
accordance with Eq. (5.24).A second check of our result is obtained by noting that
f ( t ) is a rectangle for the special case in which a = 0. Note, for this special case, that
our expression is identical with that we previously obtained in Section 5.2, Eq.
(5.23). These two checks do not guarantee the correctness of our expression, but
it does give us good confidence in our result.
As a second example, we shall use the timedifferentiation property to determine
the Fourier transform of
T (T+2a)
E/2a
With the use of the timedifferentiation property and Eq. (5.82), the Fourier trans
form of Eq. (5.88) is
E
G ( j o )= ~ (5.8 10)
CY +jo
This is the same result we previously obtained in Section 4.1, Eq. (4.11 6), by direct
integration.
The essence of the method is to differentiate until either (a) all impulses are
obtained as in the first example or (b) an impulse plus a function are obtained as
in the second example. If all impulses are obtained, the Fourier transform is obtained
with just a bit of algebra as in our first example. If an impulse plus a function are
obtained as, for example, A6(t) +p(t), then the Fourier transform of p ( t ) can be
obtained either by direct integration, or by differentiating p(t), or by forming a
differential equation as in our second example, Eq. (5.88). The technique of obtain
ing the Fourier transform of a function by differentiation is seen to be very useful,
especially because using direct integration to obtain the Fourier transform can be a
bit tedious.
The timedifferentiation property also can be used to determine a relation between
the smoothness of a time function and the asymptotic behavior of its Fourier trans
form. Consider an L , function for which the first n  1 derivatives contain no
impulses and the nth derivative contains K impulses. We then can express the nth
derivative off(t) as
so that
(5.8 12)
By use of the timedifferentiation property, we just showed that the smoother a time
function, the faster its Fourier transform goes to zero with increasing 0. By use of
the scaling property, we also note that the width of a time function and the width of
its Fourier transform are inversely related. Because the transfer function of an LTI
system, H ( j w ) , is the Fourier transform of its unitimpulse response, h(t), the result
shown in this section means that the bandwidth of an LTI system is inversely related
to the time width of its unitimpulse response. For example, consider a stable LTI
system with the unitimpulse response h,(t) = e'u(t). The system transfer function
is
H I(j w ) = / 00
00
h , (t)ejw' dt = e'ejw' dt
(5.91)
Let us now vary the time width of h,(t) by letting h(t) = ,$,(at), where a > 0. Then
Note that the time width of h(t) is proportional to l l a . For example, if we define the
time width to be the time at which h(t) drops to l / & of its maximum value, then
the time width is T = ln(2)/2a = (0.34657)la. Now, using the scaling property of
the Fourier transform, we have
a
(5.93)
This, of course, is the result we have previously obtained by direct integration, Eq.
(4.116). The gain of this system is
(5.94)
The graph of the gain is a bellshaped curve with a maximum value equal to l / a at
w = 0. As discussed in Section 4.3 following Eq. (4.313), the gain drops by 3 dB
(that is, to 1/& of its maxuum value) at o = a. Thus the system is is a lowpass
filter with a 3dB bandwidth, B, equal to a. Using the time width, T = (0.34657)Ia
defined above, we have that the product of T , the time width of h(t), and B, the
system 3dB bandwidth, is TB = 0.34657. Thus we observe that the bandwidth is
inversely proportional to the timewidth of h(t).
The inverse relation between width of a time function and the bandwidth of its
transform imposes certain tradeoffs in system design. For example, one method of
transmitting data from one computer to another is to send a sequence of pulses along
a transmission line in which a pulse represents one of the binary values. In order for
the receiving computer to determine the pulse sequence that was sent, it is necessary
that the pulses not overlap very much. This means that the time from the start of a
pulse to the start of the next pulse, T,, must be proportional to the pulse width, Tp.
Now the data rate, which is the number of pulses per second that can be sent, is
N = I / T , . Because T, is proportional to Tp,we have that N is proportional to l / T p .
In accordance with our discussion above, the pulse width, Tp, is inversely propor
tional to its bandwidth, B, so that B is proportional to l/Tp. Because I/Tp is
proportional to N , we observe that the data rate is proportional to the required
transmission line bandwidth so that the higher the data rate, the larger must be
the transmission line bandwidth. This is one of the reasons why fiberoptic transmis
sion lines have replaced coaxial transmission lines for communication between
computers.
154 THE FOURIER TRANSFORM
In this section we shall derive and discuss an important and useful relation called a
Parseval r e l a t i ~ nThis
. ~ relation will be derived using the results we have already
obtained as a further illustration of their application.
For the derivation of the Parseval relation, we consider two complex L , functions,
f i ( t ) andf,(t). The convolution offT(t) withf,(t) is
The Fourier transform of g(t) is, in accordance with the convolution property, Eq.
(5.520),
Now, to obtain the Fourier transform offr(t), we first use the symmetry property
given by Eq. (5.54) to obtain
Then, with the use of the scaling property with c = 1, Eq. (5.515), we obtain
Now, the inverse Fourier transform of G( j w ) evaluated at t = 0 is, from Eq. (5.13),
g(O) =
2n
J 00
G ( j w ) dw
(5.106)
4MarcAntoine Parsevaldeschknes, 17551836. Many relations of the form discussed in this section are
called Parseval relationsalthough many only remotely resemble Parseval’s original result, which he
considered only as a formula for summing certain types of series. His result was later extended to Fourier
theory and more abstract treatments of analysis. Many of the relations so obtained are called Parseval
relations. The relation obtained in this section is such an example.
5.10 A PARSEVAL RELATION AND APPLICATIONS 155
Because both Eqs. (5.106) and (5.107) are equal to g(O), we have (with the use o f t
instead of as the dummy variable of integration)
This is the desired Parseval relation. An important special case is that for which
fi 0)= m= A t ) is
(5.109)
This last relation is often called the energy theorem. To understand the reason for
this name, let f ( t ) be the current through a 1R resistor. Then the total energy
dissipated in the resistor is given by the lefthand side of Eq. (5.109). Because
the righthand side of this equation is the integral over all frequencies of
1/274F( jo)12,we can interpret it as an energy density spectrum in joules/radian
per second. Equivalently, because one radian per second equals 27~hertz, we can
interpret IF( jw)I2 as an energy density spectrum in joules/hertz. For this reason,
IF( jw)I2 is often called the energy density spectrum off(t).
To make the interpretation of an energy density spectrum more concrete, consider
an ideal method to measure the energy density spectrum of a waveform, x(t). Ideally,
to measure the energy of x ( t ) in the band of frequencies 0 < o < ol, we would
apply x ( t ) to an ideal lowpass filter with the cutoff frequency wl. An ideal lowpass
filter is one that has unity gain in the passband and has zero gain for all frequencies
above the cutoff frequency. That is,
(5.10 10)
As shown in Fig. 5.101, the output of the ideal lowpass filter is a voltage
which is applied to a 142 resistor. In accordance with the convolution property,
Eq. (5.520), the Fourier transform of the output waveform, y(t), is
156 THE FOURIER TRANSFORM
Y ( j w ) = H ( j w ) X ( jo).Thus, with the use of the energy theorem, Eq. (5.109), the
total energy dissipated in the 10 resistor is
(5.101 1)
Thus, with the use of Eq. (5.10lo), the total energy dissipated in the resistor is
(5.10 12)
Similarly, to measure the total energy contained by the waveform x(t) in the
frequency band o,< o < 0 2 ,we would use an ideal bandpass filter with unity
gain in the given band and zero gain outside the given band. In accordance with Eq.
(5.101 l), the total energy dissipated by the 1R resistor is then
00
00
k(t)l2dt =1' 271
WI
o*
IX(jo)I2do IX(jo)12 d o (5.1013)
(5.1015)
J W Jr,
5.1 0 A PARSEVAL RELATION AND APPLICATIONS 157
forO5tsT
(5.1016)
0 otherwise
Rather than construct an LTI system with the desired unitimpulse response, it is
simpler to approximate the desired system by constructing an LTI system with the
unitimpulse response
We use the integral square error as the measure of this difference, which is
00
E= [
J 00
IE(jw)12 d o (5.1018b)
Now
(5.1019)
Rather than evaluating this integral, we shall determine the transfer fimction by using
the Fourier transform properties derived in Section 5.5 in order to further illustrate
how they can be used to simplify calculations. You should draw diagrams of the
various functions discussed below in order to fully understand the various equations.
We begin by differentiating hd(t), which is
where
(5.1020b)
158 THE FOURIER TRANSFORM
1
a t ) =p ( t )  w  TI1 (5.1022)
(5.1023)
so that
(5.1024)
(5.1025)
You should verify this result by actually evaluating the integral in Eq. (5.1019) and
note how much effort was saved by using the Fourier transform properties. Now,
from Eq. (4.116), the Fourier transform of h,(t) is
(5.1026)
The error, E, can now be evaluated by substituting the expressions for H d ( j w )and
H,(jw) into Eqs. (5.1018) and evaluating the resulting integral. This approach
unfortunately leads to integrals that are difficult to evaluate. A much better approach
is to use the energy theorem, from which we have
E= 100
00
IE(jw)I2 dw = 271 1
00
oo
le(t)I2 dt (5.1027)
5.10 A PARSEVAL RELATION AND APPLICATIONS 159
where
Thus we have
00
le(t)t2 dt = 271 1 00
oo
Ihd(t)  h,(t)I2 dt
= 271 1
00
00
hi(t) dt + 271 100
00
hi(t) dt  471 1
00
00
h,(t)h,(t) dt
(5.1029)
Now
00
hi(t) dt = 1: [ 1 ]; 1
dt = T (5.1029a)
00
A2
dt =  (5.1029b)
2a
and
(5.1030)
A better form of the error is the normalized error, which is the error E divided by the
integralsquared value of &(jm). With the use of Eq. (5.1029a), this value is
D= 1 00
00
IHd(jm)I2 d o = 271
271
3
(5.1031)
(5.1032)
160 THE FOURIER TRANSFORM
3
E,, = 1 + A2
2
 6e’A (5.1033)
The value of E,, in this expression is the smallest for A = 2 / e = 0.736, for which it
is E,, = 1  6e2 = 0.188. The percentage integralsquare error thus is 18.8%. In
the design of a filter, the transfer function is specified. The system impulse response
of the desired filter may be one that is difficult to construct, and so a system with an
impulse response that is close to the desired one is considered. The method used in
this example is a good method for comparing various approximations and to deter
mine the best values of parameters (such as A and a in the example above) to use in
the approximation.
As a final illustration of an application of the Parseval relation, we illustrate its
use in the evaluation of definite integrals. This application is not directly related to
system theory. However, you may have wondered in the past how many of the
integrals you’ve seen in integral tables were determined. Well, one method is to
use a mathematical result by which the value of the definite integral can be indirectly
determined. One of the mathematical results used is the Parseval relation, Eq.
(5.108). To illustrate its use, consider the evaluation of the integral
sin(aco) sin(bw)
do, a>b>O (5.1034)
To use the Parseval relation, Eq. (5.1049, we first want to extend the lower limit of
the integral to co. For this, we note that the function being integrated is an even
function of w. Thus, from our discussion in Section 5.3 of even and odd functions,
Eq. (5.33b), we have
(5.1035)
Now the Fourier transform of a rectangle was determined in Section 5.2, Eq. (5.23).
From that result, we observe that the function under the integral above is similar to
the product of the Fourier transform of two rectangles. Thus we express the fimctions
being integrated in the form of the product
O0 sin(ao) sin(bo)
do (5.1036)
and h ( t ) = r
:h ('2+hh)
~ (5.103 8)
ab
I = 271
2
1c13
oo
fi*(t&(t) dt
(5.1039)
= 2711
ab 1 O0 r (ty+)ar ( T t) + b dt
2 4ab oo
+ + +
Now note that r[(t a)/2a]r[(t b)/2b] = r[(t b)/2b] because a > b > 0. Thus
the value of the integral in Eq. (5.1039) is 2b because it is just the area of a rectangle
with a height equal to one and a width equal to 2b. We thus have that the value of the
integral, Eq. (5.1035), is
ab 1 71
I=2~2b=b (5.1040)
2 4ab 2
The inputoutput mapping of any BIBOstable LTI system was shown in Section 3.7
to be completely determined by its unitimpulse response, h(t). In Section 3.5, we
discussed the fact that every physical system is causal and we showed that a neces
sary and sufficient condition for an LTI system to be causal is that h(t) = 0 for t < 0.
This condition imposes certain constraints on the transfer function, H ( jo).Some of
the limitations causality imposes on H ( j w ) are discussed in this section. Thus we
consider only physical LTI systems for which h(t) = 0 for t < 0 and is a real
function o f t for our development in this section.
'David Hilbert (1 8621 943) was one of the leading mathematicians of the twentienth century who made
major contributions to many fields of mathematics.
162 THE FOURIER TRANSFORM
relation between the real and imaginary parts of the transfer function is called a
Hilbert transform, which we develop and discuss in this section.
For our development, first express h(t) in terms of its even and odd parts as
where, from our discussion in Section 5.3, the even part of h(t) is
(5.1 12a)
1
h,(t) =  [h(t) h(t)] (5.1 12b)
2
We assume that neither h,(t) nor h,(t) is zero for all t. The resolution of the unit
impulse, d(t), into two nonzero components as in Eq. (5.1 11) is impossible because
as discussed in Section 3.3, the unit impulse, d(t), is an even function. Consequently,
unitimpulse responses, h(t), that contain an impulse at t = 0 are excluded from our
present discussion but will be included at the end of this discussion.
Because h(t) = 0 for t < 0, we have that h(t) = 0, for t > 0. Consequently,
from Eqs. (5.112) we obtain
1
h,(t) = h(t) for t > 0 (5.1 13a)
2
and
1
h,(t) = h(t) for t > 0 (5.1 13b)
2
Note that h,(t) = h,(t) for t > 0 and consequently h,(t) = h,(t) for t < 0. This is
logical because, in order that h(t) = 0 for t < 0, the sum of the even and the odd
parts of h(t) must equal zero for t < 0. This immediately implies that the even and
the odd parts of h(t) must be equal for t > 0, from which Eqs. (5.1 13) follow. Thus
we observe that h,(t) can be determined from h,(t) and also that h,(t) can be
determined from h,(t) of a causal LTI system.
Now, we showed in Section 4.1, Eq. (4.18), that the transfer function of a stable
LTI system is given by the relation
H(jw)= 1 00
bo
h(t)e@” dt (5.1 14)
This is recognized as the Fourier transform of h(t) in accordance with our discussion
in Section 5.1. Again, it should be noted that the transfer function only arose in
5.1 1 TRANSFER FUNCTION CONSTRAINTS 163
H(jo)= H r ( j o )+ j H ; ( j o ) (5.115)
where Hr( jo)is the real part and Hi( jo)is the imaginary part of H( jo).We now
make use of one of the symmetry properties of the Fourier transform we obtained in
Section 5.5, Eqs. (5.510). With the substitution of h(t) forf(t), these equations are
H,(jo) = 1 00
00
h,(t) cos(ot) dt (5.1 16a)
and
00
H,(jo) = 
S, h,(t) sin(ot) dt (5.116b)
Because h,(t) cos(wt) and h,(t) sin(ot) are even functions of t , the integral of these
functions from oo to 0 is equal to their integral from 0 to 00. Consequently, as in
Section 5.3, Eq. (5.33b), we can express Eqs. (5.1 16) as twice the integral from 0
to 00 as
(5.1 17a)
and
H ; ( j w ) = 2
1: h,(t) sin(wt) dt (5.117b)
Because these integrals are only over positive values of t and because from
Eqs. (5.1 13) we have h,(t) = h,(t) for t > 0, we can replace h,(t) by h,(t) in Eq.
(5.1 17a) and we can replace h,(t) by h,(t) in Eq. (5.117b) to obtain
H r ( j o )= 2
I h,(t) COS(W)dt (5.118a)
164 THE FOURIER TRANSFORM
and
H i ( j o ) = 2
:J h,(t) sin(ot) dt (5.118b)
From Eq. (5.13) [also Eq. (4.415)], the inverse Fourier transform of H(jo)is
h(t) =
271
/ 00
m
H(jo)ejw' dt (5.1 19)
+ j)~ ~ ( j o ) ] [ c o s ( o+
H(jo)ejwt = [~,.(jo t )j sin(wt)]
= [Hr(jo)cos(ot)  Hi( jo)sin(wt)] (5.1110)
+ j [ H , ( j o ) sin(wt) + Hi(jw)cos(ot)]
Note that Hr( jo)cos(ot) is an even function of o because both H r ( j o ) and cos(ot)
are even functions of o.Also note that H,(jw)sin(wt) is an even function of w
because both Hi(jo)and sin(@ are odd functions of o and the product of two odd
5.1 1 TRANSFER FUNCTION CONSTRAINTS 165
functions is an even function. Using Eq. (5.33b), Eq. (5.1112) also can be
expressed as twice the integral from 0 to 00:
'J
h(t) =  [H,( j w ) cos(ot)
n o
 Hi( j w ) sin(ot)] dw
= 1
710
00
Note that the first integral in Eq. (5.1 113) is an even function o f t becaue cos(wt) is
an even function o f t and the second integral is an odd function o f t because sin(ot)
is an odd function of t . Thus, with Eq. (5.1 1l), we have
and
Notice that Eqs. (5.117a) and (5.1114a) are a transform pair; also Eq. (5.117b) and
(5.1 114b) are a transform pair. This observation, along with the fact from Eq.
(5.1 13) that he(?)= h,(t) for t > 0, means that H r ( j w ) and H i ( j w ) are related.
To determine this relation explicitly, we substitute Eq. (5.1 114b) in Eq. (5.1 1Sa).
In order not to confuse the w in Eq. (5.1 18a) with the integration variable, o,in Eq.
(5.1 114b), we first use u for the integration variable instead of w in Eq. (5.1 114b)
to express it as
(5.1 115)
These are important equations. Equation (5.1 116a) shows that the real part of the
transfer function can be determined from the imaginary part of the transfer function.
Note that Eq. (5.1116b) is the inverse of Eq. (5.1116a), which states that the
166 THE FOURIER TRANSFORM
imaginary part of the transfer function can be determined from its real part. Trans
forms of this type are known as Hilbert transforms, and Eqs. (5.1 116) are called a
Hilbert transform pair. Thus we have shown that the real and the imaginary parts of
the transfer function are related by a Hilbert transform. Again note that, to this point,
we have excluded from our development unit impulse responses that contain an
impulse at t = 0.
As a simple example illustrating the Hilbert transforms, let the real part of a
transfer function be given as
1
H,.(jo) =  (5.1117)
1+ o 2
We then can determine the imaginary part of the transfer function required for the
system to be causal from the Hilbert transform, Eq. (5.1116b), by substituting Eq.
(5.1117) to obtain
Hi(
jo)=  
n
TIw
0 0
1
cos(ut)
1+u2
sin(ot) du dt (5.1118)
To evaluate this double integral, we first integrate with respect to the variable u. This
integral, whose value can be obtained from a standard table of definite integrals, is
(5.1 119)
Substituting this result in Eq. (5.1 118) we obtain, again with the use of a standard
table of definite integrals,
w
e' sin(ot) dt =   (5.1 120)
1 +o2
By combining Eqs. (5.1 117) and (5.1 120), the transfer function obtained is
(5.1121)
In Section 4.1, this was shown to be the transfer function of a causal and stable LTI
system with the unitimpulse response h(t) = e'u(t).
Even though the integrals in Eqs. (5.1 1 16) are generally not easy to evaluate, the
importance of the Hilbert transform relations for us is that they show that the real
and the imaginary components of the transfer function of a causal and stable LTI
system are related so that they cannot be independently specijed. Equations are
important not just for calculation purposes, but also for their theoretical statements
as in our present instance.
5.1 1 TRANSFER FUNCTION CONSTRAINTS 167
(5.1 122)
where H ( jo)is the system transfer function. For such systems, the theorem states
that the LTI system is not causal if
(5.1 123)
Furthermore, if the value of the integral, I , in Eq. (5.1 123) is finite, then there exists
a phase function Q(w) such that
The PaleyWiener criterion was first published as Theorem XI1 in Paley, R. E. A. C., and Wiener, N. The
Fourier Transjorms in the Complex Domain, American Mathematical Society Colloquium Publication,
Vol. 12, 1934, Chapter I , QuasiAnalytic Functions.
'Although a proof of this result is contained in the work cited in the reference given in footnote 6 , an
easiertofollow proof is contained in Zadeh, L. A,, and Desoer, C. A. Linear System Theoy, the State
Space Approach, McGrawHill, 1963, pp. 423428.
168 THE FOURIER TRANSFORM
theorem, Eq. (5.109), from which we have that the class of LTI systems to which the
criterion applies are those for which
00
It can be shown that Eq. (5.1 125) is satisfied if h(t) is a bounded L , function. This
means that E given by Eq. (5.1 122) is satisfied by LTI systems which are stable so
that h(t) is an L , function and for which h(t) contains no impulses at all (so that h(t)
is bounded). The theorem states that such systems cannot be causal if I given by Eq.
(5.1 123) is infinite.
To examine the restriction imposed by Eq. (5.1 123), first note that it only
involves the system gain, IH(jw)l. Thus the criterion involves a constraint only
on the system gain. First consider the ideal lowpass filter for which the gain is
given by Eq. (5.1010). Such a system cannot be causal because IH(jw)l = 0 for
101 > w , so that I In IH(jw)ll is infinite for 101 > w , and consequently I = 00 in
Eq. (5.1 123). From this example we see that the gain of a causal filter cannot be
zero over any frequency interval. However, the gain of a causal system can possibly
be zero at discrete frequencies.*
Next, consider an LTI system for which the gain is
Such a system also cannot be causal for p 2 1 because we then obtain from Eq.
(5.1 123)
(5.1 127)
From this example we observe that, as w + 00, the gain of any causal LTI system
must go to zero slower than an exponential in wthat is, slower than e P w .
In practical applications, the PaleyWiener criterion is not as restrictive as it first
appears. For example, even though an ideal lowpass filter cannot be causal, we can
make the gain very small for IwI > 0,. For example, consider an LTI system with
the gain
(5.1 128)
in which E is very small (but not zero!). For this example, the value of I given by Eq.
(5.1123) is finite so that a causal LTI system with this gain function does exist.
'The gain can even be zero at an infinite set of discrete frequencies, w = w, for n = I , 2 . 3 , . . .
5.1 1 TRANSFER FUNCTION CONSTRAINTS 169
We note from Eq. (5.1123) that, for the ideal gain functions in our examples, I is
infinite due to the behavior of the system gain over frequency intervals where the
gain is very small. However, as in the example above, even though a causal LTI
system does not exist for such ideal gain functions, it does exist for a system with a
gain function that differs slightly from an ideal gain function only in frequency
intervals where the ideal gain function is very small. This often is an acceptable
approximation.
The relation between causality and prediction was discussed in Section 3.5. Thus
it should not be surprising that there is a close connection between this criterion and
one for the prediction of a waveform. Let the meansquare value of a waveform,f(t),
be finite and let its power density spectrum be @(a). Then it can be shown that the
future off(t) can be completely determined from its own past with arbitrarily small
error if9
(5.1 129)
Note that this is Eq. (5.1123) with IH(o)I replaced by @(w). From our discussion
above, we note that the future of any waveform for which its power density spectrum
is zero in any frequency interval or for which its power density spectrum goes to zero
faster than an exponential in w as w + 00 can be predicted with arbitrary small
error." Thus, the power density spectrum of your speech waveform cannot be
nonzero just in the audio band but must be nonzero even in the microwave band
and for all frequencies above. The power density spectrum will be rather small at
very high frequencies but not zero because it is that small amount of power in the
very high frequencies that makes the prediction error grow with increasing time into
the future at which the prediction is made. If the future of your speech waveform
were predictable with arbitrary small error, then all that you will say in the future is
predetermined and you would not be able to change it. Thus your free will would
definitely be limited. Ethics and morality then become questionable concepts
because without free will how can we hold a person responsible for what he or
she says or does?
For an LTI system to be causal, we require h(t) = 0 for t < 0; and for the LTI
system to be stable, we require h(t) to be an L , function. These are easy constraints
to impose in the design of a causal and stable LTI system in the time domain.
However, the design in the frequency domain is more difficult because the specified
transfer function must satisfy the constraints discussed in this section which are not
simple to apply. Even though these constraints are not simple to apply, they are
important to understand because they identify fundamental limitations on the trans
' Wiener, N. Extrapolation, Interpolation, and Smoothing of Stationary Time Series, The Technology Press
of MIT and John Wiley & Sons, New York, 1949.
l o Schetzen, M., and AIShalchi, A. A. Prediction of Singular Time Functions, M.I.T. Quarterly Progress
Report 67, Oct. 15, 1962, pp. 126137.
170 THE FOURIER TRANSFORM
fer function of a causal LTI system. Design and analysis in the frequency domain,
however, can be greatly simplified by working in a complex frequency plane. For
this we develop the bilateral Laplace transform in the next chapter. We shall see that
the Fourier transform is a special case of the bilateral Laplace transform. The use of
the complex frequency plane associated with the bilateral Laplace transform also
will enable us to gain insight into many frequency domain operations.
PROBLEMS
51 In Chapter 3 it was shown that any positive pulse with an infinitesimal width
can be used as a unit impulse. This was illustrated with the rectangular pulse
in Section 5.2. As another example, consider the function f ( t ) = Ae'IfI,
where a > 0.
(a) Determine F(jo),the Fourier transform off(t).
@) Show that F(0) = Jym f ( t ) dt and determine A so that F(0) = 1.
(c) With the value of A determined in part a, show that the width off(t)
decreases as a increases and that limu,m f ( t ) = s(t).
(d) For a given value of a, for what range of o will 1 p F(jo)p 0.99 so
that f ( t ) will be a very good approximation of the unit impulse in this
frequency range?
52 Determine and sketchf,(t) andf,(t), the even and odd parts respectively of the
following functions.
(a) fi 0) = W T )
(b) h(t)= (1  t / T ) r ( t / T )
(c) h(t)= e+u(t)
(d) h(t)= cos(ot)u(t>
(e) &(t) = sin(ot)u(t)
54 Show that the value of second term of Eq. (5.42a) is zero.
56 (a) Show that iff(t) is a real function (Le., its imaginary part is zero), then
S{f(t)I = F*(jo).
(b) Use the result of part a to obtain the Fourier transform off(t) = $‘u(t)
and verify your result by direct integration.
(c) Use the result of part a to show that the Fourier transform of the even part
of a real function is a real function of w and that the Fourier transform of
the odd part of a real function is an imaginary function of o.
For this determination, use the convolution property together with the result
given by Eq. (4.223).
510 In Section 4.4, we obtained the Fourier transform pair given by Eqs. (4.48)
and (4.412). This was obtained by evaluating the integral, which was not
simple. The same result will be obtained in this problem in a simple manner
172 THE FOURIER TRANSFORM
1
+
F’( jo) wF( jo)= 0
2a2
Because the differential equations in parts a and b have the same form,
their solutions must have the same form. Use this observation to show
that F( jo)= Cew2/4az.
Determine the constant, C, by using Eq. (5.12) at w = 0 and Eq. (5.13)
at t = 0.
h(t) = cos(7lt)r(2t) =
{~ ( n r ) for o 5 t 5
otherwise
Rather than evaluating this integral, determine the transfer hnction by using
the Fourier transform properties derived in Section 5.5 in order to hrther
illustrate how they can be used to simplify calculations.
514 Verify the result given by Eq. (5.68) by performing the convolution, Eq.
(5.61).
PROBLEMS 173
515 Let
Use the result obtained in Problem 59 together with the frequencyshift
theorem to obtain G ( j u ) .
517 Adapt the frequencyshift property and use the result of Problem 511 to
obtain the Fourier transform of g(t) = e" sin(o,t +
$J)r(t/T).(Note that the
Fourier transform of g(t) exists for any value of a because g(t) is nonzero
only over a finite interval and so g(t) is L,.)
518 Obtain the Fourier transform of the functionf(t) shown in Fig. 5.81 by direct
integration using Eq. (5.12) and so verify the result given by Eq. (5.86).
cos(oT)
H(jo)= ~
3 +jo
521 For each gain function given below, determine whether it can be the gain of a
causal LTI system.
(a) i ~ , ( j o )=
l e3wz
(b) IHb(jo)l =
(4I H c ( j o ) l = r(lol/W
(d) IHd(i0)l = 0.1 + r(lol/w)
522 Use the Parseval relation to determine the value of the integral,
I = Jrm
[1/(a2 o2)ldo.+
CHAPTER 6
In the last two chapters, we observed some of the advantages of analyzing LTI
systems in the frequency domain. It is the convolution property of the Fourier
transform, Eqs. (5.520), that is the basis for many of these advantages because
equations that involve convolution in the time domain become algebraic equations
in the frequency domain. However, a difficulty is that the Fourier transform of
functions that are not L , may not exist. For example, if x ( t ) = u(t), the unit step,
then the Fourier transform integral, Eq. (5.12), diverges for o = 0. Thus we could
not work with such functions in the frequency domain using our development of the
Fourier transform. Also, a problem with which we shall be concerned is the stabi
lization of unstable LTI systems. We could not analyze such problems using the
Fourier transform because the transfer function of an unstable system may not exist.
To extend the class of functions with which we can work in the frequency domain,
the Fourier transform is generalized. This generalization is called the bilateral
Laplace transform.’ As we shall see, the bilateral Laplace transform is just an
extension of the Fourier transform to a complex frequency plane. This extension
into a complex frequency plane will enable us to analyze the stabilization of unstable
systems. Also the bilateral Laplace transform enables one to develop a great deal of
insight and intuitiveness concerning LTI systems.
’ Pierre Shone de Laplace (17491827) was a protCgC of D’Alembert. Laplace made notable contribu
tions to cosmology, propagation of sound, and probability.
175
176 THE BILATERAL LAPLACE TRANSFORM
(6.1la)
F(jo)= 100
cJ
f(t)e'"' dt (6.1 1b)
converges so that IF(jo)I < 00 for all values of o and alsof(t) can be retrieved
from F ( j o ) using the inverse Fourier transform,
(6.1 1C)
As we discussed in Section 5.1, Eqs. (6.1lb) and (6.1lc) are called a Fourier
transform pair because if one equation is true, then so is the other. That is, if
F ( j o ) is obtained by Eq. (6.11b), thenf(t) can be retrieved by Eq. (6.1lc) and
iff(t) is obtained by Eq. (6.1 1c), then F ( j o ) can be retrieved by Eq. (6.1 1b). From
the viewpoint of the Fourier transform being a mapping of functions f ( t ) into func
tions F ( j o ) , the result that Eqs. (6.11) are a Fourier transform pair is the same as
stating that the mapping is onetoone.
To extend the class of functions for which a transform exists, we must modify the
condition given by Eq. (6.1la) which requires thatf(t) be an L , function. For this,
we define a function g(t) as
in which CT is a real constant which we choose so that g(t) is an L , function. That is,
we choose CT so that
I = 100
00
Ig(t)l dt = 100
oo
If(t)e"'l dt < 00 (6.12b)
The exponential, e"', is called a weighting function because the values off(t) are
"weighted" by it to make the integral, Eq. (6.12b), converge. We then have in
accordance with Eqs. (6.11) that the Fourier transform of g(t) converges so that
IG(jw)) < 03 for all values of w. Also, g(t) and G ( j o ) are a Fourier transform pair
for values of CT for which I in Eq. (6.12b) is finite.
6.1 THE BILATERAL LAPLACE TRANSFORM 177
G ( j o )= 1 00
00
g(t)ejwtdt (6.13a)
Substituting Eq. (6.12a), we have for values of o that satisfy Eq. (6.12b)
G(jw) = 1 00
oo
f (t)e''eiw' dt
The reason eP' was chosen as the weighting function is that e"ejw' = e('+Jw)t,
so that this equation can be written as
00
G(jw)=
J
[ 00
f (t)e('+iw)tdt (6.13b)
Now compare this last integral with that in Eq. (6.1lb). Note that the only difference
is that (jo)in Eq. (6.1lb) has been replaced by (o+jw). In Eq. (6.11b), the value
of the integral is F(jw). Thus, in accordance with the notation of Eq. (6.11b), the
value of the integral in Eq. (6.13b) is F ( o + j w ) . That is,
G ( j o ) = F ( o +jo) (6.13~)
S = o+jw (6.14a)
F(s) = 1 CC
00
f(t)e" dt (6.14b)
The funtion, F(s), is called the bilateral Laplace transform of f ( t ) . The adjective
bilateral is used because the integration with respect to t is from XI to +XIso that
it is over both (positive and negative) sides of the t axis. If the time function in Eq.
(6.14b) were the impulse response of an LTI system, h(t),then its transform, H(s),
is called the system function of the LTI system. That is,
H(s) = 100
00
h ( r ) P ' dt (6.14~)
It is important to note that the only values of o that can be used in Eqs. (6.14b) or
(6.14c) are those values for which the integral, I, in Eq. (6.12b) is finite. We shall
discuss this restriction in the next section. For the moment, however, observe in Eq.
178 THE BILATERAL LAPLACE TRANSFORM
(6.12b) that if I < 00 for cs = 0, we then can lets = 0 + j w in Eq. (6.14~)to obtain
the transfer function
H(jo)= 1 00
00
h(t)e@" dt (6.14d)
so that H ( j o ) = H(S)~,=,~.
We thus observe that the transfer hnction is a special
case of the system function. The system hnction and its use in system analysis will
be discussed in later chapters. Before continuing, we shall do some illustrative
examples to fix the ideas developed to this point.
In this expression, a is a real number that can be either positive or negative. Note that
if a = 0, thenf,(t) = Au(t), so that the step function is a special case of this example.
The first step in determining the bilateral Laplace transform is to determine the
values of cs for which Eq. (6.12b) is satisfied. For this we have that
In Eq. (6.12b), I is the area under If,(t)e"'l. This area is finite only if the exponent,
+
(a a), is greater than zero because, from Eq. (6.16), If,(t)e"'I is zero for t < 0
+
and, for (a a) > 0, it is a decaying exponential for t > 0. Observe that the area is
+
not finite if (a a) 5 0. We thus have that, in Eq. (6.12b), I < 00 only for those
+
values of cs for which (a 0)> 0 or, equivalently, for a > a.
The range of values of cs for which Eq. (6.12b) is satisjed is called the RAC of
f ( t ) . RAC is an abbreviation for the range of absolute convergence. That is, it is the
range of values of cs for which the integral of the absolute value of f,(t)e"',
If,(t)e"'l, converges. This requires the value of the integral, I , to be finite. Note
that it is not necessary to determine the actual value of I in Eq. (6.12b) because we
are only interested in determining whether I is finite. Thus the RAC off,(t) is
CS > a.
We now can determine the expression for FJs) from Eq. (6.14b). For a > a,
F,(s) = 1 00
00
f,(t)e" dt
(6.17)
6.1 THE BILATERAL LAPLACE TRANSFORM 179
The RAC must always be included in the expression for the Laplace transform of a
function.
Let us go through the details of evaluating the last integral in order to really
observe why we require a > a for this example. For this, first note that 00 in an
integral just denotes a limit. That is,
integral in Eq. (6.17) is, in reality,
Jr
really stands for limT+m s,'. Thus the
(6.1 8a)
For a limit to exist, the value of the function must approach a definite finite value.
For example, limt+,msin(wt) does not exist because as t increases, sin(wt) keeps
varying between +1 and 1 and thus does not approach a definite value. Also
limr+me"sin(wt) does not exist if a 5 0. However, the limit does exist if a > 0
and the value of the limit is zero.
For our case, Eq. (6.18a), first note that the limit does not exist if a = a
because then a + s =j w and so = ejot. For this case, the integral in Eq.
(6.18a) is
A
,jot dt = lirn [1  ejwT] (6.18b)
T+, 00 T+mJO
Because eJwTdoes not approach a definite value as T + 00, we have the limit in
Eq. (6.18a) does not exist for the case in which a = a. We now examine Eq.
(6.18a) for a # a. For this case, we obtain from Eq. (6.18a)
(6.18~)
To determine this limit we use the rectangular form of s as given by Eq. (6.14a) to
note that
Thus
In obtaining this last result, we have used from Appendix A that lejotI = 1. Also,
the magnitude bars about e('+a)T were removed because it is not negative. Thus we
note that the magnitude of +
grows without bound as T + 00 if (a a) < 0
+
so that the limit in Eq. (6.1%) does not exist for this case. However, if (a a) > 0,
then e(a+u)Tgoes to zero as T 4 00. Thus we have the result that the limit in Eq.
+
(6.18c) exists only if (a a) > 0 or, equivalently, for a > a. This is the RAC of
h(t)which we determined above. For a > a, we have limT+me(a+u)T= 0 so that
the limit in Eq. (6.18c) is Fa@) as given by Eq. (6.18a). The convergence of the
180 THE BILATERAL LAPLACE TRANSFORM
integral for the Laplace transform of a function is ensured if o is in the RAC of the
function.
We have obtained the result that the bilateral Laplace transform off,(t) given by
Eq. (6.15) is FJs) given by Eq. (6.17). This transform can be represented in the
complex s plane as shown in Fig. 6.11 for the case in which a < 0.
Fa@)in Eq. (6.17) is infinite for s = a. Values of s at which Fa@)is infinite are
calledpoles of the Laplace transform and are denoted by an x as shown in Fig. 6.11.
The RAC (region of absolute convergence in the s plane) is o > a, which is
indicated in Fig. 6.11 by the shaded region. Note that the constraint c > a is
independent of w so that, for example, o = a is the vertical line s = a + j w for
all values of w in the s plane. Thus, in the s plane, the RAC is all of the s plane to the
right of the vertical line s = a + j w .
Note that the pole at s = a does not lie in the RAC. In fact, the RAC of any
function, f (t),cannot include anypoles of its transform,F(s), because F(s) is infinite
at a pole and F(s) cannot be infinite within the RAC off (t). The reason is that, for
values of o within the RAC off (t), we have from Eq. (6.13c) that F(s) = G ( j w )
and IG(jo)l < 00 because g(t) is an L, function in accordance with Eq. (6.12b).
Consequently,
The first step in determining the bilateral Laplace transform is to determine the
values of CJ for which Eq. (6.12b) is satisfied. For this we have that
Ifh(t)e"'l = IBledUfb)'u(t).In Eq. (6.12b), I is the area under Ifb(t)e"'l. This
area is finite only if the exponent CJ + b < 0 because, from Eq. (6.1 12), Ifb(t)e"'I is
zero for t > 0 and, for t < 0, Ifb(t)e"'I decays exponentially to zero as t + co if
(T+ +
b < 0. Observe that the area is not finite if CJ b 2 0. Thus we have that, in Eq.
+
(6.12b), I < co only for those values of CJ for which CJ b < 0 or, equivalently, for
CJ < b. Thus the RAC (the region of absolute convergence) offb(t) is CJ < b. As
in Example 1, note that it is not necessary to determine the actual value of I in Eq.
(6.12b) because we are only interested in determining whether I is finite.
We now can determine the expression for F&) from Eq. (6.14b). For rs < b,
00
= J'
cc
BeCb'ePsfu(t)dt
(6.1  13)
J cc

B
 ~
CJ < b
s+b'
To fully understand the evaluation of the integral in Eq. (6.113) and to really
understand why the RAC offb(t) is CJ < 6, you should go through the details of
evaluating the last integral in the same manner as in Example 1. This transform can
be represented in the complex s plane as shown in Fig. 6.12 for the case in which
b < 0. As in the first example, note that the pole at s = b does not lie in the RAC,
which is indicated by the shading.
transform of
where a is a real number that can be positive or negative. Figure 6.13 is a graph of
this function for B = 1, tl = 0.4/s, w,, = 8 rad/s, and 4 = 0. In order to emphasize
thatf,(t) = 0 for t < 0, the graph is plotted starting at t = 2.
As in our previous two examples, we first must determine the RAC off,(t). For
this, in accordance with Eq. (6.12b), we must determine those values of a for which
(6.11 5)
Now,
A graph of this function for our example is shown in Fig. 6.14. The maxima of the
humps have been connected by a line. Clearly, the line is an exponential with the
equation Be("+")'u(t)because the maximum value of the cosine is one. The value of
the integral, I , in Eq. (6.115) is the sum of the areas of the humps. However, this
area clearly is less than the area under the exponential curve, which is finite if
+
(a 0)> 0. Consequently, I is finite if a > a. Now, if (a + 0)= 0, then the
amplitude of each hump is the same so that Z is infinite because there are an infinite
number of humps, each with the same area. Consequently, I is infinite for r~ = a.
+
Finally, if (a a) < 0, then hump amplitudes increase exponentially so that I is
clearly infinite for a < a. In summary, we have shown that the integral, I , in Eq.
(6.115) is finite only if a > a. Thus the RAC forf,(t) is a > a. Note that it was
not necessary to evaluate the integral, I , in Eq. (6.115), because our only interest is
whether it is finite. This determination often can be made by choosing an appropriate
upper or lower bound as we did in this example. Note the RAC for our example is
independent of the frequency wo and also does not include 0 = a.
We now can use Eq. (6.14b) to determine the expression for Fc(s) for 0 > a.
Fc(s)= 1 cc
cc
f,(t)ePs'dt
(6.1 17)
= 5,; Bepatcos(w,,t + 4)es' dt
This integral is not easy to evaluate in its present form. The exponential, however, is
one of the functions that is easy to integrate. The integrand of our present integral
6.1 THE BILATERAL LAPLACE TRANSFORM 183
1 ! d
2 0 2 4 6 8 10
t i m e in s e c o n d s
can be put into exponential form by expressing the cosine as the sum of exponentials
[see Eq. (A14)]:
With this and using the exponential property, @eb = e(a+b),the integrand in Eq.
(6.117) can be expressed in the exponential form
To fully understand the evaluation of the integrals in Eq. (6.119) and to really
understand why the RAC off,(t) is o > a, you should go through the details of
evaluating the integrals in the same manner as in Example 1.
The expression for Fc(s) in Eq. (6.119) can be put into a better form. For this,
first note that iff(t) is a real function, then, for s = o + j O , the integrand in Eq.
(6.14b) isf(t)e"', which is a real function o f t so that the value of the integral,
which is F(s) with s = o +jO, must then be real. That is, i f f ( t ) is a real function of
t, then F(o) must be a real function o f o . Becausef,(t) is a real function o f t in our
present problem, F&) with s = o +jO must be a real function so that we must be
able to eliminate thej's in Eq. (6.119). To obtain the desired expression, we add the
two terms and use the exponential expressions for the sine and cosine functions
[Eqs. (A14) and (Al5)].
1 (s
F,(s) =  B
+ CI +jwO)eJ++ (s + CI jco,)ej+
2 (s + a>2+ Of
1 (s + .)(e'$ + e++) + j o , ( e j + e++)

=B (6.120)
2 (s + a)2 + Of
(s + E) cos(4) coo sin(#)

=B , o>a
(s + a)2 + o;
for which
s+a
Fc,(s)= B o > a (6.12 1b)
(s + + Of'
and for 4 = 71/2ra4 we have
f,,(t) = Bepatsin(w,t)u(t) (6.122a)
for which
WO
Fc,(s)= B o > a (6.122b)
(s + a)2 + Of'
6.1 THE BILATERAL LAPLACE TRANSFORM 185
For the general case given by Eq. (6.120), there are two poles: One is at
s = a + j o o and the other is at s = a jo,. Also there is one zero that, for
cos(4) # 0, is located at s = a + m0 tan(4). Note that the pole locations are
conjugates of each other. This is a consequence of the fact that F(o) is a real function
iff ( t ) is a real function. In fact, generally, all poles and zeros that are not located on
the o axis must occur in conjugate pairs iff ( t ) is a realfinction oft. The reason for
this will be more clear when we discuss the splane in more detail in Chapter 9.
The transform, F,(s), can be represented in the complex s plane. The case in
which a > 0, 4 = n/4rad, and oo> a is shown in Fig. 6.15. As in our previous
examples, note that the poles do not lie in the RAC, which is indicated by the
shading. For the case illustrated, also note that the zero lies in the RAC. A zero
can lie anywhere; there is no restriction on its location. The only restriction is that a
pole cannot lie in the RAC.
Example 4 As our last example, we shall determine the bilateral Laplace trans
form of
for t p 0 (6.123)
for t 2 0
where a and b are real numbers. Note that we can expressfd(t) in terms of the
functionsfa(t) andfh(t) of the first two examples in this section as
Fig. 6.15 The splane representation of FJs) for the case c( > 0, 4 = n/4, and wo > a.
186 THE BILATERAL LAPLACE TRANSFORM
for the case in which A = 1 and B = 1. In accordance with our previous examples,
we must first determine the RAC offd(t) which are the values of CT for which I < 00
in which
Z= 100
cc
Ifd(t)e"'I dt (6.125a)
By substituting we have
(6.125b)
From our discussion of the RAC off,(t) and offb(t), the value of the first integral in
Eq. (6.125b) is finite only for G < b and the value of the second integral is finite
only for G > a. For I to be finite, we require both integrals to be finite so that we
require G < b and also G > a. Combining these two inequalities, we require
This is the RAC offd(t). Note that this requires b < a. This means that the bilateral
Laplace transform offd(t) does not exist if b ? a. You should make some drawings
offd(t) and Ifd(t)e"'I for cases in which b < a and in which b 2 a and note that the
area under Ifd(t)e"'I can be made finite only for the case in which b < a by
choosing a value of G between a and b.
For the case in which Eq. (6.126) is satisfied, we then have
Fd(s)= 1
00
fd(t)ePstdt
cc
1 1 ba

 +
s+b s+a

(s+a)(s+b)'
U < G < b
Figure 6.16 is the splane representation of Fd(s) for the case in which a > 0 and
b < 0 with the RAC indicated by the shaded region. Note that the RAC is the region
between the two pole but does not contain the poles.
The examples given in this section illustrate the basic direct techniques for
determining the bilateral Laplace transform of a function. The basic techniques
shown, however, can be somewhat tedious and lend no real insightful understanding
of the transform. For this, we shall first examine some properties of the RAC and
then some properties of the transform. These results then will be used to gain a better
understanding of the bilateral Laplace transform and to simplify its determination.
6.2 SOME PROPERTIES OF THE RAC 187
Fig. 6.16 The splane representation of F&) for the case in which a = 0 and b < 0.
In the last section, the RAC of any function,f(t), was defined to be those values of CJ
for which
(6.21)
and the determination of the RAC of some functions was illustrated. Their determi
nation was seen to require some effort. However, there are some properties of the
RAC which simplify its determination. Furthermore, some important properties of
the function,f(t), can be determined directly from its RAC. We shall discuss some
of these properties in this section because they will be important in our later
discussions.
I = I,
00
We now let z = t  to. With this change of variable in the integral, we obtain
(6.22b)
188 THE BILATERAL LAPLACE TRANSFORM
NOW e41+'0) = eGeOT . Observe that 0 < e"'O < 00. Also, because the integra
tion is with respect to t, the factor ePufOis a constant during the integration. Thus we
can express the integral in Eq. (6.22b) as
(6.22~)
for any value of B in the RAC off ( t ) . Because e"'O < 00, we conclude that
(6.23)
for any value of B in the RAC off (t  to).But values of B for which Eq. (6.23) is
satisfied are values of B in the RAC off ( t ) .Thus we have shown that the RACs of
f ( t ) and f ( t  to) are identical. Physically, this property states that the RAC of a
function does not depend on the point we call t = 0; it is depends only on the shape
of the function.
(6.24a)
First express this integral as one over negative values o f t plus one over positive
values o f t as
bo
If (t)e"OfI dt = J
00
If (t)e'O'l dt + J0 If (t)e"Ofldt
(6.24b)
6.2 SOME PROPERTIES OF THE RAC 189
Now, because no < 0 2 , note that ec'o' < e'2' for negative values o f t so that
If (t)eCUn'1< If (t)e'2'1 for negative values of t. Thus,
I, = 10
00
1f(t)e"n'Idt < 10
00
If (t)e'2'I dt < 00 (6.25a)
I, < 00 because
If (t)eQ'I dt 5 1
03
00
If (t)e"2'1 dt < 00 (6.25b)
We obtain the first in equality in Eq. (6.25b) by noting that the second integral
equals the first integral plus the area under the curve for positive values o f t . The
second integral is finite because o2 lies in the RAC off (t).Thus we have shown that
I , < 00. We now show that I2 < 00 by first noting that e'o' < e'I' for positive
values of t because oo > 0,. Consequently, If (t)e"o'I < If (t)e"l'I for positive
values o f t . Thus,
l2= loIf00
I2 < 00 because
1; If(t)e"I'I dt 5
00
m
If (t)e"l'I dt < 00 (6.26b)
We obtain the first inequality in Eq. (6.26b) by noting that the second integral
equals the first integral plus the area under the curve for negative values o f t . The
second integral is finite because o, lies in the RAC of f ( t ) . Thus we show that
I, < 00. With the use of Eq. (6.24b), we have shown the correctness of Eq. (6.24a)
because we have shown that I, < 00 and I, < 00. Thus o = oo also lies in the RAC
o f f (t).
Because a pole cannot lie in the RAC, an immediate consequence of the interval
property is that the RAC of a function cannot be on both sides of a pole. For
example, consider the RAC of &(t) in Section 6.1. Its RAC is between the two
poles at a and 4.Its RAC could not be, say, b < o < a and also o > a because
there is a pole at o = a which, as we discussed, cannot lie in the RAC. Thus we
note that the RAC of a function,f( t ) , is an interval between the poles of its trans
.form,m).
190 THE BILATERAL LAPLACE TRANSFORM
The two integrals in Eq. (6.27a) are equal becausef(t) = 0 for t < 0 so that the
integral over negative values oft is zero. The value of the integral is finite because cro
lies in the RAC off(t). Now let crl > cro. We then note that for positive values of t,
< e" 0 f so that If(t)e"l'l < 1f(t)e"ofI. Consequently,
so that cr, lies in the RAC. Because cr, was any value of cr larger than cro, we
conclude that the RAC must include cr 2 go.
Note that we have only shown that the RAC is to the right of all the poles if
f ( t ) = 0 for t < 0. We cannot conclude that the function,f(t), is necessarily zero for
t < 0 if the RAC is to the right of all the poles. As an example, considerf,(t 2). +
This is the functionf,(t) discussed in Section 6.1 advanced by 2 s. We then have that
+
f , ( t 2) # 0 for 2 5 t 5 0. However, by the timeshift property of the RAC, the
RAC off,(t  to) is the same as the RAC off,(t), which is cr > a.
'A function that is zero for f i0 is sometimes called a causal function in the literature because such a
function can be the impulse response of a causal LTI system. To avoid confusion, I do not use that
terminology because causality means that there is a causal relation as we discussed in Section 3.5. A
function, however, is just a mapping as discussed in Section 1.4 and so there is no causality concept
involved.
6.3 SOME PROPERTIES OF THE BILATERAL LAPLACE TRANSFORM 191
response of a BIBOstable LTI system. From Eq. (6.2l), the RAC of f ( t ) is those
values of (T for which
(6.28)
(6.29)
then the RAC includes (T = 0 because Eq. (6.29) is Eq. (6.28) with (T = 0. From
Eq. (6.28), we further note that if (T = 0 lies in the RAC, then f ( t ) is an L , function.
That is, f ( t ) is an L , function if and only if (T = 0 lies in the RAC o f f ( t ) . In the s
plane, (T = 0 is the w axis because it is the vertical line s = 0 +jo.We thus have the
result that f ( t ) is an L , function ifand onZy $ in the s plane, the o axis lies in the
RAC off ( t ) . Consequently, because an LTI system is stable if and only if h(t) is an L ,
function, we have the important result that an LTI system is stable ifand only ifthe w
axis lies in the RAC of the system function, H(s), given by Eq. (6.14c).
From Eq. (6.15), it can be seen that the functionf,(t) is not an L , function for
a 5 0. This same result can be obtained directly from Eq. (6.17), from which we
have that the RAC off,(t) is (T > a so that the RAC off,(t) does not include (T = 0
if a 5 0. Note from the splane representation of Fa@)shown in Fig. 6.11 that the
RAC does not include the w axis for the case a < 0. If a > 0, thenf,(t) is an L ,
function and we note from the splane representation shown in Fig. 6.11 that the o
axis is included in the RAC in accordance with the L , property of the RAC.
Similar to the properties of the Fourier transform discussed in Chapter 5, the various
properties o f the Laplace transform not only enable us to obtain the Laplace trans
form of a function more easily, but also enabled us to obtain a deeper understanding
of the Laplace transform and its applications. We expect the Laplace transform
properties to be similar in form to their Fourier transform counterparts because, as
discussed in Section 6.1, the Fourier transform is a special case of the Laplace
transform.
where z1 and z2 are complex quantities. With the use of this inequality, we have
so that
I = 1 00
00
If(t)e"'I dt i 1 00
00
Ifi(t)e"'I dt + 1
00
00
Ih(t)e"'I dt (6.33)
The RAC are those values of g for which I < 00. Now if the RACs offi(t) andf,(t)
overlap then, for values of n in the overlap, both integrals in Eq. (6.33) are finite so
that I < 00. The overlap is thus contained in the RAC off(t). For values of o in the
overlap, we then have
00
f(t)e"' dt = 1
00
00
[Clfi( t ) + C2f2(t)]e"'dt
= Cl 1
00
00
fi(t)e"' dt + C2 100
00
fi(t)e"' dt (6.34)
Note that the condition for the linearity property to hold is that the RACs of the
two functions being summed overlap. This is physically reasonable because the only
values of s that can be used in F , (s) are those in the RAC of f i ( t ) and, similarly,
the only values of s that can be used in F2(s)are those in the RAC off2(t). Because
the same value of s must be used in each of the three functions in Eq. (6.34), the
value of s used must lie in the RAC of each function, which means that it must lie in
the overlap of the RACs. If the RACs of fi ( t ) and f2(t) do not overlap, then Eq.
(6.34) is not valid. However, in such cases, it still is theoretically possible for the
+
Laplace transform of f ( t ) to exist. For example, let fi ( t ) = 1 = u(t) u (  t ) and
h(t)= u(t). The Laplace transform of f i ( t ) for this example does not exist
because it has no RAC (observe from Section 6.1 thatfi ( t ) =fd(t) with a = 0 and
b = 0). The Laplace transform offZ(t) does exist, and its RAC is a < 0 (note from
Section 6.1 that h(t)=fb(t) with b = 0). However, f ( t ) =fi ( t )+f2(t) = u(t). Its
Laplace transform does exist (note from Section 6.1 that u(t) = f , ( t ) with a = 0)
and its RAC is a > 0. The linearity property only concerns functions that have
overlapping RACs.
f(0= f i ( c t ) (6.35a)
is
(6.35b)
To obtain this property, we first must determine the RAC off(t). In accordance with
Eq. (6.12b), we must determine the values of a for which I < 00 in which
I = 1
00
00
If(t)e"'l dt = 1 00
00
Ifi (ct)e"'l dt (6.36a)
To express the second integral in the form of the RAC offi (t),we make the change
of variable, z = ct in the second integral. Then, in a manner similar to that in Section
5.5, we obtain
(6.3 6b)
Except for the constant l/lcl, we recognize the integral as that for the RAC offi(t)
but with a replaced with a/c. Because the RAC off ( t ) is aa < a < ab, we conclude
that I in Eq. (6.36b) will be finite for aa < a/c < ab.This then is the RAC off(t) as
given in Eq. (6.35b). We now can determine F(s) for a in this range.
00 00
~ ( s=
)
JL f(t)e"' dt = J 00
f i ( c t ) e P dt (6.3 7)
194 THE BILATERAL LAPLACE TRANSFORM
To express the second integral in the form of the Laplace transform offi (t),we make
the change of variable, z = ct in the second integral. Then in a manner similar to the
proof of the Fourier transform scaling property in Section 5.5, we obtain
(6.38)
The integral is recognized as the Laplace transform offi ( t ) with s replaced with s/c
so that we have
(6.39)
is
with the RAC aa < o < o b or, equivalently, ob < o < oa.
As an example of this result, letfi ( t ) =f,(t) wheref,(t) is defined in Section 6.1.
Then
1 1
F(s) =  , ~ o t a (6.31 lb)
s+a sa
Observe from Section 6.1 thatf(t) =fb(t) with b = a. With this substitution in Eq.
(6.31 lb), note that we do obtain the transform forfb(t).
is
6.3 SOME PROPERTIES OF THE BILATERAL LAPLACE TRANSFORM 195
The RAC off(t) is the same as that offi(t) as a consequence of the timeshift
property of the RAC discussed in Section 6.2. To obtain the expression for F(s), we
have
f(t)e"' dt = .[
00
'x
f i ( t  to)e" dt (6.3 13a)
This can be put in the form of the Laplace transform offi(t) by the change of
variable 7 = t  to with which we obtain
(6.3 13b)
D > a (6.314b)
Note that the RAC is to the right of the pole of F(s) but f ( t ) # 0 for t < 0. Later,
we'll determine conditions on F(s) from which we can determine whetherf(t) = 0
for t c 0. As stated previously, such a function is important in our study because it
can be the impulse response of a causal LTI system.
in which so = +jwo is
z= 100
'x
If(t)e"'I dt = 1
03
00
Ifi(t)eso'eu'I dt (6.316a)
(6.316b)
which is obtained using results from Appendix A. Observe that the integral in Eq.
(6.316b) is that for the RAC offi(t) but with f~ replaced with (r  o0. We thus
conclude that the RAC o f f ( t )is o0 < f~  go < bb, which is equivalent to that given
in Eq. (6.315b). With CT in this interval, we then have that the Laplace transform of
f ( t ) is
Let the Laplace transform offi(t) be F,(s) with the RAC a, < a < ab, and let the
Laplace transform off2(t) be F2(s)with the RAC ac < a < ad.We shall show that if
the RACs offi(t) andf,(t) overlap, then the Laplace transform off(t) is
with CT in the overlap of the RACs offi(t) andfi(t). Before proving this property,
note that it is what we intuitively would expect because if a = 0 lies in the RAC of
f i ( t ) and offZ(t), then we can let a = 0 in Eq. (6.319b) to obtain Eq. (5.520b), the
convolution property of Fourier transforms. Further since Eq. (6.319b) involves
both F,(s) and F2(s), the equation can be valid only in the overlap of the RACs
of both functions for the same reason given following Eq. (6.34).
We shall prove this property by a somewhat different procedure than that used
previously; we shall simply obtain the expressions for F(s) and note the values of a
for which the expressions are valid. To begin, the integral expression for F(s) is
F(s) = 1 W
W
f(t)e"' dt (6.320a)
F(s) = /:_",[J'" W
fi(z&(t  z) dz]e" dt (6.320b)
The integration in this double integral is first with respect to z and then with respect
to t. Let us interchange the order of integration by first integrating with respect to t
and then with respect to The double integral then is
F(s) = 1w
_",
fi(.,[.r"
W
f2(t  z)eFsfdt] d z (6.320~)
From the timeshift theorem, we note that the value of the integral in the brackets is
F2(s)eCS7for a in the RAC off,(t). Thus Eq. (6.32Oc) is
The function F2(s)has been factored out of the integral because it is a constant in the
integration with respect to z. The value of the integral is recognized to be F , (s) for a
For u in the overlap, the double integral converges absolutely, which is sufficient to ensure that the
interchange of the integration order used to obtain Eq. (6.32Oc) is valid.
198 THE BILATERAL LAPLACE TRANSFORM
in the RAC offi(t). Thus for u in the RAC offi(t) andf,(t), which is the overlap of
the RACs, we obtain
which is Eq. (6.319b). We shall use and illustrate this property extensively in
subsequent chapters.
(6.32 1a)
is
F(s) = 100
oo
f(t)e" dt, cra < 0 < o
(6.322)
= 100
00
f;'(t)e"' dt
To express this integral in the form of the Laplace transform offi (t), we integrate by
parts as we did in obtaining the Fourier transform timedifferentiation property in
Section 5.5. From Eq. (5.523), the integration by parts formula is
(6.323)
for which
F(s) =f;(t)e"lw
cQ
 1
w
m
(s)fi(t)e" dt (6.324)
The value of the first term is zero for 0 in the RAC offi(t). To see this, note that
(6.326)
f,(t) = A e P u ( t ) (6.327a)
This is an algebraic equation from which we obtain the algebraic expression for
F,W.
A
Fa@) = s+a (6.327d)
pole of F,(s) which is at s = a, we conclude that the RAC off,(t) must be 6 > a.
This is the result obtained previously, Eq. (6.17).
is
Fo(S) = / 03
00
h(t)e" dt, 6, < 6 <6b (6.329a)
F(s) = 
rm
: h(t)eO' dt
= / 00
cc
(tlfo(t)e"' dt = / 00
cc
&(t)eKJt dt, cra < 6 < 6 b
(6.329~)
The last integral is recognized as the Laplace transform of &(t), which was to be
shown.
As an application of this property, let h(t)= AeP'u(t>. From Eq. (6.17), the
+
Laplace transform of this function is Fo(s)= A / ( s a ) with the RAC 6 > a. Thus
we have with the use of the frequencydifferentiation property that the Laplace
transform of
is
d d A A
F,(s)= Fo(s) =    (6.330b)
ds ds s a (s + ~
is
d d A 2A
F2(s)=F*(s)= r~ > a (6.331b)
ds ds (s + a)2  (s + a)3 '
By continuing in this manner, it is easily shown that the Laplace transform of
f , ( t ) = At"e"u(t), n = 0, I , 2 , 3 , . . . (6.332a)
is
An!
F h )= r~>a, n = 0 , 1 , 2 , 3 ,... (6.332b)
+
(s a)@+')'
This function is a ramp with a slope equal to one. In terms of the ramp, we then can
express f ( t ) as
(6.334)
You should verify the correctness of this expression by drawing a graph of each term
of this expression and also the sum of the graphs. For this, show that the sum of two
straight lines is a straight line with a slope equal to the sum of the slopes of the two
lines added.
Now, with the use of the timeshift property and the linearity property, we have
that
A
F(s) =  [G(s)  2G(s)e” + G(s)eP2*’]
T
A
=  G(s)[1  2eTSe e2Ts]
T
+ (6.335)
A
=G(s)[l e 3
Ts 2
T
To obtain G(s), we have shown that the Laplace transform of u(t) is l / s with the
RAC o > 0. Thus from the frequencydifferentiation property, we have
d l 1
G(s) =   =  o>o (6.33 6)
ds s s2’
Note that G(s) also can be obtained from Eq. (6.332) for the case n = 1, a = 0, and
A = 1 . Substituting the expression for G(s) into Eq. (6.335), we obtain
(6.337)
This is the expression for F(s). But what is the RAC? The RAC of Eq. (6.335) is
c > 0 because that is the RAC of G(s). This can be seen by noting that the expres
sion involves G(s) so that the only allowed values of o are those that lie in the RAC
of g(t). However, once the expression for G(s) is substituted, the RAC could be
larger. We could determine the RAC by actually determining the values of o for
which I [in Eq. (6.21)] is finite. However, the RAC off(t) also can be determined
for our example by making use of some properties of the RAC determined in Section
6.2.
First, becausef(t) = 0 for t < 0, we use the property that the RAC must be to the
right of all the poles of F(s). Now it would appear from Eq. (6.336) that there is a
pole of F(s) at s = 0. However, things are not always as they appear at first glance.
Using the power series expansion of the exponential, Eq. (Alo), we have
(6.33 8)
PROBLEMS 203
"[ 2'!
F ( s ) =  T   T 2 s +  T 31 s 2   T s
T 3! 4!
l2
+... (6.339)
Note that h+,, F(s) = A T so that there is no pole at s = 0. The reason is that
[ 1  eTs]has a zero at s = 0 which cancels the pole there. In fact, there are no poles
in the finite s plane so that the RAC must be the whole s plane. That is, the RAC of
f ( t ) is 00 < (T < 00. Thus, the Laplace transform off (t) is
(6.3 40)
Another method by which the RAC can be determined is to note from Fig. 6.32
that
Thus, from the L , property of the RAC we have that (T = 0 must lie in the RAC. We
immediately conclude from this that there is no pole of F(s) at (T = 0 because we've
discussed, a pole cannot lie in the RAC. Thus we observe from Eq. (6.340) that F(s)
has no poles. Thus the RAC must be 00 = (T < 00 because, from our discussion of
the interval property of the RAC, the RAC is an interval between adjacent poles.
Note how use of the properties of the RAC and the Laplace transform greatly
simplified the determination of the transform of a function. Because (T = 0 lies in the
RAC off(t), we can let s = 0 in the expression for F(s), Eq. (6.14b), to obtain
00
(6.341)
Thus, if (T = 0 lies in the RAC off(t), then F(0) equals the area under f (t). You
should verify this result for our example.
PROBLEMS
61 The RAC for the hctionfd(t) of Example 4 in Section 6.1 was shown to be
< (T < b.
Verify this result by sketchingfd(t) for a = 2 and b = 0 and then showing
that the area under Ifd(t)e"' I is finite only if 2 < < 0.
Show that for the case a = 0 and b = 2 there is no value of s for which
the area under Ifd(t)e"'I is finite so that the Laplace transform offd(t) for
this case does not exist.
204 THE BILATERAL LAPLACE TRANSFORM
62 Show that the Laplace transform off (t) = Ad(t) is F(s) = A with the RAC
 0 0 < 0 < 0O.
63 The unit impulse response of a given LTI system is h(t) = b(t  to) +
e"u(t).
(a) Determine H(s), the Laplace transform of h(t). Do not forget to specify
the RAC.
(b) If the value of a is such that the given system is stable, what would be the
system transfer function?
64 (a) Determine F(s), the Laplace transform of f(t) = cos(o,t)u(t). Do not
forget to determine the RAC.
(b) Sketch the splane polezero diagram of the function F(s) and specify the
location of each pole and zero.
65 +
The unit impulse response of a given LTI system is h(t) = [l e*']u(t).
(a) Determine H(s), the Laplace transform of h(t). Do not forget to specify
the RAC.
(b) Is the given system stable? Give a short timedomain and an splane
statement of your reason.
(c) Is the given system causal? Give a short statement of your reason.
66 (a) Determine the bilateral Laplace transform off (t) = e31r1.
Do not forget
to specify the RAC.
(b) Could f (t) be the unitimpulse response of a stable LTI system? Give a
short time domain and an splane statement of your reason.
(c) Could f(t) be the unitimpulse response of a causal LTI system? Give a
short statement of your reason.
67 Letf,(t) = u(t) andf,(t) = u(t). Because the two functions are not equal,
their transforms should not be equal. How do they differ?
68 Let f(t) = 0 for t > 0. Show that all the poles of F(s) lie to the right of the
RAC off (t).
69 The RAC of the function shown in Fig. 6.32 was determined to be
00 < 0 < 00. This is a special case of a general result, which is: The
RAC of any bounded function, f (t), which is nonzero only over ajnite range
oft, t, < t < t2, is 00 < (T < 00. This means that the RAC of such a
function includes the whole s plane. Prove this general result.
610 Go through the details of the derivation of Eq. (6.39) to show that the
constant after the equal sign is l/lcl.
PROBLEMS 205
61 1 Use the result given by Eq. (6.120), and use the scaling property to
determine the bilateral Laplace transform off(t) = Beat cos(o,t  O)u(t).
613 Determine the Laplace transform off(t) = (1  t)r(t)by using the difientia
tion property.
615 We have shown that the Laplace transform offi ( t ) = u(t) is F , (s) = 1/s with
the RAC c > 0. Use this result and the frequencyshif?property to obtain the
Laplace transform off(t) = Be" COS(W,~ + $)u(t).
616 Use the frequencydifferentiation property and the transforms obtained in the
text to determine the bilateral Laplace transform of the following functions:
(4fa@) = tu(t)
(b) fb(0= tcos(oot>u(t>
(c) L(t)= W / T )
618 A technique that can be used to determine the Laplace transform of some
functions is to obtain a differential equation of which the transform is a
solution and then solve the differential equation for the transform. Some of
the examples in Section 6.3 used this technique. In this problem, we'll further
illustrate this technique by determining the Laplace transform of
f ( t ) = e(@)+,c1 > 0.
(a) First show that the RAC of,f(t) is 00 < c < 00.
+
(b) Show that f ( t ) satisfies the differential equation f ' ( t ) crtf(t) = 0.
(c) Use the properties in Table 7.42 to show that the Laplace transform of
f ( t ) satisfies the differential equation F'(s)  (1/tx)sF(s)= 0.
(d) Note that the differential equation in part c is similar to that in part b. The
essential difference is that CI has been replaced with l/a. From this
observation, conclude that the solution of the differential equation in part
d must be F(s) = Ke('/2a)sz, 00 < c < 00, in which K is a constant.
(e) To determine the constant K , show that K = s", dt.
e(a/2)t2
206 THE BILATERAL LAPLACE TRANSFORM
(f) Show that the Fourier transform off(t) exists and determine it.
This system would be stable and causal. Note that the difference is
(b) For a given input, x(t), show that the difference between the output of the
desired system delayed by to seconds and the output of the constructed
system is y,(t) = h,(t)*x(t).
(c) Show that for a bounded input for which Ix(t)l < M,, we have
so that the difference decreases for increasing delay and goes to zero as
to +. 00. Observe that, by this technique, the response of any stable but
noncausal LTI system can be obtained with arbitrarily small error by
accepting arbitrarily large delay.
We now examine how the transfer function of the constructed system
compares with that of the desired system. For this, we first determine the
transfer function of the constructed system. For this, well determine the
system function, H(s),and then lets =io to obtain the transfer function,
H U o ) . This can be done because the constructed system is stable so that
the o axis lies in the RAC. A nice way to determine the system function,
H(s), is to use the differentiation property of the Laplace transform.
PROBLEMS 207
+
(e) Show that g'(t) = ae"'o6(t)  2a6(t  to) a2h(t).
(f) Use the differentiation property together with the results of parts d and e
to show that H(s) is
(g) Because the constructed system is stable and causal, we have from
Section 6.2 that all the poles of H(s) must lie in the left half of the s
plane. Thus there should not be a pole of H(s) at s = a. Show that this is
so.
(h) Now obtain the transfer function of the constructed system and show that
for large enough delay, the gain of the constructed system is approxi
mately that of the desired system and the phase shift of the constructed
system is approximately that of the desired system minus otoradians.
The difference, wto, is the phase shift due to the delay as discussed in
Section 5.7.
CHAPTER 7
The use of the Laplace transform simplifies and also lends insight into many time
domain operations. However, for it to be useful for our purposes there must be a one
toone mapping from the time domain to the s domain. That is, it is useful only if
there is only one transform, F(s) , for every time function,f(t), and vice versa. If the
mapping is onetoone, then there must be a method by which the time function,f(t),
can be obtained from its Laplace transform, F(s). The time function so obtained is
called the inverse Laplace transform of F(s). We shall develop the formula for the
inverse Laplace transform and methods for evaluating it in this chapter.
[see Eq. (6.12a)l and choose (T so that g(t) is an L, functionthat is, so that
I = 1
00
00
Ig(t)l dt = 1
00
52
If(t)e"'I dt < 00 (7.11b)
[see Eq. (6.12b)l. We found that the values of (T for which I < 00 lie in an interval,
(T,< (T < ob, which we call the RAC off(t). This is the interval property of the
RAC obtained in Section 6.2. Thus, in accordance with our Fourier transform result,
Eqs. (5.1l), (5.12), and (5.13), the Fourier transform of g ( t ) exists for (T in the
209
210 THE INVERSE BILATERAL LAPLACE TRANSFORM
RAC off(t). With the use of the definition of g(t), the Fourier transform of g ( t ) was
expressed as
(7.11C)
where F(s) is the Laplace transform off(t). That is, the Laplace transform off(t) is
simply the Fourier transform of g(t).
To develop the inverse Laplace transform, we first note that because g(t) is an L ,
function for o in the RAC of f ( t ) , oa < o < rsb, we have in accordance with
Eqs. (5.13) that the inverse Fourier transform of g(t) exists (so that the mapping
of g(t) to GGo) is onetoone) and is
g(t) = 1'271 
00
G(jo)e'"' do (7.12a)
Because the exponential, e"', is not equal to zero for any value of ot, we can divide
both sides of this equation by ePutto obtain
The exponential was put under the integral because the integration is with respect to
o and the exponential is not a function of o.This equation is the desired inverse
transform. However, because all our expressions for the Laplace transform are in
terms of s, a nicer form of this expression is in terms of s and not in terms of its
component parts, o and o. To obtain the desired expression, we substitute
s = o +jo to obtain
To complete our substitution, we must express the integral in terms of s. For this, we
note that the integral is with respect to w, so that o is a constant with a value o = o,,
7.1 THE INVERSE LAPLACE TRANSFORM 211
This is the desired expression for the inverse Laplace transform in terms of s.
Let us first note that we only used the inverse Fourier transform of G ( j o ) to
obtain Eq. (7.13a). Thus, because the mapping of g(t) to G ( j o ) is onetoone,
we conclude that the mapping off(t) to F(s) for CJ in the RAC of f ( t ) also is
onetoone. This means that the Laplace transform off(t) given by
F(s) = 1
00
00
f(t)e" dt, oU < c~ < o b (7.13b)
and the inverse Laplace transform given by Eq. (7.13a) are a transform pair. Thus, if
Eq. (7.13b) is evaluated withf,(t) to obtain F&), then the evaluation of Eq. (7.13a)
with Fa@)and o0 in the RAC off,(t) results in the same function,f,(t), with which
we started. For example, we determined in Section 6.1 that the Laplace transform of
f , ( t ) = Ae"'u(t) (7.14a)
is
1
Fa($ = , CJ > a (7.14b)
s+a
Thus, if the integral in Eq. (7.15) were evaluated with a negative value o f t , the
value of the integral would be zero. Also, if a positive value o f t were used, the value
of the right side of Eq. (7.15) would be Ae"'. To better understand the integration
in Eq. (7.15), consider an splane view of it as shown in Fig. 7.11. The figure is
drawn for a > 0. As shown, the integration is along a vertical line in the s plane
because cr, is a constant in the integral. The line is to the right of the pole at s = a
because go > a. The value of the integral will be the same no matter what value of
o0 is used as long as it is greater than a. To actually perform this integration
requires results from an area of mathematics called complex variable theory. For our
study of LTI system theory, we won't need those results because we can make use of
our result that Eqs. (7.13) are a Laplace transform pair as we did to evaluate the
integral in Eq. (7.15).
212 THE INVERSE BILATERAL LAPLACE TRANSFORM
I"" Q
Fig. 7.11 Depiction of the integral in Eq. (7.15) (drawn for the case a > 0).
What would be the value of the integral in Eq. (7.15) if the integration were
performed with a value of oo which is less than a? For this, first consider the result
obtained in Section 6.1 that the Laplace transform of
is
B
F&) = s + b '
~
o ib (7.16b)
Thus, if the integral were evaluated with a negative value oft, the value of the right
hand side of Eq. (7.17) would be Bebt. Also, if a positive value o f t were used, the
value of the integral would be zero. The splane view of this integral is shown in
Fig. 7.12.
The integration is along a vertical line as shown in the splane because oo is a
constant in the integral. The line is to the left of the pole at s = b because
oo < b. The value of the integral will be the same no matter what value of oo is
used as long as it is less than b.
Fig. 7.12 Depiction of the integral in Eq. (7.17) (drawn for the case b < 0).
7.2 THE LINEARITY PROPERTY OF THE INVERSE LAPLACE TRANSFORM 213
We can now determine the value of the integral in Eq. (7.15) if the integration
were performed with a value of eo which is less than a. For this, choose B = A
and b = a in Eq. (7.17). We then have
The integrand in Eq. (7.18) is the same as that in Eq. (7.15). The only difference is
that the integration in Eq. (7.15) is along a line to the right of the pole while the
integration in Eq. (7.18) is along a line to the left of the pole. Note that the
integration along a line to the right of the pole results in a time function that is
zero for t < 0 while integration along a line to the left of the pole results in a
fimction that is zero for t > 0.
We now have evaluated the inverse Laplace transform integral for two cases. One
case is given by Eq. (7.15) and depicted in Fig. 7.1 1. The second case is given by
Eq. (7.17) and depicted in Fig. 7.12. To continue our development of the inverse
Laplace transform, we now need a linearity property.
Let the Laplace transform off(t) be F(s) with the RAC ea < 0 = eh.Now express
F ( s ) as the sum of two functions as
Then, with the use of the inverse Laplace transform equation, Eq. (7.13a), we obtain
where qo lies in the RACs of both fi ( t ) and f2(t). This property states that if we
express F ( s ) as the sum of two functions as given by Eq. (7.2la) and obtain the
inverse Laplace transform of each hnction individually, thenf(t) will be given as the
214 THE INVERSE BILATERAL LAPLACE TRANSFORM
sum of two functions as given by Eq. (7.2lb) in which the RAC offi(t) and the
RAC off2(t) overlap with o0 lying in the overlap.
As an illustration of this important property, consider the f i c t i o n
ba
F(s) = a < < b (7.22a)
(s + a)(s + b) ' IS
This is the Laplace transform offd(t) determined in Section 6.1. First express F(s) as
the sum of two functions as
1 1
F(s) =  
s+b s+a'
+ a < IS < b (7.22b)
f ( t ) = f i ( t ) +f2(t) (7.23a)
where
(7.23b)
and
We note in the integral forfi(t) that go < b so that the integral is along a vertical
line in the s plane which is to the left of the pole at s = b. Thus this integral is
exactly the same as that in Eq. (7.17) with B = 1, so that
f i ( t ) = epb'u(t) (7.24a)
Also, we note in the integral forf2(t) that go > a, so that the integral is along a
vertical line in the s plane which is to the right of the pole at s = a. Thus this
integral is exactly the same as that in Eq. (7.15) with A = 1, so that
This function isfd(t) given by Eq. (6.123) as it should be because Eq. (7.22a) is its
Laplace transform.
7.2 THE LINEARITY PROPERTY OF THE INVERSE LAPLACE TRANSFORM 215
+ +
The function F(s) = (b  a)/(s a)(s b) has two poles, one at s = a and the
other at s = b. From the properties of the RAC determined in Section 6.2, we
found that the RAC always is an interval that is between adjacent poles. Thus, there
are three possible RACs for this function. They are: CJ < a, a < CJ < b, and
CJ > b. The time function given by Eq. (7.24c) is the inverse Laplace transform of
F(s) for the case in which the RAC is a < CJ < b. We would obtain a different
time function if the RAC were one of the other two possibilities. There corresponds a
different time function for each different RAC. It is the RAC that makes the Laplace
transform a onetoone mapping.
To illustrate, let us determine the time functions that correspond with the other
two possible RACs for F(s). For this, we first express F(s) as the sum of two
functions as in Eq. (7.22b):
F(s) =
ba 
1
(s+a)(s+b)s+b
+s +1a (7.25)
(7.26a)
(7.26b)
and
(7.26~)
If the RAC is CJ < a, we must choose CJ,, < a in Eqs. (7.26). Then, for a value of
CJ,,which is less than a, we have that the integral forfi(t) is along a vertical line in
the s plane which is to the left of the pole at s = b. This integral is thus exactly the
same as that in Eq. (7.17) with B = 1 so thatf,(t) = eb'u(t). In the integral for
h(t),we note that the integral is along a vertical line in the s plane which is also to
the left of the pole at s = a. This integral is thus exactly the same as that in
Eq. (7.17) with B = 1 and b = a so that h(t)=  P ' u (  t ) . Consequently, for
CJ < a, we have
We now consider the case for which the RAC of f ( t ) is CJ > b. We then must
choose CJ,, > b in Eqs. (7.26). For a value of CJ,, that is greater than b, we have
that the integral forb ( t ) is along a vertical line in the s plane which is to the right of
216 THE INVERSE BILATERAL LAPLACE TRANSFORM
the pole at s = b. Thus this integral is exactly the same as that in Eq. (7.15) with
A = 1 and a = b so thatfi(t) = e&'u(t). In the integral forh(t), the integral is
along a vertical line in the s plane which is also to the right of the pole at s = a.
Thus this integral is exactly the same as that in Eq. (7.15) with A = 1 so that
h(t)= e"u(t). Consequently, for r~ > b, we have
We note that, depending on the RAC, there are three possible time functions
corresponding to the function F(s) given by Eq. (7.25). If the RAC is r~ < a,
then the time function is that given by Eq. (7.27). If the RAC is a < 0 < b, then
the time function is that given by Eq. (7.24c). If the RAC is 0 > b, then the time
function is that given by Eq. (7.28). For uniqueness of the mapping, the RAC must
be specified! This is why the Laplace transform of any function must include a
specification of the RAC.
We shall see in Section 8.3 that an important class of Laplace transforms in
system theory is that in which F(s) can be expressed as the ratio of finitedegree
polynomials:
(7.29)
Such functions are called rational functions. The function F(s) in Eq. (7.25) is an
example of such a function in which m = 0 and n = 2. Our determination of the
inverse Laplace transform was facilitated by expressing F(s) as the sum of simple
fractions as in Eq. (7.25). This is true even in the general case given above. A
procedure for obtaining such an expression is called partial fraction expansion,
which is discussed in the next section.
(7.31)
rational function can be expressed as the sum of simple fractions as in Eq. (7.25) so
that the inverse Laplace transform can be determined as in Section 7.2. In our
development, please observe that the partial fraction expansion is an algebraic
identity that is valid for all values of s and not just for values of s in some particular
range.
Before discussing the partial fraction expansion, note that any rational function
for which m 2 n can be expressed as the sum of an (m  n)degree polynomial plus
a strictly proper rational function by dividing the denominator polynomial into the
numerator polynomial. As an example, consider
+6
3s2  5s
2s2 + 4s + 346s" + 2s3 + s2 + 4s + 5
sS4+ 12s3 + 9s2
 1oS3  8s2 + 4s + 5
(7.32b)
 1os3  2oS2  15s
12s2 + 1Is + 5
12s2 + 24s + 18
13s + 13
Now, for the partial fraction expansion of a strictly proper rational function, we
first must factor the denominator polynomial in Eq. (7.31). In accordance with our
discussion of the fundamental theorem of algebra in Appendix A, there are exactly n
roots because the denominator is an nthdegree polynomial. Thus its factored form is
The roots are denoted byp, for k = 1,2, . . . , n because they are the poles of F(s). If
the roots of q factors are equal, the root is said to be a qthorder root. For example, in
the fourthdegree polynomial
the roots at s = 2 + j and s = 2 j are firstorder roots. Firstorder roots are usually
called simple roots. Because the root at s = 3 occurs twice, it is called a second
order root. We first describe the partial fraction technique for the case in which all
denominator roots, the poles of F(s), are simple.
Case for Which All Poles Are Simple The partial fraction expansion of a
strictly proper rational function in which all poles are simple is
F(s) =
+ + + a l s + a.
urnsrn aml~rnl . . .
bnsn+ bn1sn' + . . . + bls + bo
The form of the expansion in Eq. (7.34a) is obtained by choosing the most general
form for which the least common denominator is that of the given F(s). If the
fractions of the expansion were added together, the numerator of the resulting
rational function would be a polynomial of degree less than n so that the sum
would be a strictly proper rational function with the desired denominator. The
constants ck could be determined by choosing them so that the numerator of the
sum is the same as that of the given F(s). However, a better method is to use
Eq. (7.34b) to determine them. To shown the validity of this equation, consider
the case for k = 1. Then from Eq. (7.34a),
(SpI)F(s)=cl + SPn
(7.35)
The second term on the righthand side of Eq. (7.35) is zero when s = p 1 because
p k # p l for k = 2,3, . . . ,n. Thus we obtain c1 in accordance with Eq. (7.34b) by
letting s = p l . This establishes the validity of Eq. (7.34b) for k = 1. The proof is
similar for any other value of k. As an illustration, consider
+ +
3 2 2s 1
+ ++
F(s) = C1 c2 c3 (7.36a)
+
(s + +
l)(s 2)(S 3)  s 1 s +2 s +3
7.3 THE PARTIAL FRACTION EXPANSION 219
=
3s2 + 2s + 1 32+1
=1 (7.3 6b)
CI
+ 2)(s + 3)
(s
s= I
3s2 + 2s + 1 124+1
c2 = 
 = 9 (7.36~)
(s + l)(s + 3) s=2 (1)(1)
and
(7.36d)
3s2+2s+ 1  1 9 11
F(s) =
(s+ l ) ( s + 2 ) ( s + 3 )  s + 1
++
s+2 s+3
(7.3 6e)
Case for Which There Are HigherOrder Poles If some of the poles are not
simple, the procedure described above must be modified because the form given by
Eq. (7.34a) is not the most general form for which the least common denominator is
that of the given F(s). To simplify our discussion, we first consider the case in which
there is one nthorder pole and no other poles. We then will extend the partial
fraction expansion technique to include cases in which there are poles of various
orders.
The partial fraction expansion of a strictly proper rational function in which there
is only one nthorder pole and no other poles is
(7.37a)
(7.37b)
220 THE INVERSE BILATERAL LAPLACE TRANSFORM
The form of the expansion in Eq. (7.37a) is obtained by choosing the most general
form for which the least common denominator is that of the given F(s). If the
fractions of the expansion were added together, the numerator of the resulting
rational function would be a polynomial of degree less than n so that the sum
would be a strictly proper rational function with the desired denominator. The
constants could be determined by choosing them so that the numerator of the sum
is the same as that of the given F(s). A better method to determine the coefficients is
to use Eq. (7.37b). The validity if this equation is easily verified by noting from
Eq. (7.37a) that
d2
(s  p J F ( s ) = (n  2)(n  l ) C l ( S  p I ) n  3+ (n  3)(n  2)c,(s  p l y 4
ds2
+ . . . + c,3(2)(3)(s PI) +m ,  2 (7.38C)
S=PI
This is Eq. (7.37b) for k = 2. By continuing this process, we arrive at the general
form for Eq. (7.37b).
To illustrate this method, consider
s2+3
F(s) = ~
+  Cl
+ +++
c2
(s 4)3 = (s 4) (s 412 (s +c3413 (7.39a)
7.3 THE PARTIAL FRACTION EXPANSION 221
and
1 d2 1 d2 1
[e
c , = 2 ds2 + 4 ) 3 ~ ( s )s=4
] = [dsi(s2 + 311
s=4 = 2[2] = 1 (7.39d)
s2 + 3 1 8 19
F(s) = ~
(s +4p 
+
++
 (s 4) (s + 4)2 (s + 4)3 (7.39e)
3s3
F(s) = (7.3loa)
(s + 1)2(s+ 2)
This rational function is not strictly proper because the degree of the numerator is
m = 3 and the degree of the denominator is n = 3. Thus the given function can be
expressed as a strictly proper rational function plus a polynomial with the degree
m  n = 3  3 = 0. The polynomial thus is just a constant, co. The strictly proper
rational function can be expanded in fractions. From our discussion above, the
general form of the total expansion of F(s) must be
so that co = 3. Also
(s
3s3
+ 112 
+A
(s +] 112( s + 2) + c3 (7.31Od)
The first term on the right is zero for s = 2. Thus, be letting s = 2 in this
equation, we obtain c3 = 24. Note that this is the same technique of obtaining
the coefficients for simple pole as discussed above. Now, following the procedure for
higherorder poles, we consider
(s + 1)2F(s)= s 3s3
+2
= cl(s + 1) + c2 + [co + $ ] ( s + 1)2 (7.31Oe)
(7.31Of)
d
(s
ds
+ 2)2F(s)= dd s s3s3

+2
(7.3 1Og)
so that
d
[z(s + 2 ) W S ) ]s= 1
[ 1
= d 3s3
ds s + 2
= 12 = C] (7.310h)
3s3 12 3 24
F(s) =
(s + q2(S +2) =3+
s+ 1
++
(s+ 1)2 s+2
(7.31Oi)
c2, and c3 were determined beore c1. We thus could have determined c1 by starting
with
3 24
F(s) =
(s +
3s3
U2(S + 2) =3+ C1
++
s + l (s+1)2 s+2
(7.31Oj)
This is an equation with only one unknown. Because this equation is valid for all
values of s, we just need to choose one value of s and solve for c,. A convenient
value to choose is s = 0 for which we have
We have shown in Section 7.1 that the Laplace transform is a onetoone mapping so
that Eqs. (7.13a) and (7.13b) are a transform pair. It is for this reason that tables of
Laplace transforms are so usefd as illustrated in Sections 7.1 and 7.2. Often,
extensive tables are not necessary. The reason is that the inverse Laplace transform
often can be determined with a short table by using the partial fraction expansion
and Laplace transform properties.
Table 7.41 lists some of the specific Laplace transform pairs that we have
determined. Reference equation numbers are given for some listed transform pairs
so that you can review their determination. Some of the listed pairs are slight
generalizations of those given by the reference equations. Entries with numbers
followed by a letter are special cases of the entry with the same number without a
letter. In the table, z = a +jb.
Table 7.42 lists some of the specific Laplace transform properties that were
shown in Section 6.3. In the table, the Laplace transform offi(t) is F,(s) with the
RAC (To < (T < Ob.
We'll illustrate the technique for determining the inverse Laplace transform with a
few examples from which we shall draw some important general conclusions.
Example 1 For our first example, we determine the inverse Laplace transform of
3s3
F(s) = 2 < (T < 1 (7.41)
(s + 1I2(s+ 2) '
224 THE INVERSE BILATERAL LAPLACE TRANSFORM
TABLE 7.41 Some of the Specific Laplace Transform Pairs That We Have Determined
No. f( t ) F(4 RAC Ref. Eq.
1 I co<a<cc
( n  l)!
2 n = 1 , 2 , 3 ,. . . a > a 6.332
(s + z)"
1
2a a > a 6.17
s+z

1
2b a10
S
(n  l)!
3 n=1,2,3, ... a < a
(s + z)"
1
3a a < a 6.113
s+z
1

3b a<o
S
4
(s + a)cos(4)  coo sin(4) a > a 6.120
(s + a)2 + W:
4a a > a 6.121
4b a > a 6.122
TABLE 7.42 Some of the Specific Laplace Transform Properties Shown in Section 6.3
No.
1
2
3
4
5
6
7
7.4 CONCLUDING DISCUSSION AND SUMMARY 225
For this, we first require the partial fraction expansion of F(s). This was determined
in Section 7.3. We have from Eq. (7.31Oi)
12 3 24
F(s) = 3 + ~  ~ 2 < o < 1 (7.42a)
s + l (s+1)2 s+2'
Thus, using the technique discussed in Section 7.2 and Table 7.41, we have
Example 2 For our second example, we shall use the Laplace transform properties
to determine the inverse transform of
s+a
F(s) = o > a (7.43)
+ +
(s a)2 m; '
This is pair 4a of Table 7.1 1. However, we shall obtain the inverse transform of this
function by a circuitous route in order to illustrate how the transform properties can
be used to manipulate a function. We first note that by replacing (s a ) with s, we +
obtain
S
F,(s)= o>o (7.44a)
~
s2 + m; '
so that by use of property 4, we have
Thus we just need to determine fi (t). For this, we could expand F , (s) by partial
fractions. However, to hrther illustrate the uses of the properties, note that
F , (s) = sF2(s)where
1
F2(s) = o>o (7.45a)
~
s2 + m; '
so that, by property 6,
(7.45b)
226 THE INVERSE BILATERAL LAPLACE TRANSFORM
S2

+ 0; (S +jOo)(S jOo) j20, S
+
+j00 5200 S  j W o
, a>O
(7.46a)
Using the technique discussed in Section 7.2 and transform pair 2a, we then obtain
1 1 1 .
+
~ ( t=)ejwofu(t) dwofu(t)=  sin(wot)u(t) (7.46b)
5200 5200 0 0
The real form forf,(t) is obtained by using the identity derived in Appendix A,
Eq. A15. We then obtain from Eq. (7.45b)
Note that the derivative, f i(t), does not have an impulse at t = 0 because f,(t) is
continuous there. We now use Eq. (7.44b) to obtain
32 + 2s + 1 1 (7.4loa)
Fl(s) = (s + l)(s + 2)(s + 3) ' cT>
then
To determineh(t), we fist obtain the partial fraction expansion of F,(s). This was
determined in Section 7.3. From Eq. (7.36e),
1 9 11
F,(s) =  
s + l s+2 s+3'
+ a>1 (7.41 la)
PROBLEMS 227
Thus using the technique discussed in Section 7.2 and also using transform pair 2a,
we have
PROBLEMS
71 For each function given below, determine all possible RACs and the inverse
Laplace transform for each possible RAC.
1 1
(a) FAs) =  
s+2
+
s+3
1 I 1
@) Fd.4 =   
s2
+
s+3 s5
+
72 For each function given below, determine all possible RACs and the inverse
Laplace transform for each possible RAC.
(a) F,(s) = [s:
2 +
s+3
1
le'.
e' 8
(b) F~(s)
= 
s+2 s+3
+
73 For each function given below, determine all possible RACs and the inverse
Laplace transform for each possible RAC.
1 1
(a) F,(s) =
+ + +
(s 2)
~
(s 312
228 THE INVERSE BILATERAL LAPLACE TRANSFORM
75 Determine the inverse Laplace transform of each of the following functions:
76 Determine then inverse transform of F(s) = e'', co < cs < 00.
78 A simplified form of a transform that arises in feedback systems with delay is
F(s) = 1/(1  e'). cs > 0. In this problem, we illustrate one method for
determining the inverse transform, f ( t ) .
(a) We first determine the location of the poles. They are at those values
of s for which e' = 1. To solve this equation, we have e' =
e(.+Jo) = e'eJW = 1. Show that the solutions of this equation are
s =j2kn for k = 0, f l , f 2 , . . . so that there are infinitely many poles
uniformly distributed along the o axis and the RAC is to the right of all
the poles.
(b) To determine the inverse transform, we expand F(s) in a power series.
Show that the power series expansion is
03 00
710 Use pair 4b of Table 7.41 and property 7 of Table 7.42 to obtain the Laplace
transform o f f ( t )= te'' sin(o,t)u(t).
s1
4. F4(s)=
s[(s + 212 + 41
CHAPTER 8
The timedomain relation between the output, y(t), of an LTI system and its input,
x ( t ) , was derived and analyzed in Chapters 2 and 3. It was shown there that the
output, y(t), is equal to the convolution of the input, x ( t ) , with the system unit
impulse response, h(t):
00
*
y(t) = h(t) ~ ( t=) (8.1 1a)
and this equation is valid only for 0 in the overlap of the RACs ofy(t), h(t), and x(t).
The Laplace transform of h(t), H(s), is called the system function of the given
system. Recall that H ( j w ) is the transfer function while H(s) is the system function
of the given system. We can let s = j w in the system function, H(s), to obtain the
transfer function, H ( j w ) , only if the w axis, 0 = 0, lies in the RAC of h(t). Observe
that the relation between the input and output is a simple algebraic expression in the
231
232 LAPLACE TRANSFORM ANALYSIS OF THE SYSTEM OUTPUT
to the input
The system function, H(s), is the Laplace transform of h(t). From Table 7.41, we
have
1 S
H ( s ) = 1 = fJ> 1 (8.14a)
s+l s+l’
and
E E 4E
X ( s ) = +  2<0<2 (8.14b)
s2 s+2(s2)(s+2)’
Y(s) =  4Es
(s  2)(s +
2)(s + 1) ’ l<CJ<2 (8.15a)
The RAC of Y(s) is determined in the following manner: First. in accordance with
the interval property of the RAC, it must be an interval between the poles of Y ( s ) so
that the only possible choices are f~ < 2, 2 < cr <  1,  1 < f~ < 2, and cr > 2.
Second, in accordance with Eq. (8.12), the RAC ofy(t) must be chosen so that there
is an overlap of the RACs of x(t), y(t), and h(t). The only interval that satisfies these
conditions is the one given in Eq. (8.15a). We now determine y(t) as discussed in
Section 7.4. The partial fraction expansion of Y(s) is
2 1
Y(s) = E
3 s2
+ 2Es +12  E,
4
3 s+l
1
1<a<2 (8.15b)
(8.16)
8.1 THE LAPLACE TRANSFORM OF THE SYSTEM OUTPUT 233
In this example, the input, x(t), isfd(t) in Example 4 of Section 6.1 with a = 2 and
b =  2 . It was shown in Section 6.1 that the Laplace transform of f d ( t ) does not
exist if a p b. Thus we could not determine y(t) as in this example if the system
input, x ( t ) , werefd(t) with a I6. However, for such cases, note that from Eq. (6.1
24) we can expressfd(t) as the sum
With the use of superposition, the LTI system response can then be expressed as
where y,(t) is the system response tof,(t) and yb(t) is the system response tofb(t).
From Eq. (8.12) we then could use Laplace transforms to determine
The inverse Laplace transforms of each of these expressions then can be determined
as in the example above to obtain y,(t) and yb(t).The total output, y(t), then is the
sum of the two functions.
To illustrate this technique, let the input of the given system be
We'll obtain the general expression for y(t) by the procedure described above and
then choose specific values for a and b. For this case,
1 1
F,(s) = ~,
s+a
0 > a, Fb(4 = s+b 1 CJ < b (8.110)
s
1
Y,(s) = H(s)F,(s) = ~~ CJ > a (8.11 la)
s+ ls+a'
and
s 1
Y,(s) = H(s)F,(s) = ~~  1 < CJ < b (8.11 lb)
s+ls+b'
234 LAPLACE TRANSFORM ANALYSIS OF THE SYSTEM OUTPUT
Expanding the above expressions by partial fractions and using Table 7.41, we
obtain
1
y,(t) = ~ [ e d (8.112a)
1a
and
(8.112b)
4
If a = 0 and b = (Remember, this solution is valid only for a 5 b < 1) we have
For the special case in which a = b = 0 so that x(t) = 1 for all t, note that the system
response is y(t) = 0. The reason is that the system dc gain is zero because H ( 0 ) = 0.
This characterizes the recording gauge. Now, while in operation, let the observed
output pressure recording be
This pressure reading starts at zero and gradually rises to three units. What was the
actual pressure variation within the vessel? For this, we have from Eq. (8.12) that
(8.1 16a)
3 3 3
Y(s) =   = ~ ~
o>o (8.1 16b)
s s+l s(s+l)’
(8.1 17a)
From Eqs. (8.1 15) and with the use of Table 7.41, we have
P P 3P
Y1(s)== a>o (8.1 17b)
s s+3 s(s+3)’
and
P
X,(s) =  , a>0 (8.1 17 ~ )
S
so that
3P s 3
H(s) =  a > 3 (8.1 17d)
s(s + 3 ) P  s + 3 ’
As in the first example, the RAC was chosen so that the RACs of the three functions
overlap. Finally we have
+
236 LAPLACE TRANSFORM ANALYSIS OF THE SYSTEM OUTPUT
Again, the RAC was chosen so that the RACs of the three functions overlap. The
partial fraction expansion of X ( s ) is
3 2
X(s)=  0>0 (8.1 18b)
s s t l '
In contrast to y(t), the observed gauge reading, we observe that x(t), the actual
pressure within the vessel, really started at one unit and gradually rose to three units.
Example 3 As our third example, we choose a problem type that occurs in system
identification. The response of a given LTI system is observed to be
to the input
What is the unitimpulse response to the given system? In the time domain, we are
required to determine h(t) given x(t) and y(t) in Eq. (8.11). A problem of this type is
called an integral equation because the unknown function, h(t), is part of the inte
grand just as a differential equation is one in which the unknown function is differ
entiated. Integral equations are often difficult to solve. However, integral equations
of the convolution type with which we are concerned are not difficult to solve
because we can determine the unitimpulse response of a given LTI system by
using Eq. (8.12). First, the Laplace transforms of the input and the output are
2 1  s+l
X ( s ) =   D > 2 (8.12 la)
s + 3 s+2(s+2)(s+3)'
and
1
Y(s) =  D > 2 (8.12 1b)
s+2'
Y(s) = s
H(s) =  + 3 = 1 + 2 (8.122)
X(s) s
~
+1 s+l
8.2 CAUSALITY AND STABILITY IN THE sDOMAIN 237
The problem now is to choose the RAC of h(t) so that its RAC overlaps that of x(t)
and y(t). Note that only two possible RACs of h(t) are a < 1 and a > 1. If we
choose a <  1, then there is the overlap 2 < rs <  1. If we choose a >  1, then
there is the overlap a >  1. Thus, there is an overlap with either choice. This means
that either choice is mathematically valid! If we choose the RAC to be u <  1, the
unitimpulse response would be
Either one of these systems would produce the given output for the given input.
Note that, for other inputs, the outputs of the two systems would differ so that
we could determine which of the two possibilities is the given system by observing
its response to other inputs. However, the given data are all that are available in our
problem. Thus the choice must be made using other considerations. For example, in
accordance with our discussion of causality in Section 3.5, we can be wellassured
that the system is causal if it is a physical one. Thus, if the unknown system is
known to be a physical system, we would choose the system unitimpulse response
to be h2(t) given by Eq. (8.123b).
We observed in the last section that many types of analyses concerning LTI systems
are more easily done using Laplace transforms. A niajor concern in such analysis is
causality and stability. In this section, results we already obtained will be used to
analyze the causality and stability of an LTI system only from its system function,
H(4.
8.2A Causality
We showed in Section 3.5 that an LTI system is causal if and only if its unitimpulse
response, h(t), is equal to zero for t < 0. To examine what this condition imposes on
H ( s ) , we use the result obtained in Section 6.2, which was that if a functionf(t) = 0
for t < 0, then the RAC off(t) is to the right of all the poles of its transform, F(s).
However, we showed there that the converse is not necessarily true. That is, it is not
necessarily true that if the RAC off(t) is to the right of all the poles of F(s), then
f(t) = 0 for t < 0. From this, we immediately have the result: A necessary (but not
suficient) condition that an LTI system be causal is that the RAC of h(t) be to the
right of all the poles of H(s).
An important case that we shall discuss later is that for which H(s) is a rational
function. For that case we refer to our discussion in Example 3 of Section 7.4, where
238 LAPLACE TRANSFORM ANALYSIS OF THE SYSTEM OUTPUT
it was shown that if F(s) is a proper rational function, then its inverse transform,f(t),
equals 0 for t < 0 if and only if the RAC off(t) is to the right of every pole. From
this we immediately have the following result:
If the system function, H(s), is a proper rational function, then the LTI system is causal
if and only if the RAC is to the right of every pole of H(s).
For the case in which H(s) is a rational function but not a proper rational func
tion, we can express H(s) as a polynomial plus a strictly proper rational function as
discussed in Section 7.3. For example, let H(s) be the rational function given by Eq.
(7.32a) so that
We then have
If the system function, H(s), is a rational function (not necessarily proper), then a
necessary and sufficient condition that the system be causal is that the RAC be to the
right of every pole of H(s). If H(s) is not a rational function, then the RAC being to the
right of every pole is a necessary but not a sufficient condition for the LTI system to be
causal.
8.2B Stability
We showed in Section 3.6 that an LTI system is BIBOstable if and only if h(t) is an
L , function. That is, if and only if
(8.23)
From the L , property of the RAC discussed in Section 6.2, we immediately have the
following result:
An LTI system is BIBOstable if and only if, in the s plane, the o axis lies in the RAC
of h(t)
8.2 CAUSALITY AND STABILITY IN THE %DOMAIN 239
To understand this result in more depth, recall our discussion in Section 3.6,
where it was shown that if the output of a given system contains the derivative of
the input, then that system is not BIBOstable. The reason was based on the discus
sion in Section 3.3, where it was shown that the derivative of a bounded waveform
that is discontinuous at some instant contains an impulse at that instant. Because the
impulse is not bounded, this would be a bounded input waveform for which the
output is not a bounded waveform.
With this in mind, let us consider the important special case in which H ( s ) is a
rational function. For example, let H(s) be the rational function given Eq. (8.21).
The Laplace transform of the output is then given by Eq. (8.22). As discussed
d2 d
above, the inverse transform of the first two terms is 3 x(t)  Sx(t). If the
dt2 dt
input is the bounded waveform x(t) = sin(wot)u(t), then the output due to these
+
terms is 3w06(t)  [3wi sin(wot) 5w0 cos(oot)]u(t), which is unbounded due to
the impulse. We can see that, generally, the system output will contain terms that are
the derivative of the system input if the degree of the numerator polynomial of H(s)
is greater than that of the denominator polynomial. In such a case there always is an
input for which the output contains an impulse as in our example. We thus observe
that the system is not stable if the degree of the numerator polynomial exceeds that
of the denominator polynomial. We thus conclude the following:
S2
H(s) = (8.24)
(s  2)(s + 3)
This is a proper rational function with a secondorder zero at s = 0, a pole at s = 2,
and a pole at s =  3 . In accordance with the interval property of the RAC, there are
three possible RACs: CJ > 2, 3 < CJ < 2, and CJ < 3. Let us consider the causality
and stability of this system for each of these cases.
1. If the RAC were CJ > 2, the system would be causal because the RAC would
be to the right of both poles. However, the system would not be stable because
the RAC does not contain the w axis (which is the line CJ = 0).
2 . If the RAC were 3 < CJ < 2, the system would not be causal because the
pole at s = 2 is to the right of the RAC. However, the system would be stable
because the RAC would contain the w axis and H(s) is a proper rational
function.
3 . If the RAC were G < 3, the system would not be causal because there are
poles to the right of the RAC. Also, the system would not be stable because
the w axis would not be in the RAC.
240 LAPLACE TRANSFORM ANALYSIS OF THE SYSTEM OUTPUT
We thus note that the LTI system with the system function given by Eq. (8.24)
cannot be both causal and stable. For it to be both causal and stable, we would
require all the poles to be to the left of the RAC and would also require the RAC to
contain the o axis. But both these conditions can be satisfied only if all poles lie in
the left half of the s plane. We thus observe the following:
For the case in which H(s) is a rational function, the system is both stable and
causal if and only if H(s) satisfies the following conditions:
The System lnverse The system inverse was defined in Chapter 1, Section 1. It
was shown there that, for any system, a system inverse exists if and only if the
system mapping is onetoone. Unfortunately, this condition is not very useful in
determining a system inverse. In practice, we are concerned not only with the
existence of an inverse, but also with determining the inverse if one exists. This
determination can be done for LTI systems with the results we now have obtained.
For this, consider Figure 8.21. In the figure, the system function of the given system
is H(s). It is connected in tandem with its system inverse with the system function
G(s). Observe that the tandem connection is an LTI system with an output equal to
its input. Thus the unitimpulse response of this system is d ( t ) , for which the Laplace
transform is 1 and the RAC is all values of CJ. Therefore the system function of the
tandem connection is
H(s)G(s)= 1 (8.25)
This equation is valid for all values of 0 which lie in the overlap of the RACs of H(s)
and G(s). The algebraic solution of this equation for G(s) is
(8.26)
; x(t
I y(fy G(s)
,........................................................................... !
Consider the case for which H(s) is a rational function. Clearly then, G(s) is a
rational function. From our previous discussion, the system inverse is both stable
and causal if and only if G(s) satisfies the following conditions:
Because G(s) is the reciprocal of H(s), we note that the poles of G(s) are the zeros
of H(s) and the zeros of G(s) are the poles of H(s). Thus we can translate the
conditions for the causality and stability of the system inverse to conditions on
the given system. For example, if the given system is causal and stable, then a
causal and stable system inverse exists if and only if H(s) satisfies the following
conditions:
1. H(s) has the same number of zeros as poles (because both H(s) and G(s) must
be proper rational functions).
2. All the poles of H(s) are in the left half of the s plane (because the given
system is causal and stable).
3. All the zeros of H(s) are also in the left half of the s plane (because these are
the poles of G(s) which must be in the left half of the s plane in order for the
system inverse to be both causal and stable).
System functions with all its poles and zeros located in the left half of the s plane
are called minimumphase functions, which we discuss in Section 9.3. They are
important not only in relation to system inverses, but also, as we discuss in Section
8.4, in relation to impedance functions.
The reason for our specific interest in rational functions is that the system function of
any lumped parameter LTI system is a rational function. Such systems are an
important class that we discuss in this section.
There are two major classes of LTI systems for which the input and output can be
related by a differential equation: lumped parameter systems and distributed para
meter systems. In mathematical terms, an LTI lumped parameter system is one for
which the input, x(t), and output, y(t), can be related by a total differential equation
with constant coefficients. The differential equation relating the input, x(t), and
output, y(t), of an LTI distributed parameter system is a partial differential equation
with constant coefficients. This is the essential mathematical difference. Physically, a
lumped parameter system is one in which each of the system elements is located in a
given place (Le., the system elements are lumped). An example of an LTI lumped
parameter system is an electric network that is composed of linear inductors,
resistors, and capacitors whose values do not vary with time. The node and loop
242 LAPLACE TRANSFORM ANALYSIS OF THE SYSTEM OUTPUT
equations of the network are total differential equations with constant coefficients
whose values are determined by the network elements. Another example is a
mechanical system composed of masses, springs, and dash pots whose values do
not vary with time.
A distributed parameter system is one in which the elements are not located in
one place but, rather, are distributed throughout the space occupied by the system.
An example of an LTI distributed parameter system is an electric transmission line.
The resistance, inductance, and capacitance of the transmission line are distributed
along the transmission line wires. The system model of electromagnetic wave propa
gation in free space is a distributed parameter system. Similarly, the system model of
sound propagation in air or water is a distributed parameter system because the
physical parameters that determine the sound wave propagation are distributed
throughout the air or water.
We shall concentrate on lumped parameter systems. In our discussion, some of
the essential differences of the characteristics of the two types of systems will be
pointed out. As stated above, the input, x(t), and output, y(t), of a lumped parameter
LTI system can be related by a total differential equation with constant coefficients:
The coefficients are real numbers whose values are determined by the lumped
parameters of the LTI system. The determination of the output, y(t), for all time
obviously requires knowledge of the input for all time. If the input is only known
after some time instant, to, and not before to, then the output can be determined for
t > to only if the system initial conditions are known. In this section we shall use
Laplace transforms to analyze the case in which the input is known for all time. In
Section 8.5, we'll use the Laplace transform to analyze the case in which the input is
only known after some time instant.
To obtain the Laplace transform of Eq. (8.31) for the case in which x ( t ) is
known for all time, we first note from property 6 in Table 7.42 that the RAC
of the derivatives of x ( t ) is at least that of x(t). Thus the RAC of each term of the
righthand side of Eq. (8.31) overlaps so that, from property 1 of Table 7.42, the
Laplace transform of the righthand side of Eq (8.31) is equal to the sum of
the Laplace transforms of each term with the RAC being that of x(t). Similarly,
the Laplace transform of the lefthand side of Eq. (8.31) is equal to the sum of the
Laplace transforms of each of its terms, with the RAC being that of y(t). Now, the
lefthand side of Eq. (8.31) is equal to its righthand side so that the Laplace
transform of its lefthand side is equal to that of its righthand side. Thus, with
the use of property 6 in Table 7.42, the Laplace transform of Eq. (8.31) is
with cs in the overlap of that ofx(t) andy(t). This is an algebraic equation that we can
factor
Y(s)=
+ + + + w
urnsrn arnl~rnl . . . a l s a.
(8.3 2 ~ )
b,s" +b,ls"' + +
. . . + bls bo
with cs in the overlap of the RACs of x ( t ) and y(t). This equation is in the form
where
H(s) =
+
urnsrn arnlsrnl+ . . . + a,s + a o
(8.33b)
b,s" + +
b,l~"' . . . + bls + bo
The inverse transform of Eq. (8.33a) is, from transform property 5 of Table 7.42,
where H(s) is the Laplace transform of h(t) so that H(s) is recognized as the system
function of the lumped parameter LTI system. We thus observe from Eq. (8.33b)
that the system function of any lumped parameter LTI system is a rational function.
The denominator polynomial of H ( s ) is the coefficient polynomial of Y(s), and the
numerator polynomial is the coefficient polynomial of X(s). Consequently,
The poles of H ( s ) are the roots of the coefficient polynomial of Y(s), and the zeros of
H ( s ) are the roots of the coefficient polynomial of X(s).
Note that if H ( s ) is known, then the differential equation relating x(t) and y(t) can be
obtained by starting with Eqs. (8.32c), crossmultiplying to obtain Eq. (8.32b), and
then using Eq. (8.32a), from which the differential equation, Eq. (8.3l), is
obtained. In this manner, we not only can obtain H ( s ) from the differential equation,
but we also can obtain the differential equation from H(s). Observe that the system
function, H(s), is a rational function if and only if the LTI system is a lumped
parameter system. To determine the RAC of H(s), we must know something
about the system being analyzed. For example, if the system is known to be
causal, then, from our discussion in the previous section, the RAC must be to the
right of all the poles.
In general, the Laplace transform of an impulse response is not necessarily a
rational function. Consequently, because the system function of a lumped parameter
244 LAPLACE TRANSFORM ANALYSIS OF THE SYSTEM OUTPUT
system must be a rational function, not every impulse response can be realized by a
lumped parameter system. For example, consider a delay system for which the
output, y(t), is the input, x(t), delayed by to seconds so that y(t) = x(t  to) for
any input, x(t). From the timeshift property of the Laplace transform (property 3
of Table 7.42), the Laplace transform of this equation is Y(s) = X(s)ePS'o,with the
RAC being that of x ( t ) . Thus, in accordance with our discussion in Section 8.1, the
system function of the delay system is H(s) = e''o, with the RAC being the whole s
plane. This system function can be that of a distributed parameter system, but it
cannot be that of a lumped parameter system because the exponential function, e''o,
is not a rational function. This means that a delay system cannot be realized by a
lumped parameter system. This is a fundamental theoretical restriction and not just a
practical limitation that can be overcome someday as technology develops. However,
all is not lost. By approximating the exponential function e"o by a rational function,
we can obtain an approximate model of a delay system by a lumped parameter
system.
Assume B = 0 lies in the RAC of x(t). We then have Y ( j w )= H ( j w ) X ( j o )so
that
(8.35)
(8.36)
From this result, we observe that to approximate a delay of to seconds, we then only
need to make H ( j w ) % eJoto in the frequency range 0 5 w < W . This observation
is the basis of obtaining a lumped parameter system for which y(t) % x(t  to) for
such input waveforms.
There are many techniques available for obtaining this approximation. Each
technique results in a different type of approximation, so that the specific technique
that one should use is determined by application of the approximate delay system.
One approximation technique that is used is the Pad6 approximation. A Pade
approximation technique is one in which the power series of a given function is
approximated by a rational function in which the degree of the numerator polyno
mial is M and the degree of the denominator polynomial is N . The coefficients of the
polynomials are chosen so that the power series expansion of the rational function
agrees with the power series expansion of the function H(s), being approximated
8.4 PASSIVE SYSTEMS 245
+
through the term of degree M N . For example, the first two Pad&approximations
for which M = N of the system function H(s) = P ' O are'
tos  2 2
H , (s) =   rJ>  (8.37a)
+
tos 2 ' to
and
tis'  6tos + 12 3
rJ> (8.37b)
H2(s) = tis2 + 6tos + 12 '

t0
The output of each of these causal lumped parameter systems will be approximately
equal to the input delayed by to seconds if to is sufficiently small. For the same input,
x(t), the output of the lumped parameter system with system function H2(s) will be
better approximation to x(t  to) than the output of the system with the system
function H , (s). Note, however, that these systems are only approximate models of
a delay system. Depending on the application, there are other possible approxima
tions of a delay system by a causal and stable lumped parameter system. It is
important to remember that approximate models of delay are required because it
is theoretically impossible for a lumped parameter system to model a delay system
exactly; for that, a distributed parameter system is required.
Causality and stability impose certain constraints on the unitimpulse response, h(t),
of an LTI system. In Section 5.1 1 we discussed a number of the constraints they
impose on the transfer function H ( j o ) , and in Section 8.2 we discussed a number of
the constraints they impose on the system function H(s). Additional constraints on
the system function arise from other physical considerations such as that discussed
in Section 8.3. In this section we shall illustrate how physical properties of a system
can be used to determine properties of the system function by determining some of
the properties of the impedance function of a passive system. Our considerations
also will serve to review and lend further physical significance to some of the results
we have developed. For our discussion, we shall use an electric network in which the
terminal quantities are voltage and current. Our discussion, however, also applies to
passive mechanical networks in which the terminal quantities are, for example, force
and velocity.
' A table of Pad6 approximations for various values of A4 and N is in Wall, H. S., Continued Fractions,
Chapter 20, Van Nostrand, New York, 1948.
246 LAPLACE TRANSFORM ANALYSIS OF THE SYSTEM OUTPUT
q(t) = f
m
i(z) dz (8.42)
Physically, u(t) = z(t) volts if (at some instant that we call t = 0) a charge of one
coulomb were placed on the terminals. In accordance with our discussion in Section
3.5, we expect the system to be causal because it is a physical one. Thus we expect
z(t) = 0 for t < 0. Although one cannot prove that every physical system must be
causal, we will prove that every linear (timeinvariant or timevarying) passive
system must be causal. This very interesting and important result strengthens our
unproven conviction discussed in Section 3.5 that all physical systems are causal.
p ( z ) dz =
SI, u(z)i(z)dz joules (8.44)
We define a passive system as one for which w(t) 2: 0 for all t . We would have
w(to) < 0 at some instant t = to,if the system delivered more energy than it received
up to that instant. Thus a passive system is one that can never deliver more energy
than it has received.
Causality From our basic discussion of causality in Section 3.5, a linear time
invariant or timevarying system is causal if, for any value of to,the system response
to any input, x(t), for which x(t) = 0 for t < to is y(t) for which y(t) = 0 for t < to.
We shall use this to prove that any linear TI or TV passive system is causal.
For this, we first let u l ( t ) be the output due to an input il(t).Note that the input
i ,( t ) can be any input of our choosing. Then because the system is passive, we have
(8.45)
We now choose another input i2(t) which is zero for t < to and let the system
response to it be u2(t). We now shall prove that if the system is passive then the
system must be causal by proving that passivity requires that u2(t) = 0 for t < to.For
+
this, we consider the system input i(t) = i , ( t ) ci2(t)in which c is some constant
whose value is at our disposal. Because the system is linear, we have by super
+
position that the system response to the input i(t) is u(t) = u1( t ) cu2(t). From Eq.
(8.44), the energy absorbed by the system at the time t = to is
w(to)=
s", u(t)i(t)dt 2 0 (8.46a)
= 1 to
co
[ul(t) + cu2(t)][il(t)+ ci2(t)]dt 2 0 (8.46b)
= / to
cc
ul (t)il( t ) dt +c oo
u2(t)il( t ) dt 2 0 (8.46~)
= wl(t0) +c 1to
cc
u2(t)i,(t)dt 1 0 (8.46d)
The value of each of the two terms in Eq. (8.46b) involving i2(t)is zero because
i2(t)= 0 for t < to and we are only integrating over values o f t less than to.Because
the system is passive, wl(to) 2 0 and w(to)L 0 in Eq. (8.46d). Now c is at our
disposal. Thus if the value of the integral in Eq. (8.46d) involving u2(t) were not
zero, we could choose the value of c so that w(to) 0, in contradiction to the
passivity requirement. Thus we require the value of the integral involving u2(t) to
be zero. Now, if u2(t) # 0 for t < to,then, because i l ( t )is any input of our choosing,
we could choose it so that the value of the integral in Eq. (8.46d) is not zero for the
given u2(t). Thus we require u2(t) = 0 for t < to. The restriction t < to is because the
248 LAPLACE TRANSFORM ANALYSIS OF THE SYSTEM OUTPUT
integral is only over that range. We thus have proved that u2(t) = 0 for t < to so that
the system is causal. Consequently,
If the linear passive system is TI, then, in Eq. (8.4l), the unitimpulse response,
z(t) = 0 for t < 0, in accordance with our discussion of causality in Section 3.5.
V ( s )= Z(s)Z(s) (8.47)
The function Z(s) is called the impedance function of the passive network. The RAC
must be to the right of all the poles of Z(s) because we have shown that the system
must be causal.
In network theory, z(t) is called the opencircuit natural response. The reason is
that v(t) = z(t) when i(t) = d ( t ) so that the terminals are an open circuit for t > 0
because i(t) = 0 for t > 0. As we discussed above, the unit impulse of current is
obtained by placing one coulomb of charge on the terminals at t = 0, which causes
the network to “ring” similar to the ringing of a bell. This “ringing” is the sum of a
number of oscillations whose frequencies are called the opencircuit natural frequen
cies. The oscillations cannot grow with time because the system is passive. We
generally would expect the amplitude of all the oscillations to decrease with time.
However, they will not decrease if the network is lossless. It also is possible that the
amplitude of just some of the oscillations may not decrease with time if the network
contains strategically placed lossless components. Thus we conclude that, for a
passive system, the amplitude of z(t) cannot increase with time. This means that
the poles of Z(s), whose locations are the opencircuit natural frequencies, must lie
in the left half of the s plane or possibly on the w axis. We arrive at this conclusion
using our discussion in Section 7.4 from which we have that, because the RAC is to
the right of all the poles of Z(s), a pole in the right half of the s plane would result in
a term of z(t) whose amplitude increases with t . On the other hand, a pole on the w
axis results in a term of z(t) whose amplitude does not increase or decrease with t,
and a pole in the left half of the s plane results in a term of z(t) whose amplitude
decreases with time. Thus we conclude that all the impedance function poles of a
passive system must lie in the left half of the s plane or on the w axis. Because the
RAC is to the right of all poles, the RAC would contain the w axis only if there are
no poles on the w axis.
If the w axis lies in the RAC, we can let s =j w in Z(s) to obtain Z(jw), which is
called the input impedance. Thus the input impedance is the Fourier transform of the
unitimpulse response, z(t). For this case, the input
+
i(t) = A C O S ( W ~4~) (8.48a)
8.4 PASSIVE SYSTEMS 249
would, in accordance with our discussion in Section 4.2, produce the output
where
Because the average value of a sinusoid is zero, we note from this result that the
average value of p ( t ) is the constant term so that
Observe that w(t) would be negative if Pa, were negative. Because w(t) 2 0 for
passive systems, we conclude that, for passive systems,
Thus, the real part of the impedance of a passive system cannot be negative at any
frequency.
where
(8.4 13)
The finction Y ( s )is called the admittance function of the twoterminal network. The
inverse transform of Eq. (8.412) is
03
*
i(t) = y(t) u(t) = (8.414)
In network theory, y ( t ) is called the shortcircuit natural response. The reason is that
i(t) = y ( t ) when u(t) = 6(t) so that there is a short circuit across the terminals for
t > 0 since u(t) = 0 for t > 0. In this form of representation, the roles of i(t)and u(t)
250 LAPLACE TRANSFORM ANALYSIS OF THE SYSTEM OUTPUT
are interchanged from that of our previous discussion. Note that all of our previous
discussions and proofs are valid if the roles of the voltage and current are inter
changed. Thus we can immediately conclude that y(t) = 0 for t < 0. Furthermore,
we conclude that all the poles of Y(s), whose locations are the shortcircuit natural
frequencies, must be in the left half of the s plane or on the o axis and the RAC is to
the right of all the poles. If the o axis lies in the RAC, we can let s = j w in Y(s)to
obtain Y ( j o ) ,which is called the input admittance. Thus the input admittance is the
Fourier transform of the unitimpulse response, y(t). Also, the real part of the input
admittance cannot be negative at any frequency.
Z(s) =
a,sm + .. . + als + ao  a,(s  zl)(s  z), . . . (s  z,) (8.415)
b,s" + . . . + bls + 6, 
b,(s  p l ) ( ~p2). . . ( S pn)
In accordance with our discussion of passive networks, all the poles of a passive
network lie in the left half of the s plane or on the o axis, and the RAC is to the right
of all the poles. Note from Eq. (8.413) that the zeros of Z(s) are the poles of Y(s).
Because the poles of Y(s)also must lie in the left half of the s plane or on the o axis,
we have the result that the zeros of Z(s) also must lie in the left half of the s plane or
on the o axis. Rational functions in which all the poles and zeros lie in the left half
of the s plane are called minimumphase functions. The reason for their name and
some properties of minimumphase functions are discussed in the next chapter.
An Example To illustrate the general results we have obtained, consider the RC
network shown in Fig. 8.42.
The input admittance function of this network is
1
1 (8.416)
Y(s) = = c, s 2 +=+s);+:( , D > 
V(s> 1 ZP
s+
ZP

Fig. 8.42 An RC network.
8.4 PASSIVE SYSTEMS 251
(8.4 17)


(s + 0.190983)(s + 1.309017) io? (T> 1
s+l
1 s+ 1
Z(s) = =
~
Note that the poles and zeros of the impedance function are in the left half of the s
plane. Because the o axis lies in the RAC, we can let s = j w to obtain the input
impedance
(8.419)
$02+4 1
Re(Z(jo)} = (8.420)
(4  w2)2+(;o)2
Note that Re(Z(jo)) 2 0 for all values of w in accordance with Eq. (8.41 1).
There are many other properties of the impedance and admittance functions of
passive systems which can be obtained. For example, it can be shown for lumped
parameter networks that, in Eq. (8.415), the degree of the numerator polynomial, m,
and the degree of the denominator polynomial, n, cannot differ by more than one,
which means that the only possible values of m  n are 1, 0, and 1. It also can be
shown that for s = (T +io, Z(o) is real and also Z(a) 2 0 for (T 2 0. Functions with
these two properties are called positivereal @r.) functions. Many properties of
impedance and admittance functions can be derived using the p.r. property. For
example, using the p.r. property, it can be shown that any poles or zeros on the o
axis must be simple. Because the p.r. property is a fundamental property of impe
dance and admittance functions, the detailed study of p.r. functions is a basic topic of
linear network theory.2
' A n excellent discussion of this topic is contained in Guillemin, E. A,, Synthesis of Passive Networks,
John Wiley & Sons, 1957.
252 LAPLACE TRANSFORM ANALYSIS OF THE SYSTEM OUTPUT
If only the system function, H(s), of the lumped parameter system is known, this
differential equation can be obtained using the technique discussed in Section 8.3.
For our analysis, we choose the starting time of the input, to, to be zero so that
x(t) = 0 for t < 0. Thus y ( t ) = 0 for t < 0 because a physical system is assumed
causal. We desire to know the output, y ( t ) , for t > 0 given the input x(t) for t 2 0 and
also the initial values of the output, y ( t ) , and its derivatives. The initial values, y(O+),
y'(O+), y"(O+), . . . ,y("')(O+), are called the initial conditions. Note that the initial
conditions are determined at t = O+ because the initial conditions are the initial rates
of change of the output. They are determined from the network initial values such as
the capacitor voltages and inductor currents in the case of an electric network.
To illustrate our discussion so far, consider the RC circuit shown in Fig. 8.42.
We desire to determine the terminal voltage, u(t), to an input current, i(t), which is
zero for t < 0 for the case in which the initial voltage of capacitor CI is El and the
initial voltage of capacitor C2 is E2. From Eq. (8.416) we have
8.5 THE DIFFERENTIAL EQUATION VIEW OF LUMPED PARAMETER SYSTEMS 253
1
V(S) = SZ(S) + Z(s) (8.52b)
TP
1
+ i(t)
= i’(t) (8.53)
TP
For our problem, the input is i(t), the output is u(t), and the initial conditions are
u(O+) and u’(O+). To determine the initial conditions, we note from Fig. 8.42 that
u(t) is the voltage across the capacitor C2 so that
u(O+) E2 (8.54)
Also, because u(t) is the voltage across the capacitor C2, we have
1
4 t ) =q2(t) (8.55a)
c
2
1 1
u’(t) = q$(t) = i 2 ( t ) (8.55b)
c
2 c
2
where i2(t) is the current through the capacitor C2. Thus the initial rate of change of
the voltage, u(t), is
u’(O+) = 
1 i,(O+) (8.5 5 ~ )
c
2
Now, because the sum of the currents at any node is zero, we have
Consequently,
(8.56)
The initial conditions for this example are u(O+) and u’(O+) given by Eqs. (8.54)
and (8.56), respectively. For this example, our problem is to determine the output,
254 LAPLACE TRANSFORM ANALYSIS OF THE SYSTEM OUTPUT
u(t), for t > 0, by solving the differential equation, Eq. (8.53), from knowledge of
the initial conditions and the input, i(t), for t 2 0.
It is important to observe that if i(t) contains an impulse at t = 0, then u(O+)
would not be E,. Rather, with the use of Eq. (8.55a) in accordance with our
discussion of Eq. (8.42), u(O+) would then be equal to E, plus the area of the
current impulse divided by C,. Because we are specifying u(O+), we consider i(t) not
to contain an impulse at t = 0, and the effect of any impulse applied to the circuit at
t = 0 is taken into account by the initial conditions at t = O+. Accordingly, as stated
above, the effect of any impulse applied at t = 0 is taken into account by the initial
conditions at t = 0+, and we consider our functions not to contain an impulse at
t = 0.
The Laplace transform simplified our previous analyses of LTI systems, and so
we expect that its use in problems with nonzero initial conditions also will simplify
the solution for the output, y(t). From our discussion and the example above, observe
that we are only interested in the solution of the differential equation, Eq. (8.5l), for
t > 0 because the functions are assumed zero for t < 0 and their initial values are
given.
To obtain the Laplace transform of Eq. (8.5l), we need to reexamine the Laplace
transform of the derivative of a finctionf(t), which is zero for t < 0. Thus we let
f(t) = 0 for t < 0. Also, because the effect of any input impulse at t = 0 is taken into
account by the initial conditions, we only consider time functions,f(t), which do not
contain an impulse at t = 0. Then, from our discussion of the derivative of a
discontinuous function in Section 3.3, the derivative off (t) is
(8.57a)
With the use of the Laplace transform differentiation property, property 6 in Table
7.42, the Laplace transform offi(t) is
We now can determine the Laplace transform of the derivative off;(t). For this,
we note that f;(t) is a function which is zero for t < 0 and which does not contain an
8.5 THE DIFFERENTIAL EQUATION VIEW OF LUMPED PARAMETER SYSTEMS 255
(8.5 12)
7172
Now using the concepts developed in Section 8.3 and the results we have just
developed, the Laplace transform of this equation is
7172
c 2
V(s)
1
= [sZ(s)  i(O+)] + I(s) (8.513)
5
Also, the RACs are to the right of all the poles of V(s)and Z(s) because u(t) and i(t)
are zero for t < 0. Equation (8.513) is a simple algebraic equation, for which the
solution for V ( s ) is
1
'This development from the bilateral Laplace transform is called the O+ form of the unilateral Laplace
transform.
256 LAPLACE TRANSFORM ANALYSIS OF THE SYSTEM OUTPUT
To illustrate this general result for our example, let the circuit values be those used in
Section 8.4 for which C2 = lo' F, z1 = z2 = 2 s, and zp = 1 s. For these values,
Eq. (8.514) is
The specific values for the initial conditions are obtained from Eqs. (8.55). We
observe from this equation that the system function for our example is
s+l s+ 1
+ ++
H(s) = lo5  > 0.191 (8.516)
s2 $s lo5 (s + +
0.191)(s 1.309) '
0
in which the RAC is to the right of the poles because the system is causal. Thus the
system is stable because the o axis lies in the RAC.
To illustrate this result, we determine the voltage, u(t), for a current input which is
i(t) = Au(t). For this step input, we have I(s) = A / s with the RAC G > 0 so that
where i(O+) = A for this case. The partial fraction expansion of this expression is
(8.518)
The RAC is as given because, as we have discussed above, it must be to the right of
all the poles. With the use of entry 2a of Table 7.41, the inverse transform of this
equation is
A
~ ( t=) [4  3.789e0.'91'  0.21 le1.309t]~(t)
105 (8.519)
+
 [1.171~(0+) 0.894[~'(0+) i(0+)][e0.191' e1.309'Iu@>
Observe that u(t) is the sum of two terms: The first term is due only to the input,
and the second term is due only to the initial conditions. If the initial conditions were
zero, then the second term would be zero and u(t) would be equal to the first term.
For this reason, the first term is called the zeroinitial condition response (it also is
called the zerostate response). If the input were zero, then the first term would be
zero and u(t) would be equal to the second term. For this reason, the second term is
8.5 THE DIFFERENTIAL EQUATION VIEW OF LUMPED PARAMETER SYSTEMS 257
called the zeroinput response. We thus note that v(t) is equal to its zeroinitial
condition response plus its zeroinput response. This decomposition of v(t) is
already seen in Eq. (8.514). This is due to the result seen in Eq. (8.513), that
the Laplace transform of v+(t) and its derivatives in Eq. (8.512) are composed of
terms that involve only V ( s ) plus terms that involve only the initial conditions.
Observe that this result is valid even for the general case of systems described by
Eq. (8.51). Thus we have the important result:
The response of an LTI system, y(t), always can be expressed as the sum of its zero
initial condition response and its zeroinput response.
Both terms in Eq. (8.514) have the same denominator as a result of dividing by
the coefficient polynomial of V ( s ) in Eq. (8.513) to obtain Eq. (8.514). Note that
this result is also true in the solution of the general case given by Eq. (8.51). As we
discussed in Section 8.3, the roots of this polynomial are the system function poles.
Thus we note that, in general, the poles of zeroinput term are the system function
poles. If the causal system is stable, then, in accordance with our results in Section
8.2, these poles must be in the left half of the s plane while the RAC must be to the
right of all the poles. Thus with the use of the results in Sections 7.3 and 7.4, the
inverse transform of the zeroinput term for stable systems is seen to be composed of
the sum of terms that decay exponentially with time at an exponential rate that is
determined by the system pole locations. Consequently, the effect of any initial
conditions of a stable system decays with time.
Let us now consider the case for which current input of our example in this
section is i(t) = A cos(2t)u(t). For this input, from Table 7.41,
S
I(s) = A ~ o>o (8.5 20)
s2+4'
CT >0 (8.521)
The first term is the Laplace transform of the zeroinitial condition term, and the
second term is the Laplace transform of the zeroinput term. We shall consider each
term separately.
The inverse Laplace transform of the zeroinput term in Eq. (8.521) is, from Eq.
(8.519),
This is the system response due to the initial conditions. Observe that this term is
zero if the initial conditions are zero. Note that the zeroinput term decays exponen
258 LAPLACE TRANSFORM ANALYSIS OF THE SYSTEM OUTPUT
tially with time at an exponential rate determined by the system pole locations,
similar to our result for the previous case given by Eq. (8.519).
To obtain the inverse Laplace transform of the zeroinitial condition term, we
obtain the partial fraction expansion of the first term of Eq. (8.521), which is
(s + 1)s  (s + 1)s
105~
(s2 + + i)(s’ + 4)  l oSA(~2+ 4 ) ( ~+ 0.191)(s + 1.309)
$9
with the RAC D > 0. The fmt term of this partial fraction expansion is really the
sum of two terms:
c,+C2 
DsE
o>o (8.524a)
s+j2 sj2 s*+4’
+
in which D = C1 C2 and E =j2(C1  C2). However, it is not necessary to deter
mine C, and C2 as discussed in Section 7.4 to obtain D and E. The real constants D
and E can be determined directly by multiplying both sides of the partial fraction
+
expansion equation by (s2 4) and letting s =j 2 to obtain
(8.524b)
By equating the real parts and the imaginary parts of Eq. (8.524b), we obtain two
equations that are solved for the real constants D and E. The desirability of the
summed form is that its inverse Laplace transform is easily obtained from Table 7.4
1. Note from entry 4 of that table that the Laplace transform of
~ ( t=) A COS(CO,~
+)u(t) + (8.525a)
is
S C O S ~ o , s i n 4
X ( S )= A , o>o (8.525b)
S2 + CO;
Then, by equating the numerators in Eq. (8.524a) and Eq. (8.525b), we obtain
D
fp = arctan(&) and A =  (8.526b)
cos fp
With this result and using wo = 2 for our example, the inverse Laplace transform of
the zeroinitial condition term given in Eq. (8.523) is
+
465624 cos(2t  1.36)u(t)  [0.034e0.191‘ 0.063e‘.309‘]u(t) (8.527)
The zeroinitial condition term is seen to consist of a sinusoidal term plus terms that
decay exponentially with time at an exponential rate determined by the system pole
locations. The exponentially decaying terms are called the transient response. They
arise because the sinusoidal input started at t = 0. The sinusoidal term is called the
steadystate response because the zeroinitial condition term approaches
46,562Acos(2t  1.36) as t increases. You should note that this steadystate
+
response is equal to A IH( j2)l cos[2t L H ( j2)] because this is the system response
to a sinusoid that exists for all time that we determined in Section 4.2.
The result we just obtained for our example can be generalized. Consider the
response, y(t),of the system described by Eq. (8.51) to a sinusoidal input. For this,
+
let the input be x ( t ) = A cos(wOt q5)u(t). The Laplace transform of this input is
given by Eq. (8.525b). The poles of X ( s ) are on the w axis. Now, the Laplace
transform of the zeroinitial condition term is observed to be H(s)X(s).Its partial
fraction expansion is the sum of terms with the poles of H(s) and terms with the
poles of X(s). For stable and causal systems, the inverse Laplace transform of the
terms with the poles of H(s) decay exponentially because those poles are in the left
half of the s plane. The inverse Laplace transform of the terms with the poles of X ( s )
is a sinusoid that does not decay with time because those poles are on the w axis.
Thus we note that, for a sinusoidal input, the zeroinitial condition response of a
stable system is composed of terms that decay with time and a sinusoidal term. The
terms that decay with time are called the transient response, and the sinusoidal term
is called the steadystate response. Thus, for a sinusoidal input the response of a
stable system is the sum of a zeroinput term that decays with time, a transient term
that decays with time, and a steadystate term that is a sinusoid. Thus, as time
increases, a causal and stable system “forgets” the beginnings of its input, and its
output approaches its steadystate response. Thus, for the sinusoidal input given by
Eq. (8.520a), the output for large values o f t would be approximately
Because the system “forgets” the beginnings of its input, the system response
approaches its response to a sinusoid that exists for all time which we determined
in Section 4.2. This is a frequencydomain interpretation of our timedomain discus
sion in Section 3.5.
260 LAPLACE TRANSFORM ANALYSIS OF THE SYSTEM OUTPUT
PROBLEMS
81 Use convolution to show that in Example 3 of Section 8.1, an LTI system
with the unitimpulse response h , ( t ) and one with the unitimpulse response
h2(t) have the same output for the given input.
82 For each input, x(t), and unitimpulse response, h(t), given below, determine
the Laplace transform of the system responses, Y(s).Do not forget to specify
the RAC.
(a) x(t) = u(t) and h(t) = e"u(t)
(b) x(t) = ecb'u(t) and h(t) = e"u(t)
(c) x ( t ) = u(t) and h(t) = e" cos(o,t)u(t)
84 For the input x ( t ) = e"'u(t), a > 0, the response of an LTI system is
y(t) = r ( t / T ) . Determine the system unitimpulse response, h(t).
85 The unitimpulse response of a given LTI system is h(t) = e"u(t), M > 0.
(a) Determine the system response, y(t), to the input
x ( t ) = sgn(t) =
I 1
0
1
fort =. 0
for t = 0 .
fort > 0
For this determination, use the method described in Example 1 of Section 8.1.
+ +
@) Observe that 1 x ( t ) = 2u(t) so that H(0) y(t) = 2s(t) in which s(t) is
the system unitstep response. Verify this expectation.
87 The unitimpulse response of an LTI system is h(t) = Kr(t/T). Use Laplace
transforms to determine the system response, y(t), to the input
x ( t ) = A r ( t / T ) . Compare your result with that of Example 3 of Section 2.5.
814 The unitimpulse response of an LTI system is h(t) = e,' COS(W,$ $)u(t). +
Determine a differential equation relating the system input, x(t), and output,
Y(t).
262 LAPLACE TRANSFORM ANALYSIS OF THE SYSTEM OUTPUT
815 Determine the system function for each differential equation given below
relating the input, x(t). and output, y(t), of a causal LTI system.
+ +
(a) y”(t) 2y’(t) 3y(t) = ? ( t ) +
2x(t)
(b) f(t) + + +
2y’(t) 3y(t) = X”(t  2) 2x(t  1)
+ +
(c) f ( t  1) 29(t  1) 3y(t  1) = xlyt  3) 2x(t  4) +
+
816 The system function of a given LTI system is H(s) = l/(s l), G >  1.
(a) Determine a differential equation relating to the system input, x(t), and
the system output, y(t).
+ + +
(b) Note that H(s) = (s 2)/(s l)(s 2), G > 1 is the same system
function because the pole and zero at s = 2 cancel. However, the
differential obtained is different. Obtain the differential equation relating
the system input, x(t), and the system output, y(t).
(c) Show that we obtain the differential equation in part b by adding twice
the differential equation in part a to its derivative. To obtain differential
equation of the lowest possible order, we should always reduce H(s) by
canceling all common factors before determining a differential equation
relating to the system input, x(t), and the system output, y(t).
819 A simple pendulum with linear damping can be modeled by the differential
equation
(a) What is the maximum value of the angle, $, for which sin$(t) x $(t)
with an error less than I%? Less than 5%?
Use the approximation to determine the system response, $(t),to the
input torquef(t) = A r ( t / T ) :
(b) For the case in which a = 2 and b = 5.
(c) For the case in which a = 2 and b = 1 .
820 The system function of the echoing system discussed in Section 1.6 is
H ( s ) = 1/( 1  Ke"9 with the RAC to the right of the most righthand pole.
We first determine the location of the poles. they are at those values of s for
which K ~ C '= . ~1.
(a) To solve this equation, we have e's = eT(u+Jo)= erueJrw= 1 / K .
Show that the solutions of this equation are
s = In
1 IKI + j  ifK>O
z
M M 1
H ( s ) = C(Ke")" = Kfleflrs, 0 > I l n IKI
fl=O fl=O z
82 1 The unitimpulse response of an LTI system is h(t) = e"'u(t), a > 0. The
system inputs is x(t) = cos(wot)u(t). As discussed in Section 8.5, since the
system is stable, the system response approaches the steadystate response as
t increases.
(a) Determine the system response, y(t).
(b) Show that y(t) approaches the steadystate response as t increases.
822 The differential equation relating the input, x ( t ) , and output, y(t), of a given
circuit is
(a) Use Eq. (8.511) to determine the system response to the input
~ ( t=) A ~0~(2t)u(t).
(b) Show that the steadystate response is equal to the system sinusoidal
response.
(c) From your solution in part a, determine the initial conditions for which
there is no transient response so that y(t) is equal to the steadystate
response.
823 The differential equation relating the input, x(t), and output, y(t), of a given
+ + +
circuit is y”(t) ay’(t) by(t) = cx’(t) dx(t) in which the constants are
determined by the values of the circuit parameters. The initial values of the
circuit voltages and currents result in the initial conditions being y’(O+) and
Y(O+).
(a) Determine the initial conditions required for there to be no transient.
(b) For a given set of initial conditions, determine the iqput, Xb(t), for which
y(t) = 0 for t > 0.
(c) We would like the system response to be yc(t)= h(t) * xc(t) in which h(t)
is the system unitimpulse response. That is, yc(t) is the system zero
initial condition response. Show that the required system input is
+
x(t) Xb(t) x,(t).
SPLANE VIEW OF GA N AND PHASE
SHIFT
Some uses of the Laplace transform in LTI system analysis were illustrated in the last
chapter. Another important use of the Laplace transform which we discuss in this
chapter is the analysis of the gain and phase shift of stable, lumped parameter LTI
systems directly in the s plane. The importance of this analysis technique is that it
results in a physical view of the relation of the system poles and zeros to the system
gain and phase shift which is one of the bases of filter design.
We showed in Section 8.3 that the system function of a stable lumped parameter
LTI system has the form
in which, from our discussion of stability in Section 8.2, m 5 n and the RAC
includes the o axis. Because the w axis lies in the RAC, we can let s =j w in Eq.
(9.11) to obtain the transfer function
(9.12)
(9.13a)
265
266 SPLANE VIEW OF GAIN AND PHASE SHIFT
The constant la,/b,,l in Eq. (9.13a) is called the gain constant because it is a
constant that affects the size but not the shape of the graph of the gain versus
frequency, and La,/b, in Eq. (9.13b) is called the phase constant because it is
just an additive constant that does not affect the shape of the graph of the phase shift
versus frequency. As discussed in Section 8.3, the coefficients a, and b, are real
numbers because they are the coefficients of the differential equation relating the
system input and output. Thus the ratio am/bnis a real number so that the phase
constant is zero if the ratio is a positive real number and .n radians (180") if it is a
negative real number. Other than the gain and phase constants, the determination of
the gain and the phase shift from Eqs. (9.13) requires the determination of the
magnitude and angle of terms of the form (jo= so) in which so = z k , or so = P k
in Eq. (9.12). Although these quantities can be determined algebraically, we shall
develop a geometrical splane interpretation of them which will enable us to obtain
insight into how the system gain and phase shift is affected by the system function
poles and zeros.
To develop the geometrical interpretation, consider the splane diagsm shown in
Fig. 9.1la. The vector 2 is from the point so to the origin, the vector b is from the
origin to the point coo on the o axis, and the vector f is from the Bint so to the
+
point ooon the o axis. The relation between the three vectors, a , b , and 2,is
c =+b +
+ a
+ (9.14a)
Now note that the algebraic expressions for the vectors are
+ +
a = so and b =jwo (9.14b)
..........
0
io0
b
so
CI cr
(a> (b)
Fig. 9.11
9.1 GEOMETRIC VIEW OF GAIN AND PHASE SHIFT 267
As shown in Fig. 9.11b, we observe that the length of the vector f is equal to the
magnitude of (jo, so) and the angle of the vector f from the positive cr axis is
equal to the angle of ( j w ,  so).That is, the magnitude of ( j w  so) is equal to the
distance from the point so to the point wo on the w axis and the angle of ( j w  so)is
equal to the angle from the positive real axis to the line from the point so to the point
wo on the w axis.
In terms of this geometric interpretation, the system gain given by Eq. (9.13a)
and the system phase shift given by Eq. (9.13b) can be stated as follows:
1. The system gain at the frequency wo is equal to the gain constant times the
product of the distances from the system function zeros to the point wo on the
w axis divided by the product of the distances from the system function poles
to the point wo on the w axis.
2. The system phase shift at the frequency wo is equal to the phase constant plus
the sum of the angles from the system function zeros to the point wo on the w
axis minus the sum of the angles from the system function poles to the point
wo on the w axis.
We shall illustrate this geometric view by analyzing some basic filter types.
for the case in which a > 0 and b > 0. To discuss gain and phase shift, the system
must be stable. Thus we require b > 0 for the w axis to lie in the RAC. Figure 9.12a
is the splane diagram of this system function which has no zeros and only one pole
at s = b. From Eq. (9.13a), the gain of this system at the frequency wo is
(9.16a)
Now, from our discussion above, lp = Ijw, + 61 = ,/ so that the gain at the
frequency wo also can be expressed as
(9.16b)
268 SPLANE VIEW OF GAIN AND PHASE SHIFT
Fig. 9.12a
0.2 
0.1 
00
OO i 6 i s 6 7 i
e 'Or
Fig. 9.12b Graph of the system gain.
70 
4Q
90 00
0 1 2 3 4 5 6 7 8
' O T
Fig. 9.12c Graph of the system phase shift.
That is, the system gain at the frequency oois equal to the gain constant divided by
the distance from the pole to the point ooon the w axis. This distance, ,Ip, increases
as wo increases so that the system gain decreases as the frequency, oo,increases. In
accordance with our discussion in Section 4.2, this means that the system amplifica
tion of a sinusoid decreases as the frequency of the sinusoid increases. Thus the
system is seen to be a lowpass filter with the graph of system gain versus frequency
as shown in Fig. 9.12b.
The dc gain is lal/b because, from Fig. 9.12a, lp = b for wo = 0. As discussed in
Section 4.3, the width of the passband region is usually specified as the distance to
the frequency at which the gain is l / f i of its maximum. This is called the half
power bandwidth or, equivalently, the 3dB bandwidth of the lowpass filter. Because
the maximum gain is lal/b at which lp = b, the gain is l / f i of its maximum value
at the frequency for which lp = b f i . With the use of the Pythagorean theorem, the
hypotenuse of the right angle triangle in Fig. 9.12a is b& when the length of the
triangle vertical leg is b. Thus the 3dB bandwidth of the lowpass filter is b because
9.1 GEOMETRIC VIEW OF GAIN AND PHASE SHIFT 269
the length of the vertical leg is wo.Now note that for wo >> b, the hypotenuse length,
lp, is approximately equal to the length of the triangle vertical leg, wo. Thus for
wo >> b, an approximate expression for the gain is lal/wo.
Now, from Eq. (9.13b), the system phase shift at the frequency wo is
L H,(jwo) = L u  +
L( j ~ o6) (9.1 7a)
L H,(jw,) = i a  8 (9.17b)
where 8 is the angle from the pole at s = b to the point coo on the w axis as shown
in Fig. 9.12a. Because a is a real number, we have
La= { 0
n rad
ifa>O
if a < 0
(9.1 7c)
For our example, a > 0 so that La = 0. Thus the phase shift for the given system is
(9.17d)
Figure 9.12c is a graph of the system phase shift versus frequency for the given
system. Observe that the system phase shift is n/4 rad (or 45") at wo = b
because both legs of the rightangle triangle in Fig. 9.12a are equal for wo = b.
A good approximation for the phase shift at low frequencies can be obtained by
using the approximation tan 8 M 8 if 8 is small. Thus, in Fig. 9.12a, for small values
o f 0 we have
w0
8 M tan8 =  (9.1 7e)
b
(9.17f)
This is the equation of a straight line with the slope equal to  1/b, as can be seen in
Fig. 9.12c. We also can obtain an approximation for the system phase shift at high
frequencies. For this, we note from Fig. 9.12a that 0 = 7112  4. For 8 x 7112, $ is
small so that $ x tan 4 = b/wo. Thus an approximation for the system phase shift
for <o0>> b is
71 n b
iH,(jwo)=e=+$x+ (9.17g)
2 2 0 0
270 SPLANE VIEW OF GAIN AND PHASE SHIFT
Thus, for high frequencies, the distance in Fig. 9.12c from the phaseshift graph to
n/2 is approximately equal to b/wo.
as
H&) =  CT > b (9.18)
s+b'
for the case in which a > 0 and b > 0. Figure 9.13a is the splane diagram of this
system function which has one zero at s = 0 and one pole at s = b.
I"
Fig. 9.13a
20
10
0
  
00
1 2 3 4 6 6 7 0
= l o b
Fig. 9 . 1 3 ~ Graph of the system phase shift.
9.1 GEOMETRIC VIEW OF GAIN AND PHASE SHIFT 271
IHb(jwO)l =
4 (9.18a)
lP
which is equal to the gain constant, la[,times I,, which is the distance from the zero
at s = 0 to ooon the o axis, divided by lp, which is the distance from the pdle at
s = b to w,, on the o axis so that
(9.18b)
With the use of Fig. 9.13a, note that Eq. (9.18a) also can be expressed as
As ooincreases, 6 increases from 0 to 7112 rad so that sin 6 increases from zero to
one. Thus a graph of the system gain versus frequency is as shown in Fig. 9.13b.
The system is seen to be a highpass filter whose gain is zero at oo= 0; as o
increases, the system gain increases monotonically and approaches la1 asymptoti
cally. The frequency at which the gain is l / d the maximum gain is the frequency
for which 6 = 7114 rad at which 1, = b. Thus the frequency at which the gain is
I / & its maximum value (the halfpower frequency) is w,, = b.
A lowfrequency approximation of the gain can be obtained by noting that, for
small values of 8, sin 6 M tan 6 so that
This is the equation of a straight line with the slope lal/b as seen in Fig. 9.13b.
Now, from Eq. (9.13b), the system phase shift at the frequency is
LHh(jw0) = La + L(jw0)  L ( j 0 0 + b)
n (9.18e)
= i a + 2 e
Geometrically, the phase shift is equal to the phase constant, La,plus 7112 which is
the angle from the zero to the point o,,on the o axis, minus the angle from the pole
at s = b to the point ooon the o axis. Figure 9.13c is a graph of the phase shift
versus frequency.
By comparing Eq. (9.17a) with Eq. (9.18e), we note that
71
LHb(jO,,) = 
2 + LH,(jw,) (9.18f)
so that the phaseshift graph of the onepole highpass filter is that of the onepole
lowpass filter moved up by n/2 rad. This is due to the zero at s = 0.
272 SPLANE VIEW OF GAIN AND PHASE SHIFT
We shall use our geometrical understanding of gain and phase shift to examine
some different basic filter types in the following sections.
The polezero pattern of the highpass filter discussed in the last section is a special
case of a polezero pair. The analysis of the gain and the phase shift associated with a
polezero pair in which both the pole and the zero are on the o axis is important
because a number of different filter types can be obtained with different configura
tions of the polezero pair. Also, the analysis of the gain and the phase shift asso
ciated with a polezero pair also well illustrates the geometric view and some of the
useful analysis techniques that can be used to obtain numerical results. Conse
quently, we shall analyze this case in detail and then consider some extensions of
our results to more general polezero patterns.
(9.21)
where CI > /3 > 0. Figure 9.2la is the splane diagram of this system function which
has one pole at s = a and one zero at s = /3.
Fig. 9.2la
: I OO
In accordance with our discussion in Section 9.1, the gain of the given system at
the frequency oois
(9.22)
Note from the figure that I,, the distance from the zero to the point ooon the o axis,
is less than lp, the distance from the pole. The two distances become equal asymp
totically as the frequency wo increases. From this simple observation, we can imme
diately conclude that the gain rises monotonically from a value of p / a at o = 0 and
is asymptotic to a value of one as o + 00 so that the gain curve must have the shape
shown in Fig. 9.2lb.
To determine an expression for the gain curve, we have from Fig. 9.2 1a with the
use of the Pythagorean theorem that
1; = o i + p2 and +
1; = o; a2 (9.23a)
(9.23b)
(9.24)
(9.25)
Table 9.21 is a short table of the halfpower frequencies for some values of the ratio
PIE.
Another frequency of interest in our analysis is the frequency at which the gain is
equal to &(p/ a) (the doublepower frequency). The doublepower frequency can be
274 SPLANE VIEW OF GAIN AND PHASE SHIFT
determined in the same manner we used to determine the halfpower frequency. For
this, we have from Eq. (9.23b), that, at the doublepower frequency,
(9.26)
B
wo =
/qJ (9.27)
The doublepower frequencies for some values of ...e ratio p / a also are listed in
Table 9.21.
We now determine the system phase shift. From our discussion in Section 9.1, we
have that the phase shift, 4, is
4 = e,  e, (9.28a)
0 1 2 3 1 5 6 7 e 9 10 
Q)O
a
Fig. 9.22 Graph of the system phaseshift for p = f a .
0 0 00
tan(9,) =  and tan(9,) =  (9.28b)
B a
(9.28~)
Because tan0 zx 0 for small values of 9, we have from Eq. (9.28c) that for small
values of oO
(9.28d)
so that near zero frequency, the system phaseshift curve is a straight line with the
slope ( a  B)/a/l. The maximum of the system phaseshift curve is at the frequency
for which the derivative of Eq. (9.28c) is zero. Making use of the derivative formula
d ' 1 dx
tan x= (9.28e)
d9 1+x2d0
we obtain that system phaseshift curve is a maximum at the frequency wO= ,/$.
From Eq. (9.28c) the system phase shift at this frequency is
The maximum phase shift and the corresponding frequencies for some values of the
ratio B / a are as listed in Table 9.22.
276 SPLANE VIEW OF GAIN AND PHASE SHIFT
0 90 0
0.1 54.9 0.32~
0.2 41.81 0.45~
0.3 32.58 0.55~
0.4 25.38 0.63~
0.45 22.29 0.67~
0.5 19.47 0.7 1u
0.55 16.88 0.74~
0.6 14.48 0.77~
0.65 12.25 0.81~
0.7 10.16 0.84~
1142 9.88 0.84~
(9.29)
where a > p > 0. Figure 9.23a is the splane diagram of this system function which
has one zero at B = a and one pole at B = p. Observe that H&) = l/Hu(s).
Consequently, the system gain is
Because the gain of system B is the reciprocal of the gain of system A , the gain curve
must be as shown in Fig. 9.23b, which is the reciprocal of the gain curve of
Fig. 9.2lb.
Furthermore, the system phase shift is
Because the phase shift of system B is the negative of the phase shift of system A , the
phaseshift curve must be as shown in Fig. 9.23c, which his the negative of the
phaseshift curve of Fig. 9.22.
9.2 THE POLEZERO PAIR 277
1 
0 6 
1 2 3 4

a
(9.21 1)
where a > 0 and b > 0. Figure 9.24a is the splane diagram of this system hnction
which has one zero at (T = b and one pole at (T = a.
To determine the system gain curve, we first consider the case in which a > b.
Note for this case that, for b = fi and a = a, the lengths 1, and lp in Fig. 9.24a are
the same as that in Fig. 9.2la. Thus the system gain, which is IHc(joo)l = I Z / I p , is
the same as that for system A , which is a highpass filter.
278 SPLANE VIEW OF GAIN AND PHASE SHlFl
I" "0
A.
a b" 0
rea
8 
Now consider the case in which b > a. Compare the splane diagram for this
case, Fig. 9.24a, with the splane diagram for system B, Fig. 9.23a. Note that for
a = j?and b = a, the lengths 1, and Ip in the two figures are the same, so that the gain
curve of system C for this case is the same as that of system B, which is a lowpass
filter.
Finally, consider the case in which a = b. For this case, we note from Fig. 9.24a
that I, = lp for all values of frequency. Thus, for this case the gain is
2, = 1
IH,(jco,)I=  (9.212)
lp
A filter with a gain curve that is a constant is called an allpass$lter. Allpass filters
are important because, as we shall discuss later, they can be used to obtain a desired
phaseshift curve without affecting the gain curve.
Now, the phase shift of system C at the frequency oois
LH,(joo) = 0,  ep (9.213a)
To simplify tracking the difference of two angles as the frequency, coo, is varied, we
+
note from Fig. 9.24a that 0, = OP $. With this relation, the system phase shift can
be expressed in terms of only one angle as
(9.2 13b)
The shape of the system phaseshift curve is seen to be as shown in Fig. 9.24b.
9.3 MINIMUMPHASE SYSTEM FUNCTIONS 279
j = tan'
t + (&)
(3) tan'
= tan' (8)
+ (8) tan' = ?rad
2
(9.213~)
For our discussion in this section, CI and B are positive real numbers. We saw in the
last section that the gain curve of a system for which
Ha@)= 
sB (T>U (9.3 1a)
S t C I '
(9.3 1b)
Figures 9.3la and 9.3lb are the splane diagrams of Ha(s)and Hb(s),respectively.
The gain curve of both systems is the same because, for both systems, 1, and Zp
are the same for any value of w,,. The two systems phaseshift curves, however, are
I"
Fig. 9.3la
a
A, P
"O
Fig. 9.3lb
280 SPLANE VIEW OF GAIN AND PHASE SHIFT
different. From Figs. 9.3la and 9.3lb, the phase shift of the two systems, A and B,
are
Thus the different between the phase shifts of the two systems is
Note from the figures that $2 + 0, = x. With this relation, we can express Eq. (9.3
3a) as
From Fig. 9.3lb, we see that 6, < 7112 for any frequency, oo.Consequently, we
observe from Eq. (9.33b) that, for any frequency, oo,
Thus the phase shift of system B is always less than that of system A. Again, note
that system B is obtained from system A simply by moving its zero in the right half
of the s plane at s = 8 to its mirror image in the left half of the s plane at s = B.
Now consider a causal system with two poles and two zeros whose splane
diagram is shown in Fig. 9.32a. From the diagram, the system function is
(9.35)
The magnitude of the real constant A is the gain constant, and the angle of the real
constant A (which is 0 if A > 0 or x if A < 0) is the phase constant. The RAC is
CJ > a2 because the system is causal. The system is seen to be stable because the o
axis is contained within the RAC. Observe that the system function can be deter
mined within the constant, A , from its splane diagram. The gain and phase shift of
this system can be determined from our previous work by noting that it can be
expressed as
where
+82
H,(s) =  and H2(s)= s (9.36b)
s + a,
+
s + a2
Thus the gain of the given system is equal to IA I times the product of the gains of the
two systems given in Eq. (9.36b), and the phase shift of the given system is equal to
LA plus the sum of the phase shifts of the two defined systems. Note, too, that given
9.3 MINIMUMPHASE SYSTEM FUNCTIONS 281
P1
,. ."
1 2

a
02
 "I I
a
Fig. 9.32a

" " I
A
"I I "
p1 a1 2 I P2
Q
Fig. 9.32b
Fig. 9.32c
* A "
1 2 P2 K Q
where Hc(s)is the minimumphase system function given by Eq. (9.35) and Ho(s)is
the allpass system function
s82,
Ho(s) =  o> p2 (9.37b)
s+82
We note from this result that an LTI system with a nonminimumphase system
function can be expressed as the tandem connection of a minimumphase system
and an allpass system. From Eq. (9.212), the gain of the allpass system is one at
all frequencies. Thus, from Eq. (9.37a), the system gain is
because IHo(jo)l = 1. We thus observe that the allpass system does not affect the
gain of the tandem connection. However, the allpass system does affect the system
phase shift of the tandem connection. The phase shift of the tandem connection is
equal to the sum of the phase shift of the minimumphase system and the phase shift
of the allpass system
in which the phase shift of the allpass system above, LHo(jw), is from
Eq. (9.213b).
(9.38~)
9.3 MINIMUMPHASE SYSTEM FUNCTIONS 283
$o(o) =  o r 0 (9.39)
then its output would be y (t ) = x(t  to),so that the response of the allpass system
would be its input delayed by to seconds without distortion. In Eq. (9.39), the graph
of &(w) versus w is a straight line passing through the origin with a slope equal to
(9.3 10)
so that the slope of the phaseshift curve is equal to the time shift. Note that a
negative slope is a delay and a positive slope is an advance of x(t). However, it
was shown near the end of Section 8.3 that an ideal delay system cannot be realized
by a lumped parameter system, so that the phase shift of an allpass system with a
finite number of poles and zeros can, at best, only approximate Eq. (9.39) over some
frequency range.
Observe from our analysis in Section 9.2C that the phase shift of a lumped
parameter allpass system decreases with frequency, so that the derivative of the
allpass system phase shift with respect to o is negative. Thus we would expect the
allpass output, y ( t ) , to be a delay of some distorted version of its input, x(t).
Therefore we expect the energy of x ( t ) to be delayed. To study this, we define the
partial energy of the system output, y(t), to be
(9.31 1)
284 SPLANE VIEW OF GAIN AND PHASE SHIFT
The total energy of a waveform was discussed in Section 5.9, where the total energy
of a waveform, y(t), was defined to be
(9.3 12 )
Thus the partial energy, Ey(T),is seen to be the energy ofy(t) up to the time t = T .
Because the gain of any allpass system is one, the energy density spectrum of its
output, IY(jo)I2,is equal to that of its input, IX(jo)12,so that E, = Ey. That is, the
total energy of x(t), E,., is equal to E,,, the total energy of y(t).
The effect of the position of a zero of H(s) on the output of an LTI system is
derived and analysed in Appendix B. One result shown there is that, for any value of
T , the partial energy of the output of an allpass system, Ey(T),is less than the partial
energy of its input, E,.(T), so that we indeed can say that the output of an allpass
system is a distorted version of a delay of its input.
To illustrate this result, let the input of an allpass system be
for which
1
X(s)= ~ CJ > a (9.313b)
s+a'
sb
H(s)= C T >  ~ ,b > O (9.3 14)
s+b'
sb
Y(s) = H(s)X(s)= CT > ma(a, b) (9.315a)
(s + a)(s + b ) '
Using the partial fraction expansion discussed in Section 7.3, we obtain for a # b
1
y(t) =  [ ( a
ab
+ b)e"'  2behf]u(t) (9.3 15b)
(9.3 16 )
9.3 MINIMUMPHASE SYSTEM FUNCTIONS 285
t
Fig. 9.33 Graph of x ( t ) and y(t) for a = 2 and b = 3.
(9.317)
Thus we observe for our example that E J T ) > Ey(T)because the difference in Eq.
(9.318) is always positive, so that the energy of x ( t ) has indeed been delayed. Note
that the difference in the partial energies goes to zero as T + 00. The reason is that
as T + 00, the partial energy becomes equal to the total energy and, as discussed
following Eq. (9.312), that the total energy of x(t) and y(t) are equal.
In summary, we have shown that any stable and causal LTI system can be
expressed as a minimumphase system connected in tandem with an allpass
system. Even though the gain of the allpass system is one at all frequencies, its
output is a distorted version of its input due to the allpass system phase shift. The
total energy of the allpass system output waveform is the same as that of its input
because the system gain is one at all frequencies. However, the effect of the phase
shift is to delay the partial energy of its input waveform.
286 SPLANE VIEW OF GAIN AND PHASE SHIFT
For all the cases we analyzed in previous sections, the system h c t i o n poles and
zeros are located on the 0 axis. In this section, we extend our analysis to system
functions that contain poles and/or zeros off the 0 axis. To begin, consider the stable
LTI system with the unitimpulse response
The system fimction is the Laplace transform of h(t), which is, from pair 4b of Table
7.41,
1
H(s) = A > ci (9.42)
(s + a)2 + 0;'
0
To simplify our analysis of this system's gain and phase shift, we shall make some
approximations that require w , >> ci. As we perform our analysis, we shall determine
exactly how much larger w , must be relative to M in order that each approximation
made is valid.
Approximations are important in practical analysis. The are often made in analyz
ing problems in which the exact results are complicated and not easy to visualize.
The procedure often used is to make all approximations needed to obtain a reason
able analysis of the problem. After the analysis is complete, all the approximations
made are analyzed to determine the conditions required for all the approximations to
be valid. Computers are excellent for obtaining accurate numerical results. However,
an understanding of the general effects of various parameters, which is of basic
importance in design, is readily obtained from approximate theoretical analysis as
illustrated in this section. This is why engineering has often been referred to as a
science of intelligent approximations.
The splane diagram of the system function, H(s), is shown in Fig. 9.41. As
shown, there is a pole at s = a + j w , , a pole at s = a  j w , , and no zeros. The
system gain at the frequency coo is equal to
(9.43)
in which, as shown on Fig. 9.41, lp, and lp2 are the distances from the upper and
lower poles, respectively, to the point wo on the w axis. First, the distance lp2 is seen
to be the length of the hypotenuse of a rightangle triangle with a horizontal leg of
+
length ci and a vertical leg of length w , coo. Because a << w , , the vertical leg is
much longer than the horizontal leg so that the hypotenuse length is approximately
+
equal to the length of the vertical leg. Thus our approximation is lp2 wl wo for
all positive frequencies, wo. The worst case for this approximation is for wo = 0, in
which case we require lP2= d m x w , for our approximation to be acceptable.
9.4 BANDPASS SYSTEM FUNCTIONS 287
I"
1
T"
Fig. 9.41
(9.44)
Now observe that lp, is the length of the hypotenuse of a rightangle triangle with
a horizontal leg of length a and a vertial leg of length lo1 ooI. To approximate lpl,
we first consider small values of the frequency coo for which the vertical leg is much
longer than the horizontal leg, so that the hypotenuse length is approximately equal
to the length of the vertical leg. Thus our approximation for small values of coo is
lp, x o1 wo. Similar to our approximation for Ip2, this approximation error is less
than 5% if ( w ,  oo)2 3.13a or, equivalently, if ooI (0,  3.13~).With this
approximation, the gain for small values of coo from Eq. (9.44) is
(9.46)
288 SPLANE VIEW OF GAIN AND PHASE SHIFT
Note that the system gain is inversely proportional to lp, in this frequency range.
From this expression, we observe that the gain curve has a maximum at wo = w 1
where ZpI = a, its smallest value. Thus an approximate expression for the maximum
gain is IAl/2wla.
+
For wo 2 w1 3.13a, we can approximate lp, with an error less than 5% as
lp, x wo  wl. Thus, from Eq. (9.44), our approximation in this range is
A good approximation of the exact gain curve shown in Fig. 9.42 is obtained with
the approximations (9.45), (9.46), and (9.47) in their proper frequency ranges. For
all the approximations made to be valid with an error no greater than 5%, note that
we require w , > 30a. However, note from the figure that the approximations are
very good for w1 = 151%.
The gain curve is seen to be that of a bandpass filter with a maximum gain of
JA1(1/2awl)at wo = wl. The halfpower frequencies are the frequencies at which
the gain is 1/2 of its maximum. Similar to our discussion in Section 9.1, we have
from Eq. (9.46) that the halfpower frequencies are those for which Zpl = &a.
From Fig. 9.41, Zpl = &a at oo= w1 f a . Thus the halfpower bandwidth is
2a. Although our approximations require w1 > 30a, we can observe from Fig.
9.42 that this is a very good approximation even for w1 as low as 15a.
An important parameter characterizing a bandpass filter is its quality factor, Q,
which, in filter theory, is defined as
0.9 
0.0 
0.7 
0.6 
0.5 
0.4 
0.3 
0.2  0 = 7.5
0.1  Q=15
1
1 W
Fig. 9.42 Exact gain curves for Q =  2 = 15 and 7.5.
2 u
9.4 BANDPASS SYSTEM FUNCTIONS 289
Note that 1/ Q is the fraction the bandwidth is of the bandpass center frequency. Thus
the Q of our bandpass filter is
Q = W l
(9.49)
2ci
Note that our approximations about the passband region require o,2 30a or,
equivalently, a Q greater or equal to 15 for which the bandwidth, 2ci, is 6.7% of
the center frequency, coo. The Q of bandpass filters often are greater than this. In fact,
the Q of microwave filters often exceed lo4 and sometimes exceed lo6. To realate
the bandpass parameters to the s plane, observe that, for our approximations, the
center frequency is the distance a pole is from the 0 axis and the bandwidth is equal
to twice the distance a pole is from the o axis. In the time domain, we note from the
expression of the unitimpulse response, Eq. (9.4l), that the center frequency is
equal to the frequency of the sinusoid and the bandwidth is equal to twice the
exponential decay rate.
From Fig. 9.41, the phase shift, 0, of the bandpass filter is seen to be
(9.4 10)
0 = tan' ( y )mo
WI 
90" (9.41 I )
with an error less than 2". The phaseshift curve is shown in Fig. 9.43.
The gain and the phase shift in the filter bandpass region is the frequency range of
+
major concern. In the bandpass region, w ,  3a 5 ioo 5 o1 3a, the gain is given
by Eq. (9.46) and the phase shift is given by Eq. (9.411). Note that the pole at
s = ci j w , only contributed a constant to these equations; the shape of the gain
0
15 
Phase shift, 8 a
(degrees) 45 
80
75 
4) 
1s 
1m 
135 
150 
1s 
180 A a0
Fig. 9.44
9.5 ALGEBRAIC DETERMINATION OF THE SYSTEM FUNCTION 291
x 001
Fig. 9.45
The importance of the discussion above is that one can design a filter for a desired
shape of the gain and phaseshift curves centered about a frequency w = o1by first
designing a filter with the desired shape of the gain and phaseshift curves about
o = 0. A filter with approximately the same shape of the gain and phaseshift curves
centered about o,then can be obtained by moving the poles and zeros vertically up
an amount w I and also down by the same amount (for the required conjugate pairs).
This basic concept is often used in filter design. The lowpass Buttenvorth filter will
be analyzed in the next section. We then will use the lowpass to bandpass trans
formation concepts discussed above to design a bandpass Buttenvorth filter.
The geometric analysis technique discussed in the previous sections of this chapter
lends great insight into the relation between the gain and phase shift of a stable LTI
system and the locations of its system function poles and zeros. With this insight,
one often can determine an approximate splane polezero pattern of the system
function required for a desired system gain and phase shift. A computer then can be
used to determine the required pole and zero locations with greater precision.
Another technique used to determine the required system function is an algebraic
one. We’ll discuss this technique because of the insight it lends to splane analysis.
Rather than beginning with a general discussion of the technique, we begin by
discussing its use in the design of the lowpass Buttenvorth filter.
(9.51)
292 SPLANE VIEW OF GAIN AND PHASE SHIFT
is called an nthorder Butterworth filter. Figure 9.51 is a graph of the gain for some
values of n.
Observe that the gain curve is that of a lowpass filter for which, for any value of
n, the halfpower frequency is o,and which approaches the ideal rectangular gain
curve monotonically for increasing values of n. In accordance with our discussion of
the PaleyWiener criterion in Section 5.1 lB, the ideal rectangular gain curve can not
be achieved by a physical system. However, as we discussed, the gain curve of a
physical system can closely approximate the ideal rectangular curve. The Butter
worth filter is one such approximation.
To design a filter with the Butterworth gain curve, we need to determine the
required system function, H,(s). For this, we first eliminate the square root in Eq.
(9.51) by squaring the gain expression
(9.52)
Note that because h(t) is a real function oft, we have with the use of our result, Eq.
(5.55b),
0.7 
0.6 
0.6 
0.4 
0.3 
0.2 
0.1 
To determine the RAC, we have from our discussion of the interval property of the
RAC in Section 6.2 that the RAC of a finctionf(t) is an interval between the poles
of its transform, F(s). Now, Eq. (9.53b) requires the o axis to be included in the
RAC of H(s)H(s). Thus the RAC for Eq. (9.54) must be the interval oU < o < o b
+
in which s = ou j w is the vertical line on which the first pole to the leR of the w
axis lies and s = ob +jo is the vertical line on which the first pole to the right of the
o axis lies.
Now, the expression in Eq. (9.54) has no zeros, but it does have poles at those
values of s = sk for which
2n
=o (9.55a)
With the use of Appendix A, the solution of this 2nthdegree polynomial equation
for the 2n values of s which satisfy it is obtained as follows:
2n
&1+2k)n, k = 0, f l , f 2 , f 3 , . . . (9.55b)
where
(9.55c)
/!sk = (2
1 1+2k
+ 2n )TC rad = (z + y)
1 1+2k
180" (9.57)
Thus the solutions lie on a circle with its center at the origin and radius equal to o,
in the s plane. The pole positions for n = 2 and 3 are shown in Fig. 9.52.
Observe from Eq. (9.57) that the angle of sk is increased by an amount of 271
radians if k is increased by an amount of 2n. Thus, for any value of ko, the value ofsk
is the same for k = ko and for k = ko + 2n. Consequently, there are exactly 2n
294 SPLANE VIEW OF GAIN AND PHASE SHIFT
(9.58a)
and
(9.58b)
Example To illustrate the use of the results we've obtained, we design a lowpass
Butterworth filter with the specfication that the dc gain be 10 and not drop below 9
before a frequency of 12 kHz.In the reject band, it is specified that the gain be below
1.5 above a frequency of 3OkHz. Note from Fig. 9.51 that we can satisfy or exceed
these specifications by designing the filter such that
IH,(j2400071)1 = 9 (9.59b)
9.5 ALGEBRAIC DETERMINATION OF THE SYSTEM FUNCTION 295
and
We have three unknowns A , o,,and n in Eq. (9.51) that can be specified to satisfy
the three equations of Eq. (9.59). First, it is clear that Eq. (9.59a) is satisfied with
A = 10. We now need to determine the values of o, and n in Eq. (9.51) such that
the other two equations of (9.59) are satisfied. For this, we have from Eq. (9.52)
that
(9.5 1Oa)
The righthand side of this equation is a function only of the system gain. To
simplify our manipulations, call this function of the gain T(w).
(9.5lob)
(9.5 1OC)
In Eqs. (9.59) the gain is specified at two frequencies, o1and w2. Thus we can
eliminate o, from Eq. (9.51Oc) to obtain
(9.5 1Od)
so that, by taking the logarithm of this equation and solving for n, we obtain
(9.51 la)
We then can use the calculated value of n to determine o,from Eq. (9.51Oc):
or
o,= ol/rl/2n(ol) W, = 02/rl j 2 n (a2) (9.51 lb)
For our example, using A = 10 and the values from Eqs. (9.59) the calculated
value of n is from Eq. (9.51 la),
1 ln(=) 15.2214
 ~
it=  = 2.8492 (9.512a)
In(%) 2 0.9 163
24,00071 
24,00071 24,00071
0,=  = 30,95671 rad/s (9.512b)
(0.2346)  0.7753
(o.2346)0.1755
which is 15,478 Hz. We would obtain the required specifications with these values of
w, and n. However, the order of the filter, n, must be an integer. From Fig. 9.51, we
observe that the specification will be exceeded by increasing the value of n. Thus we
choose n = 3, which is the first integer greater than the calculated value. Therefore
designed filter is a thirdorder lowpass Butterworth with A = 10 and
w, = 15,478 Hz = 30,956nrad/s. We now determine the pole locations because
they are the solutions of Eq. (9.57) that lie in the left half of the s plane.
A
H3 (4 , CJ> 0.50, (9.515)
(:)3+2(;)2+2(;) +1
By partial fraction expansion of Eq. (9.514), the unit impulse response of a third
order Butterworth filter can be shown to be
9.5 ALGEBRAIC DETERMINATION OF THE SYSTEM FUNCTION 297
Using the time differentiation property (item 6 of Table 7.42), we obtain the inverse
Laplace transform of this equation, which is the differential equation relating the
system output, y(t), and system input, ~ ( t ) :
(9.5 18)
and
(9.519~)
The tandem connection of these two systems is the desired thirdorder lowpass
Butterworth.
A third synthesis of this filter can be obtained by noting that H3(s)also can be
expressed as
in which
H,(S) = ~
Am, , (T>o, (9.520b)
s  s,
298 SPLANE VIEW OF GAIN AND PHASE SHIFT
and
(9.520~)
Although the shape of the bandpass filter about o = o1is the same as that of the
lowpass filter because o1>> o, (for our example, wI = 29.40,), we must deter
mine A so that the gain of the bandpass filter at w1 is the same as the lowpass filter at
o = 0. It is the poles at s = sl, s3,and s5 that determine the shape of the gain curve
in the bandpass region. In this region, the distances from the poles at s = s2, s4, and
s 6 are approximately constant and equal to 20,. Thus, for the bandpass filter gain to
have the same magnitude at O , as the lowpass filter at zero frequency, we must
make A = l O ( 2 ~ , ) The
~ . factor of (20,)~cancels the approximately constant factor
of l/(20,)~ contributed by the three poles at s = s2, s4, and s6. With this choice of
A , the gain curve of the bandpass filter with the system function given by Eq. (9.5
21) has the same shape and magnitude in its bandpass region as the lowpass filter in
its passband.
PROBLEMS 299
The technique for the design of a Butterworth filter can be generalized. Consider
again Eq. (9.52), which is the basic equation specifying the gain curve of the low
pass Butterworth filter. The expression there is of the form
(9.522)
where PJo) is an nthdegree polynomial that is chosen for the filter to have the
desired gain curve. For the nthorder lowpass Butterworth filter, the polynomial is
P,(w) = w". Generally, for a lowpass filter in which the gain is approximately
constant in the passband and then decreases rapidly outside the passband and in
which the dc gain is equal to A , we require the polynomial, P,(w), to be equal to zero
at o = 0, be small for w in the passband, and then increase rapidly outside the
passband. The choice of P,(o)= on for the lowpass Butterworth filter is the
simplest polynomial that can be chosen. There are many others that can be
chosen. ORen, the name of the filter is the same as the name of the polynomial
used. For example, the nthorder lowpass Chebyshev filter is obtained by using the
Chebyshev polynomial, T,(o), and the nthorder lowpass elliptic filter is obtained
by using the Jacobean elliptic polynomial, U,(o). Each of these polynomials results
in a different approximation to the ideal lowpass filter gain curve. We shall not
discuss these specific filters because our objective in this section has been only to
introduce the algebraic technique for the determination of the system function. We
only used the Butterworth filter to illustrate the algebraic concepts and a design
technique using them. Texts specifically concerned with filter design discuss these
and other types of filters in detail.
PROBLEMS
91 Obtain the result illustrated by Fig. 9.1lb algebraically instead of geome
trically as in the text.
92 Figure 9.12b is the graph of the gain of a one pole lowpass filter. From Eq.
(9.16b), the exact expression for the gain is
Expand the expression in a power series and thus shown that, for small values
of o,the gain is approximately
so that, for small values of o,the shape of the gain curve is parabolic.
300 SPLANE VIEW OF GAIN AND PHASE SHIFT
93 For small values of 8, the approximation tan 8 x 8 was used to obtain Eq.
(9.17e) and the approximation sin 8 x tan8 was used to obtain Eq. (9.1%).
For each of these approximations, determine the maximum value of 8 in
degrees for which the approximation error is less than 1%, 5%, and 10%.
94 Show that the maximum phase shift given by Eq. (9.280 occurs at the
frequency coo = @.
96 +
The system function of an allpass system is Ha@)= (s  a ) / @ a) 0 > a.
(a) Determine a differential equation relating to the system input, x(t), and
output, r@>.
@) Determine the system unitstep response.
(c) What is the asymptotic value of the unitstep response as t + 00.
Explain.
where A > 0.
(a) Show that the system function can be expressed as H(s) = H,(s)H,(s) in
which H,(s) and Hb(s) are the system functions discussed in Section 9.1
so that the amplifier can be viewed as the tandem connection of those two
systems.
@) Use the result of part a to show that the amplifier gain is approximately A
in the audio frequency range and that the gain falls 3dB at the band
edges. Determine the lower and the upper frequencies of the band edges.
(c) Use the result of part a to show that the amplifier phase shift is
approximately zero in the audiofrequency range. What is the phase
shift at the lower and the upper frequencies of the band edges determined
in part b.
1
+ 2nk].
h(t 1 qmn)n 
(b) Thus show that 2= P I ean.
h(tm) mn
(c) Use the result of part b to show that Q = 
h(tJ
In 
h(ti7l)
The measurement technique using this result is to measure the ratio of the
value of two maxima of the impulse response at two instances separated by
(m  n) cycles of the sinusoid. The number of cycles of separation can be
obtained by counting the number of peaks of h(t). The Q is then 71 times the
number of cycles of separation divided by the logarithm of the ratio. This is
one technique used to determine the Q of a microwave cavity for which the Q
often is above 20,000.
302 SPLANE VIEW OF GAIN AND PHASE SHIFT
913 It is desired to design an LTI filter with a maximum filter gain of 100 with
two bandpass regions: one with a center frequency of 1 MHz and a 3dB
bandwidth of 50 kHz, the other with a center frequency of 2 MHz and also
with a 3dB bandwidth of 50 kHz.
(a) Use the tandem connection of two systems of the form discussed in
Section 9.4 to determine an approximate system function of the desired
filter.
(b) The maximum gain in each bandpass region is not the same. How can the
system function determined in part a be modified so the maximum gain
in each bandpass region is equal?
1 SU
HI(s)
=, > a and H2(s)= > a
s+a
(T
(s + a)2 ' (T
915 Consider the following three systems with the responses y,(t), n = 1 , 2 , or 3,
for the input x(t). For each system, a > 0 and the RAC is (T > a.
sa 1 sa
1 . H,(s) = 2. 4 ( s ) =  3. H3(s)=
~
s+a s+a
~
(s + al2
(a) Determine the unitimpulse response, h,(t), of each system.
(b) Determine the unitstep response, s,(t), of each system.
(c) Show that h3(t) = h , ( t ) * h2(t)and thus show that the third system can be
viewed as a lowpass filter connected in tandem with an allpass system.
(d) Thus show that y2(t) y 3 ( t ) = y2(t)* [2ae"u(t)] so that the difference
of their outputs is the response of a lowpass filter with the input y2(t).
916 (a) Determine the general expression for the system function of a lowpass
fourthorder Butterworth filter.
@) Determine a differential equation relating the input, x(t), and output, y(t),
of the filter.
917 (a) Use Eq. (9.51) to determine co, and n such that the gain of the
Butterworth filter is greater than or equal to 0.9 at 4000Hz and less
than or equal to 0.1 at 6000 Hz.
PROBLEMS 303
(b) Sketch the splane diagram of the system fbnction and label the pole and
zero locations.
918 Use the design technique discussed in Section 9.5 to determine the system
h c t i o n of a causal and stable LTI system required for the square of the
system gain to be
2A2 2A2
IHo’W>l2 =
2 + 3 (g+(; = [+1 p +
CHAPTER 10
INTERCONNECTION OF SYSTEMS
There are three basic ways in which two LTI systems can be connected to form
another LTI system. The three ways are parallel, tandem, and feedback. All LTI
systems that are composed of the interconnection of a number of subsystems can be
analyzed in terms of these three basic connections. The theory of the parallel and
tandem connections has been covered in the previous chapters; also, a simple model
of feedback was discussed in Section 1.6. We’ll begin by summarizing those results
because we shall need them in subsequent sections.
x ();
t j
+ :
...........................................................
respectively, then the parallel connection is an LTI system with the unit impulse
response
with the RAC equal to the overlap of the RACs of systems A and B.
where the asterisk means the convolution of the two functions. Thus the system
function of the tandem connection is
with the RAC equal to the overlap of the RACs of systems A and B.
..............................................
z(t) I B 1)I
Fig. 10.13 Feedback connection of two systems.
308 INTERCONNECTION OF SYSTEMS
thermostat, which activates the room heater and thereby causes the room temperature
to rise. This temperature rise differs from point to point in the room because of the
heat flow in the room. Because we are interested in the temperature at your chair, we
choose the output, y(t), to be the temperature at your chair. For this model then,
system A in Fig. 10.13 is the system model of the thermostat, heater, the room heat
flow from the room heater to your chair, and the resulting temperature rise at your
chair. System B is then a model of the relation between the temperature at your chair
and that at the thermostat.
Before analyzing the feedback system, let us note the following general state
ments that can be made about a feedback system:
1. If systems A and B are causal systems, then the feedback system is a causal
system. This result is easily seen because if system A is causal, then its output, y(t),
does not depend on the future of its input, e(t). If system B also is causal, then its
output does not depend on the future of its input, y(t). Thus, no operation within the
feedback system depends on the hture of its input. Consequently, the feedback
system output, y(t), cannot depend on the future of its input, x(t) so that the feedback
system is causal. Note that this result is true whether the systems A and B are linear
or nonlinear and also whether they are timeinvariant or timevarying.
2. If systems A and B are timeinvariant, then the feedback system is time
invariant because all feedback operations are then timeinvariant operations. Note
that this result is true whether the systems A and B are linear or nonlinear.
3. If systems A and B are linear, then the feedback system is linear because all the
feedback operations are then linear operations and we have shown that the parallel
and tandem connection of linear operators is a linear operator. Note that this result is
true whether the systems A and B are timeinvariant or timevarying.
We thus conclude from the above statements that if systems A and B are causal
LTI systems, then the feedback system is a causal LTI system. However, if systems
A and B are stable systems, we cannot conclude that the feedback is necessarily
stable. This observation can be noted from the analysis of the simple feedback model
in Section 1.6.
Because we are concerned with the analysis of physical LTI systems in this text,
we shall analyze the case for which systems A and B are causal LTI systems in the
next section.
We shall analyze the feedback connection in which A and B in Fig. 10.21 are causal
LTI systems with the unitimpulse responses h,(t) and hb(t), respectively. As shown,
the feedback system input is x ( t ) and the feedback system output is y(t). The input of
system A is e(t) = x ( t )  z(t), where z(t) is the output of system B. The input of
system B is y(t), so that z(t) is obtained by feeding back an LTI operation on the
10.2 ANALYSIS OF THE FEEDBACK SYSTEM 309
%++T{TJ2 Z t
................................................... ..............
system output, y(t). The path that contains system A is called the feedforward path,
and the path that contains system B is called the feedback path. The loop formed by
the feedforward and feedback paths is called the feedback loop.
Because systems A and B are considered to be LTI systems, we have from our
discussion in the previous section that the feedback system is an LTI system. Thus
the feedback system can be characterized by a unit impulse response, h(t), so that its
output, y(t), can expressed as
where the asterisk indicates the convolution. From Fig. 10.21, the system relations
are
(10.22)
(10.23)
310 INTERCONNECTION OF SYSTEMS
Observe that Eqs. (10.22) have been transformed into algebraic ones that are easily
solved for Y(s) in terms of X(s). The solution is
(10.24)
(10.25)
(10.26)
and the RAC, as we concluded above, is to the right of all the poles of H ( s ) .
A way to remember the expression for H ( s ) is to note that the numerator is the
feedforward system function. The denominator is one plus the product of the feed
forward and feedback system functions. The plus in the denominator is due to the
+
minus sign in the expression for e(t).If instead, e(t) = x(t) z(t), then the sign in the
denominator of Eq. (10.26) would be minus. Thus the sign in the denominator is
simply opposite the sign in the expression for e(t). The following examples illustrate
some applications and interpretations of this result.
0 > M (10.27a)
and
A
10.2 ANALYSIS OF THE FEEDBACK SYSTEM 311
In accordance with our discussion in Section 10.1, the feedback system is causal.
Thus the RAC must be to the right of the pole at s = (a + AB). The w axis must lie
in the RAC for the system to be BIBOstable. Thus, the feedback system is stable
only if (a + A B ) > 0. This inequality requires
a
B>  (10.29)
A
Thus the feedback system is stable only if the amplifier gain, B, in the feedback path
satisfies Eq. (10.29), whereas the feedback system is not stable if the amplifier gain,
B, in the feedback path does not satisfy Eq. (10.29).
We also can obtain these results from timedomain considerations. Note from
Eq. (10.28) that the unitimpulse response of the feedback system is
and
(10.21 lb)
Now both systems A and B in our example are simple lowpass filters. Substituting
these expressions into Eq. (10.26), we obtain
We note that there is one zero at s = /l and two poles at the roots of the quadratic in
the denominator. In accordance with our discussion in Section 10.1, the feedback
system is causal, so that the RAC must be to the right of both poles. As we discussed
in Section 8.2, the o axis must lie in the RAC for the system to be BIBOstable, so
that the feedback system is stable only if both poles lie in the left half of the s plane.
312 INTERCONNECTION OF SYSTEMS
Thus we must factor the denominator quadratic to determine the conditions for
which the feedback system is stable.
To simplify our analysis, we’ll determine the values of A and B for which the
feedback system is stable for the special case in which a = 2 and B = 4. Substituting
these values of a and B into our expression for H(s), we obtain
H(s) =
A(s + 4)  A(s + 4) (10.213)
s2 + 6s + ( 8 +AB)  (s + 3)2  (1  A B )
with the RAC to the right of both poles. To obtain the poles, we factor the denomi
nator quadratic as
Thus the two poles are at s = 3 f d m .The pole locations for several values
of AB are listed below:
Value of
AB Pole Location Pole Location
 99 7  13
 24 2 8
8 0 6
0 2 4
1 3 3
2 3 + j 3  j
5 +
3 j 2 3 j 2
17 +
3 j 4 3 j 4
37 +
3 j 6 3 j 6
Figure 10.22 is a plot of the pole locations in the s plane. Note that as AB
increases from a very large negative value to AB = 1, one pole moves from a
very large negative value of sigma to n = 3 while the other pole moves from a
very large positive value of sigma to n = 3. When AB = 1, there is a secondorder
pole at cr = 3 because both poles are there. For AB > 1, the poles move parallel to
the o axis at n = 3; one pole moves up and the other moves down such that the
two pole locations are conjugates as they must be because the coefficients of the
polynomial, Eq. (10.214a), are real. Techniques to simplify the plotting the locus of
the poles in the s plane such as this example have been developed. It is called the
root locus technique, and computer software is available with which rootlocus plots
can be generated.
Observe for our example that one pole is in the right half of the s plane if
AB < 8 and on the o axis if AB = 8, so that the feedback system is not
10.2 ANALYSIS OF THE FEEDBACK SYSTEM 313
BIBOstable if AB 5 8. However, both poles are in the left half of the s plane if
AB > 8. Thus we conclude that the feedback system is BIBOstable only if
AB > 8.
Also, from our discussion of the bandpass system in Section 9.4, observe that for
large values of AB, the feedback system is a bandpass filter with a bandwidth of
approximately 6rad/s and a center frequency of approximately d m .By
connecting two lowpass filters in a feedback arrangement, we have realized a
bandpass filter with a given bandwidth and a center frequency which can be adjusted
simply by varying the value of the gain product, AB.
4 B
H,(s) = ~
s+l'
rT> 1 and H&) = A + s+B' > B (10.215)
Substituting in Eq. (10.26), the expression for the feedback system function, H(s),
is
(10.216)
314 INTERCONNECTION OF SYSTEMS
We shall determine the values of the constants A, B, and p in the expression for
hb(t) such that the zero of the feedback system function is at s = 9 and the two
poles of the feedback system are at s =  8 + j 4 and s = 8 j4. From
Eq. (10.216), we observe that the zero is at s = p. Thus we require /?= 9. To
determine the constants A and B, we note that for the desired conjugate pole loca
tions, the denominator of H(s) must be
This equation is true for all values of s only if the polynomial coefficients are equal.
Thus we have two equations that must be satisfied:
1 + p + 4 A = 16 and p + 4 B + 4 A p = 80 (10.219)
The simultaneous solution of these two equations for A and B is A = 1.5 and
B = 4.25. Thus the feedback system function, H(s), will have the desired poles
and zero if the LTI system with the unitimpulse response
(10.22 1)
Example 4 The system inverse of an LTI system is discussed in Section 8.2. The
system inverse can sometimes be realized in the form of a feedback system. As an
example, consider the problem of eliminating ghosts on a TV screen. Ghosts are the
result of multipath. The received TV signal may arrive from the transmitting antenna
via many paths due to it being reflected from various objects such as buildings and
mountains. This results in ghosts on the TV screen because the travel time to the
receiver is slightly different for each path. To illustrate the use of feedback to
eliminate ghosts, we first consider the case in which there is only one reflected wave
that is reflected without distortion. The received signal then is
where T is the time delay due to the extra path length and IAl < 1 because, from
energy considerations, the magnitude of the reflected wave is smaller than the
incident wave. With the use of Table 7.41, the Laplace transform of this expression
is
with the region of convergence being that of x(t). Thus we can model the received
signal, y(t), as the response of an LTI system with the system function,
G(s) = 1 + A C T (10.224a)
Note that this system function is not a rational function of s. The algebraic expres
sion for the system function of the system inverse is
1 1
H(s) = ~ = (10.224b)
G(s) 1+ A c S *
By comparing this system function with that of a feedback system, Eq. (10.26), we
note that H(s) can be realized as a feedback system with
for which
The realization of this system, which will eliminate the ghost, is shown in Fig.
10.23.
For the feedback system to be causal, we choose the RAC to be to the right of all
the poles. Thus for stability we require all the poles to be in the left half of the s
plane. Note that the poles of H(s) are the zeros of G(s), which are those values of s
for which
1 +AeST = 0 (10.226)
............_.................................................
__.
~ Y(r)
We determine the solutions of this equation by equating the real and the imaginary
parts of this equation as follows. First express s as s = (T +jo to obtain
Thus we require the values of (T and o that satisfy this equation. For this, consider
the case for which A > 0. Equation (10.227) then requires
ejmT = 1 (10.228a)
and
or
(10.229a)
The correspondmg values of (T are the solutions of Eq. (10.228b), which are
1
CJ = 1nA (10.229b)
T
Thus we note that there are an infinite number of poles of H ( s ) located on a line
parallel to the o axis at (l/T)lnA +j[(2n +
l)rc/Tl for n = 0, f l , f 2 , . . . . Note
that the determination of the poles required the solution of Eq. (10.226), which is a
transcendental equation with an infinite number of roots. In accordance with our
discussion in Section 8.2, the feedback system is causal and stable only if (AI < 1
because In IAl < 0 only if IAl < 1. The ghost described by Eq. (10.222) would be
eliminated if the feedback system were connected at input of the TV receiver.
Return now to Eq. (10.26), the general equation for the system function of a
feedback system. Let the system hnctions in the feedfonvard and feedback path be
rational functions of s. The system functions then can be expressed as
(10.230a)
zeros of D(s) are the system function poles. Substituting these expressions into
Eq. (10.26), we obtain
Observe from this expression that the zeros of H(s) are the zeros of Ha@)and also
the poles of Hh(s).Note that the location of the zeros of H(s) do not change as A and
B are varied. The location of the poles of H ( s ) are a function of AB; they can be
determined by determining the roots of the denominator polynomial. For the
common case in which the total number of zeros of systems A and B does not
exceed the total number of their poles, the degree of the denominator polynomial is
observed to be equal to the sum of the degrees of the denominator polynomials of
systems A and B, so that the number of poles of the feedback system is equal to the
number of poles of system A plus the number of poles of system B.
It is not unusual for a system function to have three or more poles, so that we
expect many feedback systems to have at least several poles. It would be nice to
determine the locations of the feedback system poles as in our examples. However,
while formulas exist for the roots of second, third, and fourthdegree polynomials,
no general formulas exist for the roots of higherdegree polynomials. The reason for
the nonexistence of a general formula for a polynomial of degree higher than four is
not that no one has been sufficiently clever to determine it, but rather because it can
be shown that no such formula can exist. That is, even though, as discussed in
Appendix A, the fundamental theorem of algebra assures us that a polynomial of
degree n has exactly n roots, it can be shown that there can be no general formula for
the roots of a polynomial of degree higher than four. Of course, the feedback system
pole locations can be determined by using the computer to determine the polynomial
roots. One efficient procedure utilizes the rootlocus technique mentioned above.
However, if we only need to know whether the feedback system is stable, we do not
need to know the exact locations of the feedback system poles. In accordance with
our discussion in Section 8.2, we only need to know whether all of the feedback
system poles lie in the left half of the s plane. If they do, then the causal feedback
system is stable; if not, then the causal feedback system is not stable. An efficient
procedure has been developed to determine whether all the poles lie in the left half of
the s plane. The procedure is called the RouthHunvitz algorithm, which we discuss
in the next section.
If we only desire to know whether a causal system is stable, then it is not necessary
to determine the specific system pole locations. In accordance with our discussion in
Section 8.2, we just need to determine whether all of the feedback system poles lie in
the left half of the s plane. The efficient procedure mentioned in the last section by
318 INTERCONNECTION OF SYSTEMS
which one can determine whether all the poles lie in the left half of the s plane is
called the RouthHurwitz algorithm,' which is simple to implement on a computer.
With this algorithm, we can determine how many roots of a polynomial lie in the left
half of the s plane, how many lie on the w axis, and how many lie in the right half of
the s plane. The various derivations of the algorithm do not contribute anything to
our discussion, so that we shall only discuss the algorithm.
If either of these two conditions is not satisfied, then there is at least one root of D(s)
in the right half of the s plane or on the w axis, so that the system is not stable. Note
that these two conditions are necessary but not sufficient, so that it is still possible
that the system is not stable even though the two conditions given above are satisfied.
However, because these two conditions are so simple, it is best to determine whether
they are satisfied before proceeding with any test. If the two conditions are satisfied,
then RouthHurwitz criterion is used to determine whether all the roots are in the left
half of the s plane. For example, consider the following polynomials:
D,(s) = s2 +9 (10.32a)
D&) = s2 + 2s  3 (10.32b)
D,(s) = s4 + 3s3 + 6s2 + 38s + 60 (10.32~)
All the roots of D,(s) do not lie in the left half of the s plane because al = 0, so that
condition 1 above is not satisfied (the roots of D,(s) arej3 and j3). Similarly, all the
roots of D&) do not lie in the left half of the s plane because condition 2 above is
not satisfied (the roots of D6(s)are 3 and 1). However, all the roots of D,(s) also do
not lie in the left half of the s plane even though both conditions above are satisfied
'Equivalent algorithms were developed independently by E. J. Routh in 1877 and by A. Hurwitz in 1895.
We use both names in order to recognize the important contributions of both men. However, the specific
algorithm described above is the Routh form of the algorithm because it is easier for us to use.
10.3 THE ROUTtCHURWlTZ CRITERION 319
(the roots of D,(s) are 2, 3, 1 +j3, and 1 j3). For polynomials such as D,(s),
the RouthHurwitz criterion must be used to determine whether all the roots lie in
the left half of the s plane.
If condition 2 is satisfied, then all the coefficients of D(s) are either positive or
negative. If they are all negative, then the coefficients of D(s) are all positive and
its roots are the same as those of D(s). Thus, we need only consider the case for
which all the coefficients of D(s) are positive. For this reason, we shall assume that
all the coefficients of D(s) are positive in our description of the RouthHurwitz
criterion.
(10.34a)
( 10.34b)
(10.3 4 ~ )
(10.34d)
320 INTERCONNECTION OF SYSTEMS
The numbers in the fourth row, labeled s "  ~ , are determined in the same manner
from the determinates of the numbers in the two rows above it. Thus
(10.35a)
(10.35b)
(10.35~)
The numbers in each succeeding row are determined in the same manner from the
determinates of the numbers in the two rows above it until the last row, the (n 1)th +
row labeled so, in which the only one nonzero number is
(10.36)
The reason for the row labels will be discussed later. The shape of the Routh array
will be seen to be triangular. A usefd fact in developing the Routh array is that any
row can be multiplied or divided by a positive number in order to simplijj the
numerical calculation without altering the results of the RouthHurwitz criterion.
The criterion states that the number of roots of D(s) that lie in the right half of the s
plane is equal to the number of changes of sign of the numbers in the first column of the
array.
Thus, the necessary and sufficient condition that all the roots of D(s) lie in the left
half of the s plane is that all the numbers in the first column be positive. A special
case is one in which one of the calculated values in the first column is zero. Before
discussing this special case, we'll consider some examples that are not special cases.
As our first example, consider the polynomial D,(s) given by Eq. (10.32c). The
Routh array for this polynomial is
s 4 : 1 6 6 0 0
s 3 : 3 3 8 0 0
s2 : y 60 0 0
S I : 65 0 0 0
s o : 60 0 0 0
10.3 THE ROUTHHURWITZ CRITERION 321
The numbers in the row labeled s2 are obtained from Eq. (10.34) as follows:
38
and the number in the row labeled SI is obtained from Eq. (10.35) as follows:
There are two sign changes in the first column of the array; one from 3 to 2013 and
the other from 2013 to 65. Thus, in accordance with the RouthHunvitz criterion,
there are two righthalfplane roots of D,(s). Because there are a total of four roots,
the other two roots must lie in the left half of the s plane. As stated above, the roots
+
of D,(s) are 2, 3, 1 j3, and 1 j3, which agrees with the result obtained using
the RouthHurwitz criterion.
As a second example, we'll determine the conditions that are necessary and
sufficient that all the roots of a cubic polynomial lie in the left half of the s plane.
For this, we consider the polynomial
+
~ ( s=) aOs3 a,s2 + a2s + a3 (10.37)
In determining this Routh array, the third row was multiplied by the positive number
u l , which, as stated above, does not alter the results of the RouthHunvitz criterion.
Because all the coefficients are assumed positive, we note that all the entries in the
first column are positive if a0a3< ala2. This then is a necessary and sufficient
condition that all the roots of a cubic lie in the left half of the s plane.
As a third example, consider the sixthdegree polynomial
The first two rows of the Routh array for this polynomial are
s6 : 1 5 3 1 0
s5 : 6 4 2 0 0
322 INTERCONNECTION OF SYSTEMS
26
s 4 :  l6 1 0 0
6 6
However, this row can be multiplied by a positive number which, as stated above,
does not alter the results of the RouthHurwitz criterion. Thus we multiply this row
by 3 to obtain
s4: 1 3 8 3 0 0
We then continue determining the Routh array to obtain for the first four rows:
s 6 : 1 5 3 1 0
s 5 : 6 4 2 0 0
s 4 : 1 3 8 3 0 0
s3 : 3 fi0 0 0
Again, without altering the results of the RouthHurwitz criterion, the numbers in
the fourth row can be made more convenient for calculations by multiplying them by
the positive number 13/4 to obtain
s 6 : 1 5 3 1 0
s 5 : 6 4 2 0 0
s4 : 13 8 3 0 0
s 3 : 1 2 0 0 0
s 6 : 1 5 3 1 0
s 5 : 6 4 2 0 0
s 4 : 1 3 8 3 0 0
s 3 : 1 2 0 0 0
s2 18 3 0 0 0
s : 1 3 0 0 0 0
s o : 3 0 0 0 0
Again, for convenience without altering the results of the RouthHurwitz criterion,
the row labeled s was multiplied by the positive number, 6.
There are two sign changes in the first column; one from 1 to 18 and the other
from  18 to 13. Thus, in accordance with the RouthHurwitz criterion, there are two
righthalfplane roots of D(s). Because there are a total of six roots, the other four
roots must lie in the left half of the s plane. This agrees with the actual root locations,
which are 5.1623, 7.025, 0.3786 fj0.5978, and +0.3110 fj0.6738.
10.3 THE ROUTMURWITZ CRITERION 323
We now consider the special case in which a zero occurs in the first column of the
Routh array. There are two types of this case to consider:
1. The first entry of a row is zero, and at least one of the other entries of that row
is nonzero.
2. All the entries of a row are zero.
For the first special case, the procedure is to replace the zero in the first column by
E,which has a small positive value, and let E + O+. For example, consider the
polynomial
+
D(s) = s4 s3 + s2 + s + 2 (10.39)
The first three rows of the Routh array for this polynomial are
s 4 : 1 1 2 0
s 3 : l l O O
s 2 : 0 2 0 0
Because the first entry of the third row is zero and not all values of the row are zero,
we replace the zero in the first column by E , which has an arbitrarily small positive
value, and proceed to compute the rest of the Routh array, which is
s 4 : 1 1 2 0
s 3 : 1 1 0 0
2 : E 2 0 0
s o : 2 0 0 0
For very small positive values of E , the value of the first entry in the fourth row is
negative, for which there are two sign changes in the first column: one from E to
and the other from ( E  2 ) / ~to 2. Thus the polynomial has two roots in the
(E  2 ) / ~
right half of the s plane and, because there are a total of four roots, the other two
roots are in the left half of the s plane, which agrees with the following root
locations: 0.9734 fj0.7873 and +0.4734 &j1.0256.
The second special case is the one in which all the entries of a row are zero. This
occurs only when there are roots that are symmetric relative to the origin. A pair of
roots are symmetric relative to the origin if one is at s = so = go + j w o and the other
is at s = so = go joo. Because the complex roots of a polynomial with real
coefficients must occur in conjugate pairs, this means that if a root at s = so is
symmetric relative to the origin, then the polynomial would have roots at
s = fo, & j w o , so that, if go # 0 and wo # 0, the roots would be symmetric relative
to the 0 axis and also the w axis. Such a case would be unusual. The more usual case
324 INTERCONNECTION OF SYSTEMS
of symmetric roots for a polynomial with positive coefficients is one for which
go = 0, so that there are roots on the o axis at s = fjo,. As we shall see, a
great deal of information about the roots can be obtained when all the entries of a
row are zero.
To discuss this special case, we shall consider a specific example and generalize
our discussion. For this, consider the polynomial
The first two rows of the Routh array for this polynomial are
s 4 : 1 6 8 0
s3 : 3 1 2 0 0
Before proceeding, we divide the second row by 3 to simplify the numbers and
continue:
s 4 : 1 6 8 0
s3 : 1 4 0 0
s 2 : 2 8 0 0
We again simplify the numbers by dividing the third row by 2 and then continue:
s4 : 1 6 8 0
s3 : 1 4 0 0
s 2 : 1 4 0 0
s ' : o o o o
All the entries in the row labeled s are zero. We would have obtained this same result
without simplifying the numbers in the table. When all the entries in a row are zero,
we form a polynomial, called the auxiliary polynomial, from the entries in the row
above the row of zeros. The degree of the auxiliary polynomial is given by the row
label, the polynomial only contains every other power of s, and the entries in the row
are the polynomial coefficients. For our case, the row is the one labeled s2.Thus the
+
auxiliary polynomial for our case is p(s) = s2 4. An important property of the
auxiliary polynomial is that its roots are the symmetric roots that caused the row
below it to contain all zeros. The roots of the auxiliary polynomial for our case is
s = f j 2 , so that D(s) has these symmetric roots. We now can complete the Routh
array to determine if any of the other roots lie in the right half of the s plane. For this,
the row of zeros is replaced by the coefficients of the derivative of the auxiliary
10.3 THE ROUTHHURWITZ CRITERION 325
polynomial. The derivative of the auxiliary polynomial for our case is p'(s) = 2s, so
the completed Routh array is
s 4 : 1 6 8 0
s3 : 1 4 0 0
s 2 : 1 4 0 0
s : 2 0 0 0
s O : 4 0 0 0
Because all the entries in the first column are positive, we conclude that there are no
roots in the right half of the s plane. Our result agrees with the actual root locations
of D(s), which are 1, 2, and f j 2 .
Except for the special case of symmetric roots, the RouthHurwitz criterion does
not directly enable us to determine the exact root locations of a given polynomial.
The criterion, however, can be used to determine the real part of each root, which is
its distance from the w axis, to any desired degree of accuracy.2 For this, replace s
+
with s a in D(s), the general nthorder polynomial given by Eq. (10.31). If
+ +
D(s) = 0 for s = p , then clearly D(s a ) = 0 for s a = p or, equivalently, for
s = a + p . Thus we observe that the roots of the resulting polynomial, D(s a), +
will then be the roots of D(s) shifted to the left by a units (or to the right by a
units). The procedure, which is simple to implement on a computer, then is to
determine the shift a for which the roots of the resulting polynomial lie on the w
axis. This is accomplished by using the RouthHurwitz criterion to determine the
number of roots in the right half of the s plane of the polynomial D(s a). Now, +
The polynomial coefficients of D(s + a ) , p,, in terms of the coefficients of D(s), a,,
of Eq. (10.31) are
(10.312)
D(s)=s2+4s+16
'It is important to know more than how to utilize a specific procedure. An understanding of the concepts
involved lends deeper insight of the procedure and sometimes can be utilized to generalize and to obtain
more information than previously conceived. I've developed this procedure mainly to illustrate this.
326 INTERCONNECTION OF SYSTEMS
The roots of this polynomial are s = 2 f j2, so that it has no roots in the right half
of the s plane. However, we’ll determine this by the technique described above. Now
D(s + a) = B 2 s 2 + BlS + Bo
in which, from Eq. (10.312),
22
(2 + k ) ! k
82 = ‘2fk 2 !k!a = (a,)(l)(a 0) = a2 = 1
k=O
81 =
2 1
‘l+k
(1
l
!k
+ k)!
!ak = (q)(l)(aO)+ (a2)(2)(a) = 4 + 2a
k=O
and
BO =
20
aO+k
(0 + k)!
O!k!ak = (a& l)(a0) + (al)(l)(a) + (a2)(1)(a2)= 16 + 4a + a2
k=O
We fist try a = 5 to shift the roots to the right by +5 units. Then, from the
RouthHunvitz criterion, we determine that D(s  5) = s2  6s 21 has two roots +
in the right half of the s plane. Thus we know that the real part of the roots of D(s)
must be between 5 and 0. Next, choose a = 5/2 in order to half the range. Using
the RouthHunvitz criterion, we determine that D(s  5/2) = s2  s 12.25 again +
has two roots in the right half of the s plane. The real part of the roots of D(s) thus
must lie between 5/2 and 0. We next half the possible interval by choosing
a = 5/4. Using the RouthHunvitz criterion, we determine that D(s  5/4) =
s2+ +
1.5s 12.5625 has no roots in the right half of the s plane. Thus the real
part of the roots of D(s) must lie between 5/2 and 5/4. We again half the
possible interval by choosing a = 15/8. Using the RouthHunvitz criterion, we
determine that there are no roots of D(s  15/8) in the right half of the s plane.
Consequently, the real part of the roots of D(s) must lie between 5/2 and  15/8.
Observe that the range in which the real part of the roots can lie is reduced by a
factor of 2 each time. By continuing in this manner, the real part of the roots can be
determined to any degree of accuracy. With a computer, this is simply a DOloop.
As our last example, the application of the RouthHunvitz criterion to determin
ing the stability of a feedback system will be illustrated. For this, consider the
feedback system shown in Fig. 10.21 in which
1
HJs) = o>o ( 10.313a)
~
s2(s + 5) ’
10.3 THE ROUTICHURWITZ CRITERION 327
and
s+l
) X K ,
H ~ ( s= c> 2 (10.3 13b)
Both systems are causal because the RAC is to the right of all the poles for each
system. Thus, in accordance with our discussion in Section 10.2, the feedback
system is causal. However, note that system A is not stable because the w axis
does not lie in the RAC. We desire to know the values of K , if any, for which the
feedback system is stable. From Eq. (10.26), the feedback system function is
H(s) =
s2(s + 5) (10.314a)
1 S + l K
l+
s2(s+5) s + 2
s+2
H(s) = (10.314b)
fi + 7s3 + 10s2 +K.s +K
The RAC for H ( s ) is to the right of all its poles because the feedback system is
causal. For it also to be stable, the w axis must lie in the RAC. Thus, for the system
to be both causal and stable, all the poles must lie in the left half of the s plane in
accordance with our discussion in Section 8.2. We use the RouthHunvitz criterion
for this determination. The feedback system poles are the roots of the denominator
polynomial
The first three rows of the Routh array for this polynomial are
s4 : 1 10 K 0
s 3 : 7 K O 0
70  K
s2 : ~ K O 0
7
We multiply the third row by 7 in order to simplify calculations and continue:
s4 : 1 10 K 0
s3 : 7 K 0 0
s2 : 70K IK 0 0
(70  K)K  49K
s : 0 0 0
70K
so : 7K 0 0 0
328 INTERCONNECTION OF SYSTEMS
The fourth row was not simplified by multiplying it by (70  K). The reason is that
we can only multiply a row by a positive constant without altering the results of the
RouthHurwitz criterion. Because (70  K) would not be positive if K > 70, we
would only able to apply the criterion for values of K less than 70. Now, all the roots
of D(s) lie in the left half of the s plane if and only if all the entries in the first column
of the Routh array are positive. From the array above, we thus require 70  K > 0,
(70  K)K  49K = (21  K)K > 0, and K > 0. All these conditions are met only
if 0 < K < 21. Thus we conclude that the feedback system is stable only if
0 < K < 21.
Note from the Routh array that all the entries of the row labeled so are zero if
K = 0. From the above discussion of the second special case, this means that there
are symmetric roots of D(s) if K = 0. Of course, Hb(s)= 0 for K = 0 so that, from
Eq. (10.26), H(s) = H,(s). This also can be seen by observing from Fig. 10.21 that
if Hb(s) = 0, then z(t) = 0, so that there is no feedback and H ( s ) = H,(s). The
symmetric root for K = 0 is the double root of H,(s) at s = 0.
We also expect roots on the w axis for K = 21 because all the roots are in the left
half of the s plane for K < 2 1 and there are roots in the right half of the s plane for
K > 2 1. This means that some roots crossed the w axis, so that there must be roots
on the w axis for K = 21. The first four rows of the Routh array for K = 21 are
s4 :1 10 21 0
s 3 : 7 21 0 0
s2 : 49 147 0 0
s : o 0 0 0
+ +
The auxiliary equation isp(s) = 49s2 147 = 49(s2 3). The roots of this equation
are s = & j d . These then are the waxis roots of D(s) for K = 2 1 (the other two are
at 5.791 3 and  1.2087); thus, as K increases, the locus of the roots crosses the w
axis at w = *jd.
In our example, the determination of the values of K for which the feedback
system is stable was easily determined. For some feedback systems with more poles
and zeros, the determination of the values of K for which the system is stable may
not be as simple. However, the RouthHurwitz criterion is easily programmed on a
computer so that the system stability can be determined quickly for a large number
of values of K. From this, the range (or ranges) of K for which the system is stable
can be obtained.
The block diagram model of a physical system often is the interconnection of several
subsystems. For example, consider the physical system shown in Fig. 10.41. As
shown, the system consists of three blocks with masses M , , M2, and M3. The blocks
are connected by springs with spring constants K, and K2. Also, there is friction
between the blocks and the surface. The friction is assumed to be sliding friction, so
10.4 SYSTEM BLOCK DIAGRAMS 329
that the frictional force is proportional to the block velocity. The positions of the
blocks are x l ( t ) ,x2(t), and x3(t) as shown on the figure. The system input is the
external force, f ( t ) , and the system output is the position of the third block, x3(t).
The force equation for the first block is
To obtain a block diagram of the given system, we take the Laplace transform of the
above system equations. We work with the Laplace transform of the differential
equations since they are algebraic equations which can be manipulated easily. The
transformed equations are
and
(10.43a)
330 INTERCONNECTION OF SYSTEMS
I
I I
I I
(c)
A block diagram for this equation is shown in Fig. 10.42b. The input, X,(s), is
available as the output of the system shown in Fig. 10.42a. However, to complete
this diagram, we need X3(s).This is obtained from the third equation, Eq. (10.42c),
by solving it for X3(s)as
(10.43b)
Xl(d
I
I K.
A
with the gain K,. Furthermore, the position of the third block, x3(t), affects the
position of the second block, x2(t), because of the spring connecting the two
blocks. The model of this effect is the feedback path with the gain K2. Thus this
model allows us to determine the specific effect that one block of the system has on
another. Therefore we can obtain a great deal of insight concerning the system
behavior from its block diagram. Note that, in general, if some system components
interact with other system components as in our example, then the system model will
contain embedded feedback loops.
To determine the system output for a given input of a system with embedded
feedback loops, we could solve the system equations simultaneously. For our
example, we could solve the system equations, Eqs. (10.42), simultaneously by
eliminating X,(s) and X2(s)and thus obtain an expression for X,(s) as
in which H ( s ) is the system function. The RAC for H(s) is to the right of all its poles
because the system is causal. However, we can obtain H(s) directly from the system
block diagram by a method called block diagram reduction.
Also, we can move a pickoff point forward by noting the equivalence of the two
diagrams shown in Fig. 10.45. In the diagram, Hcl(s) is the inverse of system B, so
that H;'(s) = l/Hb(s) with a RAC which overlaps that of Hb(s). If system B is
causal, then its RAC is to the right of all its poles. The inverse of system B then can
also be chosen to be causal by choosing its RAC to be to the right of all the poles of
H;l(s). Because H;l(s) = l/Hb(s), these poles are the zeros of Hb(s), so that the
overlap of the RACs is to the right of all the poles and zeros of Hb(s). Again, the
Fig. 10.44
332 INTERCONNECTION OF SYSTEMS
Fig. 10.45
equivalence of these two systems is established by noting that Eq. (10.45) is satis
fied for each diagram in Fig. 10.45.
We also can move summation points. A summation point can be moved forward
by noting the equivalence of the two diagrams shown in Fig. 10.46. The equivalence
of these two systems is established by noting for each diagram that
Also, we can move a summation point backward by noting the equivalence of the
two diagrams shown in Fig. 10.47. In the diagram, H'(s) is the inverse of the
system so that H'(s) = l/H(s) with a RAC which overlaps that of H(s). If the
system is causal, then, in accordance with our discussion concerning Fig. 10.45, the
RAC of its causal inverse is to the right of all the zeros of H(s). The equivalence of
the two systems shown in Fig. 10.47 is established by noting that Eq. (10.47) is
satisfied for each diagram.
Block diagrams are reduced by using these four system equivalents. As an exam
ple, consider the system with the block diagram shown in Fig. 10.48 in which each
subsystem is causal. The overall system function, H(s), will be determined by block
diagram reduction. We shall do this by two different methods. Note that because
each subsystem is causal, the overall system is causal so that the RAC of H(s) is to
the right of all its poles.
Fig. 10.46
Fig. 10.47
10.4 SYSTEM BLOCK DIAGRAMS 333
The first method is obtained by noting that the summer in the block diagram is
equivalent to two summers as shown in Fig. 10.49 because, in each diagram,
(10.49)
in which the RAC is to the right of all the poles of HJs) since the system is causal.
In consequence, the block diagram of Fig. 10.410 can be reduced to that shown in
Fig. 10.411. This reduced block diagram is seen to be a feedback system. The
Y(Si
Fig. 10.410
334 INTERCONNECTION OF SYSTEMS
Fig. 10.411
system in the forward path is the tandem connection of two systems, so that its
system function is
as shown in Fig. 10.412. Systemf is causal because systems e and b are causal.
Thus the RAC of Hf(s) is to the right of all its poles, which, from Eq. (10.4lo), are
the poles of Hb(s)and H&). Another way of determining the RAC is to note that the
RAC of Hf(s) is the overlap of the RACs of He(s) and Hb(s). The RACs of systems b
and e are to the right of their respective poles because they are causal. Thus the
overlap of the RACs is to the right of the poles of Hb(s) and He@),which are the
poles of Hf(s).Thus the feedback system of Fig. 10.412 is causal with the system
function
(10.41 1)
and RAC to the right of all its poles. This also is the system function of the original
system of Fig. 10.48 because the block diagram of Fig. 10.412 is equivalent to that
of Fig. 10.48. To obtain the system function in terms of the subsystems of Fig.
10.48, we substitute Eqs. (10.49) and (10.410) in Eq. (10.411) to obtain
(1u.tl.
Fig. 10.412
10.4 SYSTEM BLOCK DIAGRAMS 335
qpp
Fig. 10.413
A second method of reducing the block diagram of Fig. 10.48 to obtain H(s) is
to begin by using the equivalence relation shown in Fig. 10.44 to move the input
pickoff point for system d from the output to the input of system b. The resulting
equivalent block diagram is shown in Fig. 10.413. Observe that the two feedback
paths are in parallel, so that, in accordance with our discussion in Section 10.1A, an
equivalent form of this block diagram is as shown in Fig. 10.414 in which
with the RAC of Hg(s)being to the right of all its poles because system g is causal.
This equivalent block diagram is observed to be the tandem connection of a feed
back system with system 6. The system function of the feedback system, Hk(s), is
( 10.414)
with the RAC being to the right of all its poles because system k is causal. Thus the
system function of the equivalent block diagram is
(10.415)
We now obtain the system function in terms of the subsystems of Fig. 10.48 by
substituting the expression for Hg(s) from Eq. (10.4 13):
( 10.4 16)
Fig. 10.414
336 INTERCONNECTION OF SYSTEMS
with the RAC being to the right of all its poles because the system is causal. This
result is observed to be the same as that obtained previously, Eq. (10.412).
Whatever method of block diagram reduction is used, the same final result must
be obtained. However, the different methods do result in different sets of equivalent
block diagrams. For example, the first method described resulted in the equivalent
block diagrams of Figs. 10.410, 10.41 1, and 10.412. Each of these are equivalent
to the original block diagram of Fig. 10.48. The various equivalent block diagrams
lead to different ways of viewing the original system and thus can result in more
insighthl views of the effect of certain subsystems on the overall system. For
example, the equivalent block diagram of Fig. 10.412 can be used to obtain a
better understanding of the effect of system d on the overall system.
As discussed in the introduction of this text, models are the substance of science
while system theory is the theory of models. Thus, system theory is basic to all
science. A model is used to understand the essence of the phenomenon modeled, and
it is also used to predict certain experimental results. However, approximations are
made to construct the model, so that the predicted experimental results differ from
those obtained with the physical system that was modeled. A smaller approximation
error can be obtained with a more complex model. Thus there is a tradeoff between
model complexity and modeling accuracy. Often, the essence of a phenomenon can
be studied with a rather simple model. An example is the model of echoing
discussed in Chapter 1. Another example is that of the Earth’s orbit about the sun
which can be studied by a model consisting of a stationary spherical sun and an
orbiting spherical earth. From such a model, the basic properties of the Earth’s
elliptical orbit can be determined. However, to determine the Earth’s orbit with
greater accuracy, the model would have to be made more complex by including
the effects of such things as the Earth’s moon, other planetary bodies, and the
nonsphericity of the Earth and sun. The desired accuracy would determine which
to include in the model.
As in the examples above, approximate models are often constructed by idealiz
ing the components of the model. The use of idealized elements in the model often
impose constraints on the model. It is important to recognize these constraints.
Nonsensical results can be obtained if they are not recognized and used. The analysis
of some practical examples illustrating these ideas are presented below.
We begin with a very simple example of the circuit shown in Fig. 10.51. In the
figure, a 4V ideal battery is connected to a 2V ideal battery by a switch, S. What
happens when the switch is closed? The surprising answer is that the question cannot
be asked because the switch cannot be closed! Why? Because, by definition, the
voltage across an ideal battery is fixed irrespective of what is connected across it. If
the switch were closed, we would be saying that 4 = 2! Thus the switch cannot be
closed in this model, and we cannot ask what happens when the switch is closed. Of
10.5 MODEL CONSISTENCY 337
Fig. 10.51
course you clearly could physically connect two real batteries as in the figure and
close the switch. The difference then is that you used real batteries and not ideal
ones. Real batteries have some internal resistance, so that a good model of a real
battery is an ideal battery in series with an ideal resistor. The model of Fig. 10.51
would then be modified by the inclusion of an ideal resistor connected in series with
each ideal battery whose value is equal to the internal resistance of the real battery;
also included would be a resistance with a value equal to the resistance of the wire
connecting the batteries. The switch then can be closed because the voltage differ
ence across the switch is made zero by the voltage drop across these resistors.
A more interesting illustration of the above example is the circuit shown in Fig.
10.52, in which an ideal capacitor with a capacitance of C, and charged to a voltage
VI is connected via a switch, S, to an ideal capacitor with a capacitance of C2 and
charged to a voltage V2. What happens when the switch is closed? As in the
preceding example, the surprising answer is that the question cannot be asked
because the switch in this model also cannot be closed! Why? Because, at the instant
the switch is closed, the model requires that the voltage across the ideal capacitors be
the same. If the switch were closed, we would be saying that, at the instant of
closing, VI = V2;thus the inconsistency is similar to that of our previous example
with the batteries.
The problem is that we have considered all resistance to be zero as in our previous
example. What we really mean by zero resistance is the limit as the value of the
resistance approaches zero, which physically means that the value of the resistance is
exceedingly small. Let us then analyze this circuit by including only the resistance in
the wires connecting the capacitors. Let the value of this resistance be R. The model
then is as shown in Fig. 10.53. To analyze this circuit, let t = 0 be the time at which
the switch S is closed. Also let u , ( t ) and u2(t) be the respective voltages across the
capacitors C, and C,. Because the sum of the voltages about any circuit loop must
be zero, we have the following for t > 0:
) Ri(t) = 0
ul(t)  ~ 2 ( t  (10.51)
Fig. 10.52
338 INTERCONNECTION OF SYSTEMS
Fig. 10.53
1 1
qI(t)  qq2(t)  Ri(t) = 0 (10.52)
C1 c
2
Because current is the rateofchange of charge, we have that i(t) = qi(t) = q’,(t).
Thus, by differentiating Eq. (10.52) and substituting these relations, we obtain the
following for t > 0:
+
Ri’(t)
[d +
 
:2l
i(t) = O
or
(10.53)
This is a differential equation of the type discussed in Section 8.5. For its solution,
we require the initial condition, i(O+). For t > 0, we have from Eq. (10.51)
(10.54a)
Because the voltage across the ideal capacitors at t = O+ must be the same as at the
instant before the switch was closed, we then have from Eq. (10.54a)
i(O+) = VI  v2
~ (10.54b)
R
We now use Laplace transforms as discussed in Section 8.5 to solve Eq. (10.53)
with the initial condition given by Eq. (10.54b).
(10.55)
10.5 MODEL CONSISTENCY 339
i(O+) 1
Z(S) = ~ 1' a> (10.56a)
z
S+
z
(10.56b)
and
(10.58b)
Without evaluating these integrals, note that the sum of the charges on the two
capacitors is
Thus we observe that charge is conserved as it must in any circuit because the sum is
a constant equal to the total initial charge on the capacitors.
Evaluating Eqs. (10.58) with the use of Eq. (10.57), we obtain
and
and
e2t/' dt = 
z Ri2(0+)
(10.5llb)
2
We now substitute the values of z and i(O+) from Eqs. (10.56b) and (10.54b):
Physically, this is the total amount of charge transferred from one capacitor to the
other. Using Eqs. (10.54b) and (10.56b), its specific value is
and
1
Y ( 4 = 44 (10.5 14)
1K
One would obtain this same result by using the feedback equation, Eq. (10.25). For
the special case in which K = 2, we have from this result that for a unitstep input
x(t) = u(t), the response is y(t) = u(t). Although this indeed is a solution of the
system equations of the model, Eqs. (10.514), it is one that the physical system
never exhibits! The reason is that there always is some delay in any physical system
due to the fact that the size, d , of any component of the physical system is greater
than zero. Because no wave can travel faster than the speed of light, c, there must be
a delay of at least d l c seconds between the input and the output of any physical
component. Thus a proper model of any physical system must include some delay.
This delay could be exceedingly small but not zero. For example, if the component
size is about 3 cm, there must be a delay of about 1OW’’ s = 0.1 ns. Thus we must be
careful when modeling a component with zero delay because the solution of the
system model then may not be consistent with the solution of the physical one. The
342 INTERCONNECTION OF SYSTEMS
Fig. 10.54
solution of a system model with zero delay must always be obtained as the limit as
the component delays go to zero or, equivalently, with the component delays being
infinitesimally small.
To understand the effect of a small delay, let us examine the model of Fig. 10.54
in which a small delay, to,is inserted in the feedback path as shown in Fig. 10.55. In
accordance with our analysis in Section 1.6, the response, y(t), of this system for
K = 2 and the input, x(t) = u(t), is a staircase function with the value 2"  1 in the
time interval (n  l)to < t < nto. With this result, we can examine the solution as
the delay time, to, tends to zero. The table below is a list of values of the system
output at t = 1/3 ps for various values of to.
106 113 1 1 .O
107 113 x 10 4 1.5 x I O
108 113 x IO2 34 1.7 x 10"
109 113 x IO3 334 3.5 x 10'00
1010 i/3 104 3334 4.3 101003
The output amplitude at t = 1/3 ps is seen to become arbitrarily large as the delay
time, to, becomes arbitrarily small. In fact, you should note that this is true for any
given value o f t because, for any given value oft, t/to increases as to decreases. Thus
we observe that the proper solution for zero delay is that y(t) is infinite. Generally,
the solution of any system model with zero delay must always be obtained as the
limit as the component delays go to zero or, equivalently, with the component delays
being infinitesimally small.
The examples in this section illustrate the care that must be taken in modeling a
physical system when certain components are idealized or some component values
of the physical system are modeled with zero value. We always must interpret zero as
the limit of the solution as the component value tends to zero or, equivalently, the
>
=
solution obtained by using an infinitesimal value for the component value. It is this

(t to)
Fig. 10.55
10.6 THE STATESPACE APPROACH 343
approach that was used to define the impulse in this text which prevents any para
doxes involving the use of the impulse.
In this text, the basic relation used to express the output, y(t), of an LTI system in
terms of its input, x(t), is the convolution integral
y(t) = h(t)*x(t) =
rm h ( ~ ) x (
t 7) dz
All our analyses of LTI systems derived from this basic relation. A relation of this
(10.61)
(10.62)
m
This integral involves the whole function x(t), so that it is a mapping of the function
x(t) into the number y(to).Because our analyses of LTI systems were developed from
the convolution integral, our formulation of LTI system analysis is referred to as a
functional theory of LTI systems.
As you might expect, there are many different methods of LTI system analysis
that can be used. Each method has advantages for certain types of applications. For
some applications, one method may be used for one part of the problem while
another method may be used for another part of the problem. We have discussed
a number of applications in this text for which the functional theory is very useful
and lends useful insight to the problem. Another method is the statespace method.
This method is especially useful for certain types of control problems such as the
control of a satellite in an orbit.
The statespace method will be described so that you can understand the essence
of the stateVariable theory of systems. However, we shall not discuss the statespace
method in detail because its discussion requires a development of a subject called
linear vector spaces, which is the mathematics required for a visualization and
understanding of the operations performed. There are many texts available which
present a good development of this a p p r ~ a c h . ~
See, for example, ( I ) Chen, ChiTsong. Linear System Theovy and Design, Holt, Rinehart and Winston,
1984; (2) Friedland, Bernard. Conhol System Design, An Introduction to StateSpace Methods, McGraw
Hill, 1986; and ( 3 ) DeCarlo, Raymond. Linear Systems, A State Variable Approach with Numerical
Implementation, PrenticeHall, Englewood Cliffs, NJ, 1989.
344 INTERCONNECTION OF SYSTEMS
Thus the state of an LTI system is the set of the system initial conditions. For
example, the state of an electric circuit can be simply the set of capacitor voltages
and the inductor currents because their values at any given time, t = to,together with
the circuit input for t 3 to, enables one to determine the unique circuit response for
t 3 to.The capacitor voltages and the inductor currents are then called state variables
of the circuit. It is quite immaterial how the capacitor voltages and inductor currents
were attained because that has no bearing on the hture behavior of the circuit. Thus,
the state of a system is a set of variables that completely characterize the effect of the
past history of the system on its hture behavior. Note then that the state variable
method describes only causal systems. Similarly, for a mechanical system, the state
variables are the positions and velocities of its masses. For example, for the state
variable description of a satellite in its orbit, the state variables of the satellite would
be its three coordinates of position in space, its three angular positions (its roll, pitch,
and yaw), and the associated velocities, so that there would be 12 state variables.
The set of variables that qualify as the state of a system is not unique. For
example, in a mechanical system, the positions and momenta of the masses also
would qualify as the state variables of the system. Observe that the system must be a
lumped parameter system for there to be a finite number of state variables. Because
this is the usual case, the systems mostly analyzed by the statespace approach are
lumped parameter causal systems.
To illustrate these concepts, the series RLC circuit shown in Fig. 10.61 will be
analyzed using the state variable approach. As discussed above, the state variables
can be chosen to be the capacitor voltage, u(t), and the inductor current, i(t). All the
circuit variables can be determined in terms of these two state variables. For exam
ple, the voltage across the resistor is Ri(t) and the voltage across the inductor is
e(t)  Ri(t)  u(t). The equations governing the state variables are called the state
equations. For our example, the state equations are
i(t) = Cu’(t) (10.63a)
e(t) = Ri(t) + Li’(t) + u(t) (1 0.63b)
Fig. 10.61
10.6 THE STATESPACE APPROACH 345
Each of these equations involve the first derivative of a state variable. For the state
variable approach, each of these two equations are solved for this first derivative as
1
u'(t) =  i(t) (10.64a)
C
1 R
i'(t) =   u(t)   i(t)
L L
+ L1 e(t) (10.64b)
These two equations are called the state equations of the system. They can be
expressed in the form of a matrix equation as
[ = [ _ _ _' R ]
L L
[ + [ (10.65)
This is the matrix form of the state equation. In general, the state equation for any
lumped parameter LTI system in which there are n state variables andp inputs can be
expressed as a matrix equation in the form
~~
+ Bu(t) (1 0.66)
where x(t)

is a column vector of the n state variables called the state vector.
(10.66a)
In accordance with our discussion of the state of a system, the state vector can be
viewed as a running collection of initial conditions. The symbol A in Eq. (10.66),
called the state transition matrix, is an n by n matrix:
... ...
.. ... (10.66b)
.. ann
Also, u(t) is a column vector of the p inputs called the input vector:
~
(1 0.66~)
346 INTERCONNECTION OF SYSTEMS
Note then that the statespace description easily incorporates the description of
systems with several inputs. The last symbol in Eq. (10.66), B, is an n by p matrix
(10.66d)
Because the state vector has n components, the state of the system at any given
time can be viewed as a point in an ndimensional space. The system behavior then
can be analyzed in terms of the position of this point as a function of time. The
mathematical theory of linear vector spaces is used in this analysis.
For our circuit example above, n = 2 andp = 1 with xI( t ) = u(t), x 2 ( t ) = i(t), and
ul(t) = e(t).The state equation, Eq. (10.66), can be solved for the state vector, x(t)
using matrix methods. Because matrix operations can be performed efficiently OK
computer, a computer can be used to perform the required calculations. This is one
attraction of the state variable approach.
To illustrate these concepts, we shall determine the state vector for our circuit
example. In the statespace method, Eq. (10.65) would be solved using matrix
theory. However, we shall determine e(t) and i(t) by solving Eqs. (10.64). For
this, we can obtain the Laplace transform of the two equations as discussed in
Section 8.5 and solve the resulting two algebraic equations simultaneously. Alter
natively, by differentiating Eqs. (10.64) we can obtain one equation involving only
u ( t ) and a second equation involving only i(t). Using the latter method, we obtain
R 1 1
+ L
+
i”(t)  i’(t) i(t) =  e’(t)
LC L
(10.67b)
These two equations can be solved using Laplace transforms as discussed in Section
8.5. For the case in which we choose to = 0, we obtain
V(s)= (10.6Sa)
 R 1
sL +s+
L LC
I(s) = (10.68b)
s2 + RLs + 1
LC
10.6 THE STATESPACE APPROACH 347
To express the solution only in terms of the state variables, we need to express u'(0)
and i'(0) in terms of u(0) and i(0).4 For this we have from Eqs. (10.64) that
1
d ( O ) =  i(O) (10.69a)
C
1 R 1
1(O) =   u(O)   i(0) +  e(O) (10.69b)
L L L
Substituting these relations into the solution for V(s) and I(s), we then have
V(s)=
E(s)
LC (E)
R 1
c
1
+ s + u(0) + i(O)

( 10.61Oa)
s2 + s +
 
L LC
and
1 1
sE(s) + si(0)  u(0)
I(s) = L L (10.6lob)
R 1
s2+s+
L LC
We obtain explicit expressions for the time fimctions by first completing the
square to express the denominator polynomial in the form
R
D(s) = s2 +s
L
+ LC1 = (s + a )2 + 0 02
 (10.6lla)
where
R
(10.6 11b)
For our illustration, we shall assume a > 0 and mi > 0. Then, Eqs. (10.610) can be
expressed as
au(0) + 1 i(0)

V ( s ) =  1 +
as) s+ +
c wo (10.612a)
LC D(s) D(s) 0 0 Do
4This step would not be required if the Laplace transform of Eqs. (10.64) were obtained and the resulting
two algebraic equations were solved simultaneously.
348 INTERCONNECTION OF SYSTEMS
and
(10.612b)
The reason for expressing the equations in this form is that we then can use entries
no. 4 in Table 7.41 directly without having to go through the route of obtaining
partial fraction expansions. Thus we obtain from entries no. 4 of Table 7.41 that the
initial condition response, the solution for e(t) = 0, so that E(s) = 0, is
+ 1 i(0)
and
au(0)
WO
C
1
sin(o,t) eatu(t> (10.613a)
1
+
ai(0)  u(0)
L
0 0
1
sin(oOt) e"u(t) (10.613b)
In accordance with our discussion, the state of the circuit at any given time can be
considered to be a point in a twodimensional space with coordinates u(t) and i(t). To
illustrate a graph of the position of this point as a function oft, we consider the case
for which R = 10 0,L = lop3H, and C = lop6F. With these values, a = 5 x lo3
and wO = 31.225 x lo3, and thus for t 2 0 we obtain
( 10.614a)
and
A graph of the position of this point as a function of t for the case in which
u(0) = 10 and $0) = 2 is shown in Fig. 10.62.
As t increases, the circuit state moves in a spiral from the point (10, 2) to the
point (0,O). This is an example of asymptotic stability mentioned in Section 3.6
because, with zero input, the system state approaches the origin asymptotically. Of
course, the motion of the point would be entirely different if the input were not zero.
10.6 THE STATESPACE APPROACH 349
20
40
Fig. 10.62 Graph of v(t) versus i(t). (Arrows point in the direction of increasing t ) .
For the general case in which there are n states, the state of a system given by
Eq. (10.66a) can be considered to be a point moving in an ndimensional space
with coordinates x,,x2,. . . , x,. In this manner, the dynamic behavior of a system
can be visualized in terms of a point moving in an ndimensional statespace. The
mathematical theory used for this study is called linear vector spaces, which makes
extensive use of matrix theory. With this formulation, the study of a dynamic system
can be viewed in terms of a study of its associated state space. Thus, the system is
asymptotically stable if, with zero input, the system state approaches the origin
asymptotically for any initial state. It is important to realize that it is possible for
a system to be asymptotically stable but not be BIBOstable.
Control is a major application of statespace theory. The statespace theory of
control can be visualized in terms of the control of the state of a satellite in its orbit.
We saw above that there are 12 state variables for this system. “To control the
satellite” means to change its six position variables and six velocity variables
from one set of values to another. The control is accomplished by some jets attached
to the satellite. The forces exerted by these jets are the system inputs and constitute
the input vector. In statespace terms, this can be viewed as the input vector moving
the state of the system from one point in the state space to another. To control the
satellite, the input vector must be able to move it from any given point in the state
space to any other desired point in the state space within a finite amount of time. If
this can be accomplished, the satellite is said to be totally controllable. If only some
of the state variables can be controlled, then the satellite is only partially controllable.
In statespace terms, the controllable state variables are the coordinates of a subspace
of the state space. This subspace is called the controllable subspace. A controller
then can be designed to control the system in this subspace.
However, to control the state of the satellite, we first must know its present state.
The reason for this is that if we do not know where the satellite is in the state space,
how can we determine the path in the state space that should be taken to bring the
satellite to the desired point in the state space? That is, how can I determine the
350 INTERCONNECTION OF SYSTEMS
direction I should walk to go home if I don't know where I presently am located? For
this, we must observe some of the state variables of the system. For example, could
we determine the values of the satellite state variables if we just observe its position
in space? Clearly not, because we would not have sufficient information to deter
mine, for example, its angular coordinates. The vector of observed variables is called
the output vector. If knowledge of the output vector and the input vector is sufficient
to exactly determine the values of all the system state variables, the system is said to
be totally observable. If the values of only some of the state variables can be
determined, then the satellite is only partially observable. In statespace terms, the
state variables that can be observed are the coordinates of a subspace of the state
space. This subspace is called the observable subspace. An observer then can be
designed to observe the system in this subspace.
Meaningful control can be accomplished only for those state variables that are
both observable and controllable. These state variables lie in the intersection (or the
overlap) of the controllable and observable subspaces. Thus, one problem in the
statespace approach is to determine the intersection of the controllable and obser
vable subspaces. An observer and a controller for that subspace is then designed.
This is the essence of the state approach to control. As indicated above, a theo
retical advantage of viewing a system as a moving point in an ndimensional state
space lends a great deal of insight into the control problem. A practical advantage of
the statespace approach is that the theory easily incorporates several inputs and
several outputs. Also, because the mathematical operations are matrix ones, the
required calculations can be performed efficiently on a computer.
In control theory, the problem of determining an observer and a controller is often
complicated by the imposition of additional design criteria. One common criterion is
that only certain paths are allowed in transferring the system from one point to
another in the state space. For our satellite example, there would be a constraint
on the allowable acceleration in order to limit the forces on the satellite. Another
complication is the everpresent problem of noise. For this, a design is determined
for which the effect of the noise is minimized. Additionally, the system parameters
oRen are not known exactly. For example, the exact satellite mass and the exact
thrust of the control jets may not be known. For this, robust control theory is used. In
robust control, the control is designed to be insensitive to the slight errors in the
values of the system parameters used.
PROBLEMS
Y U
Prob. 10.1
4
HJS) =
s+l'
cs > 0 and Hb(s) = KO +,s Kl
+2
cs > 0
(a) Determine the values of KOand K , required for the poles of the feedback
system to be at p , = 4 +jO and p 2 = 12 +jO.
(b) Determine the location of the poles and zeros of Hb(s) for the values
determined in part a.
Prob. 10.2
103 Consider the model of an echoing system shown in Fig. 1.63 with K > 0.
(a) Determine the system function, H(s), of the feedback system.
(b) Determine the pole locations. How many poles are there?
(c) For what values of K is the system stable?
(d) Let K = 0.9. For low frequencies, show that the system can be modeled
as a bandpass filter and determine its center frequency, 3dB bandwith,
and Q.
105 Determine whether any of the roots of s5 + 2s4 + 2s3 + 4s2 + s + 1 lie in the
right half of the s plane (RHP).
A
H(s) =
s3 + + +K
4s2 4s
107 The component systems of the the system below are causal LTI systems.
(a) Determine the system function, H(s), of the system with the input x(t)
and output y(t).
@) How is the RAC for H(s) determined? Your reason must be given.
Prob. 10.7
108 The component systems of the the system below are causal LTI systems with
the system functions H,(s), Hb(s),H,(s), Hd(s),and H,(s).
(a) Determine the system function, H(s), of the system with the input x(t)
and output y(t).
VU)
A C
Prob. 10.8
109 The consistency problems associated with models discussed in Section 10.5
resulted from idealizations made without a concomitant analysis of their
implications relative to the questions to be asked of the model. In fact, we saw
that certain questions, although grammatically meaningful, are logically
inconsistent and thus could not be asked. I call these “meaningless ques
tions.” Analyze the following two questions to determine whether they are
meaningless and, if so, why.
(a) What happens if the irresistible force meets the immovable object?
(b) Is there a sound generated if a tree falls in a forest and no one hears it?
A PRIMER ON COMPLEX NUMBERS
AND ALGEBRA
A.l INTRODUCTION
+
A complex number is a quantity z = x j y in which x and y are real numbers and
j = a. The number x is called the real part of z, and the real number y is called
the imaginary part of z. We often express this as x = Re(z} and y = Im(z}. It is
important to note that Im(z) is a real number. Observe that z is a real number if
y = 0. Consequently, real numbers are special cases of complex numbers in which
the imaginary part is equal to zero. If x = 0, then the complex number zis said to be
an imaginary number. Two complex numbers z1 and z2 are defined to be equal only if
Re(zl}= Re(z,} and also Im(z,} = Im(z2).
Complex numbers were developed because many polynomial equations of the
form
X" + anlX"l + . . . + a l x + a. = o (A 1)
do not have solutions if the solutions are restricted to being real numbers. The
+
simplest example is the quadratic equation x2 1 = 0. This equation does not
have a real solution. However, if complex numbers are allowed, then the equation
+
z2 1 = 0 has two solutions, z =j and z = j. Complex numbers and complex
algebra have had a long development by many individuals over the centuries. With
their use, it turns out that a polynomial equation of the form
9 + c,19l + . . . + + co = 0
CIZ ('42)
in which the coefficients co,cl, . . . , cnP1are complex numbers always has a solu
tion. This surprising result is called the fundamental theorem of algebra.' Note that
' This theorem was first proved by Gauss in 1799 as part of his doctoral thesis
353
354 A PRIMER ON COMPLEX NUMBERS AND ALGEBRA
this theorem immediately implies that Eq. (A2) has exactly n solutions because, by
the theorem, it must have at least one solution. Let the solution be z = zl. This
solution can be factored out as
'Named for the French mathematician J. R. Argand, who published an essay on the geometric
representation of complex numbers in 1806. However, Gauss had already discussed this representation
in 1799, and the Norwegian mathematician Casper Wessel published a discussion of it in 1797.
A.2 THE COMPLEX PLANE 355
so that the complex number z = x +jy also can be expressed in the trigonometric
form
and with the use of the Pythagorean theorem for right triangles we obtain
0 ifx>O
r = 1x1 and 8= (A8)
71 ifx<O
Two complex numbers that differ only in the sign of their imaginary parts are said
to be conjugates. Thus the conjugate of z = x +jy is z* = x jy. The conjugate of z
is denoted by z*. From Eqs. (A6) and (A7), we have
Note that the conjugate of any expression involving complex numbers can be
obtained by replacing every j in the expression with j. Observe that z is a real
number if and only if z = z*.
356 A PRIMER ON COMPLEX NUMBERS AND ALGEBRA
O0 u" u2 u3 u4
eu=&= l+u++++... (A10)
n=O 2! 3! 4!
which converges for all values of u. For the case in which u =j8, we obtain
in which use was made of the fact that j2=  1, j3=j2j= j, j4=j2j2 = 1, and so
on. The series in the fist parentheses is the power series expansion of cos 0, and the
series in the second parentheses is the power series expansion of sine, so that we
have obtained the important Euler formula3
Thus eJ8 and e'' are conjugates. It immediately follows from Eqs. (A12) and
(A 13) that
1 .
cos e = [el8
2
+ (A 14)
and
The algebraic operations with complex numbers are defined with the same rules used
for real numbers. This ensures that the values obtained with any algebraic operation
with a complex number z results, in the special case z = x +jO, in the same value
that would be obtained when using the real number x. Thus, if zI = x, jy, and +
+
z2 = x2 jy2, the sum is defined as
1
Re(z, 1 = [zI + zT] (A 19)
and
1
Im(z,) = [zl  zT] (A20)
2j
0 . (zl + z2). Now, because the length of any side of a triangle must be equal to or
less than the sum of the lengths of its other two sides, we have from our geometric
view of addition that
This inequality is called the triangle inequazity. This can be extended to the sum of
three complex numbers as follows:
This can be extended to the sum of n terms by continuing as above to obtain the
triangle inequality for the sum of n terms:
I n I n
(A23)
(A24)
which is obtained by using the fact that j2= 1. A nicer form for the product is
obtained by using the polar form for the complex numbers:
(A25)
and
That is, the product of the magnitude of two complex numbers is equal to the
product of their magnitudes, and the angle of the product of two complex numbers
A.4 COMPLEX ALGEBRA 359
is equal to the sum of their angles. Note that for the special case in which z2 = z?,
we obtain
Thus the square of the magnitude of any complex number can be obtained by
multiplying that complex number by its conjugate. Remember that the conjugate
of any expression involving complex numbers is easily obtained by replacing every j
in that expression by j.
Another case of interest is the case in which z2 =j . For this case it is easily seen
from its complex plane representation, Fig. A1, that Ij ( = 1 and L j = n/2. Thus its
polar form representation is
Consequently,
(A30)
(A3 1)
This can be expressed in rectangular form by multiplying the numerator and denomi
nator by the conjugate of z2 to obtain
(A32)
A nicer form for the ratio of two complex numbers is obtained by expressing them in
polar form:
(A34)
360 A PRIMER ON COMPLEX NUMBERS AND ALGEBRA
and
That is, the magnitude of the ratio of two complex numbers is equal to the ratio of
their magnitudes, and the angle of the ratio of two complex numbers is equal to the
angle of the numerator minus the angle of the denominator.
Let n be a positive integer. Then, as in the case of real numbers, z" is the nth power
of z. That is, 9 is the product of z by itself n times. Thus
(A36)
and
The polar form of z" results in some important relations. In polar form,
1
31= lzl" and Lz" = n(Lz) (A39)
Also by expressing Eq. (A38) in trigonometric form, we obtain for the special case
in which Y = 1
+ e
cos2 8  sin2 6 j 2 cos 8 sin = cos 28 +j sin 28 (A41)
4Named for the French mathematician, Abraham DeMoivre (16671754). An equivalent form had been
obtained earlier by the English mathematician Roger Cotes (16821716).
A.6 ROOTS OF COMPLEX NUMBERS 361
A single equation such as this is, in reality, a pair of equations because two complex
numbers are equal only if their real parts are equal and also their imaginary parts are
equal. We thus have from Eq. (A41)
These are two halfangle formulas that are contained in most trigonometric tables. In
this manner we can obtain, for each value of n , an expression for cosn8 and for
sin n0 in terms of powers of only sin 0 and cos 8.
For the same algebraic reasons as in the case of real numbers, we define
z0 = 1 (A43)
Let n be a positive integer; then the root, zlln, is a number that, raised to the nth
power, is equal to the number z. The polar form will be used for this determination.
First observe that if k is an integer then
Consequently,
(A45)
where k is an integer. With this form of expression for the complex number z, we
have

 [cos (O+n2n,> +jsin (O+n2nk)]
~ ~ (A48)
(A49)
for k = 0, 1, . . . ,(n  1). These values are the n distinct solutions of the equation
z" = 1 for which there must be n distinct solutions in accordance with the funda
mental theorem of algebra discussed in Section A . l . From the polar form given by
Eq. (A47), we note that, in the complex plane, the n roots of z are equally spaced on
a circle of radius r'l". The spacing is 2n/n rad (360"ln) with the root for k = 0 at an
angle of e / n rad (for e expressed in radians).
With the use of Eq. (A48), the rational power of a complex number now can be
defined as
(A50)
Let y(t) be the response of an LTI system with the input x(t), unitimpulse response
h(t), and system function H(s). As discussed in Section 5.9, the total energy ofy(t) is
E= 100
m
k(t)l2dt
We assume Iy(t)l < 00 and that E, the total energy ofy(t), is finite. Note that this
implies that
lim y(t) = 0
trfcc
(B2)
0331
The partial energy, E(T), is seen to be the energy of y(t) up to the time t = T .
Now let there be a zero of H ( s ) at s = z so that we can express H(s) as
H ( s ) = (S  z)G(s) 034)
In this appendix, we determine the effect of moving the zero parallel to the cr axis
upon the partial energy of the system output.
The Laplace transform of y(t) is
The RAC of y(t) lies in the overlap of the RACs of h(t) and x(t). For convenience,
define
Observe that
lim u(t) = 0
ttfcc
as a consequence of Eqs. (B2) and (B8). Using Eq. (B8), the square of the
magnitude of y(t) is
lY(t)l2 = Y(tlY*(t)
= Id(t)I2 + lz121~(t)12 z[~(t)][~’(t)]*
 z*[~*(t)[d(t)] (B10)
= lu’(t)I2 + I Z I ~ ~ U (2Re{z[u(t)][u’(t)]*)
~)~~
With the zero at z = z l , call the output y l ( t ) ;and with the zero at z = z,, call the
output y2(t).We shall determine the