Sundaram ECE301 Notes PDF

ECE 301: Signals and Systems
Course Notes
Prof. Shreyas Sundaram
School of Electrical and Computer Engineering

Purdue University
ii
Acknowledgments
These notes very closely follow the book: Signals and Systems, 2nd edition, by
Alan V. Oppenheim, Alan S. Willsky with S. Hamid Nawab. Parts of the notes
are also drawn from
• Linear Systems and Signals by B. P. Lathi

• A Course in Digital Signal Processing by Boaz Porat
• Calculus for Engineers by Donald Trim
I claim credit for all typos and mistakes in the notes.

The LATEX template for The Not So Short Introduction to LATEX 2ε by T. Oetiker
et al. was used to typeset portions of these notes.
Shreyas Sundaram
Purdue University
iv
Contents
1 Introduction 1
1.1 Signals and Systems . . . . . . . . . . . . . . . . . . . . . . . . . 1
1.2 Outline of This Course . . . . . . . . . . . . . . . . . . . . . . . . 4
2 Properties of Signals and Systems 5

2.1 Signal Energy and Power . . . . . . . . . . . . . . . . . . . . . . 5
2.2 Transformations of Signals . . . . . . . . . . . . . . . . . . . . . . 7
2.3 Periodic, Even and Odd Signals . . . . . . . . . . . . . . . . . . . 7
2.4 Exponential and Sinusoidal Signals . . . . . . . . . . . . . . . . . 8
2.4.1 Continuous-Time Complex Exponential Signals . . . . . . 8
2.4.2 Discrete-Time Complex Exponential Signals . . . . . . . . 9
2.5 Impulse and Step Functions . . . . . . . . . . . . . . . . . . . . . 12
2.5.1 Discrete-Time . . . . . . . . . . . . . . . . . . . . . . . . . 12
2.5.2 Continuous-Time . . . . . . . . . . . . . . . . . . . . . . . 13
2.6 Properties of Systems . . . . . . . . . . . . . . . . . . . . . . . . 14
2.6.1 Interconnections of Systems . . . . . . . . . . . . . . . . . 14
2.6.2 Properties of Systems . . . . . . . . . . . . . . . . . . . . 15
3 Analysis of Linear Time-Invariant Systems 21

3.1 Discrete-Time LTI Systems . . . . . . . . . . . . . . . . . . . . . 21
3.2 Continuous-Time LTI Systems . . . . . . . . . . . . . . . . . . . 23
3.3 Properties of Linear Time-Invariant Systems . . . . . . . . . . . . 24
3.3.1 The Commutative Property . . . . . . . . . . . . . . . . . 24
3.3.2 The Distributive Property . . . . . . . . . . . . . . . . . . 24
3.3.3 The Associative Property . . . . . . . . . . . . . . . . . . 25
vi CONTENTS
3.3.4 Memoryless LTI Systems . . . . . . . . . . . . . . . . . . 26

3.3.5 Invertibility of LTI Systems . . . . . . . . . . . . . . . . . 27
3.3.6 Causality of LTI Systems . . . . . . . . . . . . . . . . . . 28
3.3.7 Stability of LTI Systems . . . . . . . . . . . . . . . . . . . 28
3.3.8 Step Response of LTI Systems . . . . . . . . . . . . . . . 29
3.4 Differential and Difference Equation Models for Causal LTI Systems 30
3.4.1 Linear Constant-Coefficient Differential Equations . . . . 31
3.4.2 Linear Constant Coefficient Difference Equations . . . . . 33
3.5 Block Diagram Representations of Linear Differential and Differ-
ence Equations . . . . . . . . . . . . . . . . . . . . . . . . . . . . 35
4 Fourier Series Representation of Periodic Signals 37

4.1 Applying Complex Exponentials to LTI Systems . . . . . . . . . 37
4.2 Fourier Series Representation of Continuous-Time Periodic Signals 40
4.3 Calculating the Fourier Series Coefficients . . . . . . . . . . . . . 42
4.3.1 A Vector Analogy for the Fourier Series . . . . . . . . . . 44
4.4 Properties of Continuous-Time Fourier Series . . . . . . . . . . . 48
4.4.1 Linearity . . . . . . . . . . . . . . . . . . . . . . . . . . . 49
4.4.2 Time Shifting . . . . . . . . . . . . . . . . . . . . . . . . . 49
4.4.3 Time Reversal . . . . . . . . . . . . . . . . . . . . . . . . 50
4.4.4 Time Scaling . . . . . . . . . . . . . . . . . . . . . . . . . 50
4.4.5 Multiplication . . . . . . . . . . . . . . . . . . . . . . . . . 51
4.4.6 Parseval’s Theorem . . . . . . . . . . . . . . . . . . . . . . 52
4.5 Fourier Series for Discrete-Time Periodic Signals . . . . . . . . . 52
4.5.1 Finding the Discrete-Time Fourier Series Coefficients . . . 53
4.5.2 Properties of the Discrete-Time Fourier Series . . . . . . . 55
5 The Continuous-Time Fourier Transform 59

5.1 The Fourier Transform . . . . . . . . . . . . . . . . . . . . . . . . 59
5.1.1 Existence of Fourier Transform . . . . . . . . . . . . . . . 62
5.2 Fourier Transform of Periodic Signals . . . . . . . . . . . . . . . . 63
5.3 Properties of the Continuous-Time Fourier Transform . . . . . . 64
5.3.1 Linearity . . . . . . . . . . . . . . . . . . . . . . . . . . . 65
5.3.2 Time-Shifting . . . . . . . . . . . . . . . . . . . . . . . . . 65
CONTENTS vii
5.3.3 Conjugation . . . . . . . . . . . . . . . . . . . . . . . . . . 65
5.3.4 Differentiation . . . . . . . . . . . . . . . . . . . . . . . . 66
5.3.5 Time and Frequency Scaling . . . . . . . . . . . . . . . . 67
5.3.6 Duality . . . . . . . . . . . . . . . . . . . . . . . . . . . . 67
5.3.8 Convolution . . . . . . . . . . . . . . . . . . . . . . . . . . 68
5.3.9 Multiplication . . . . . . . . . . . . . . . . . . . . . . . . . 72
6 The Discrete-Time Fourier Transform 75

6.1 The Discrete-Time Fourier Transform . . . . . . . . . . . . . . . 75
6.2 The Fourier Transform of Discrete-Time Periodic Signals . . . . . 78
6.3 Properties of the Discrete-Time Fourier Transform . . . . . . . . 79
6.3.1 Periodicity . . . . . . . . . . . . . . . . . . . . . . . . . . 79
6.3.2 Linearity . . . . . . . . . . . . . . . . . . . . . . . . . . . 79
6.3.3 Time and Frequency Shifting . . . . . . . . . . . . . . . . 79
6.3.4 First Order Differences . . . . . . . . . . . . . . . . . . . . 80
6.3.5 Conjugation . . . . . . . . . . . . . . . . . . . . . . . . . . 80
6.3.6 Time-Reversal . . . . . . . . . . . . . . . . . . . . . . . . 80
6.3.7 Time Expansion . . . . . . . . . . . . . . . . . . . . . . . 81
6.3.8 Differentiation in Frequency . . . . . . . . . . . . . . . . . 82
6.3.10 Convolution . . . . . . . . . . . . . . . . . . . . . . . . . . 82
6.3.11 Multiplication . . . . . . . . . . . . . . . . . . . . . . . . . 83
7 Sampling 87
7.1 The Sampling Theorem . . . . . . . . . . . . . . . . . . . . . . . 87
7.2 Reconstruction of a Signal From Its Samples . . . . . . . . . . . 89
7.2.1 Zero-Order Hold . . . . . . . . . . . . . . . . . . . . . . . 89
7.2.2 First-Order Hold . . . . . . . . . . . . . . . . . . . . . . . 90
7.3 Undersampling and Aliasing . . . . . . . . . . . . . . . . . . . . . 90
7.4 Discrete-Time Processing of Continuous-Time Signals . . . . . . 91
viii CONTENTS
8 The Laplace Transform 95

8.1 The Laplace Transform . . . . . . . . . . . . . . . . . . . . . . . 95
8.2 The Region of Convergence . . . . . . . . . . . . . . . . . . . . . 98
8.3 The Inverse Laplace Transform . . . . . . . . . . . . . . . . . . . 101
8.3.1 Partial Fraction Expansion . . . . . . . . . . . . . . . . . 102
8.4 Some Properties of the Laplace Transform . . . . . . . . . . . . . 103
8.4.1 Convolution . . . . . . . . . . . . . . . . . . . . . . . . . . 104
8.4.2 Differentiation . . . . . . . . . . . . . . . . . . . . . . . . 104
8.4.3 Integration . . . . . . . . . . . . . . . . . . . . . . . . . . 104
8.5 Finding the Ouput of an LTI System via Laplace Transforms . . 105
8.6 Finding the Impulse Response of a Differential Equation via Laplace
Transforms . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 106
Chapter 1
Introduction
1.1 Signals and Systems

Loosely speaking, signals represent information or data about some phenomenon
of interest. This is a very broad definition, and accordingly, signals can be found
in every aspect of the world around us.
For the purposes of this course, a system is an abstract object that accepts input
signals and produces output signals in response.
Input Output
System
Figure 1.1: An abstract representation of a system.
Examples of systems and associated signals:
• Electrical circuits: voltages, currents, temperature,...

• Mechanical systems: speeds, displacement, pressure, temperature, vol-
ume, ...
• Chemical and biological systems: concentrations of cells and reactants,
neuronal activity, cardiac signals, ...
• Environmental systems: chemical composition of atmosphere, wind pat-
terns, surface and atmospheric temperatures, pollution levels, ...
• Economic systems: stock prices, unemployment rate, tax rate, interest
rate, GDP, ...
• Social systems: opinions, gossip, online sentiment, political polls,...
• Audio/visual systems: music, speech recordings, images, video, ...
2 Introduction
• Computer systems: Internet traffic, user input, ...
From a mathematical perspective, signals can be regarded as functions of one

or more independent variables. For example, the voltage across a capacitor in
an electrical circuit is a function of time. A static monochromatic image can
be viewed as a function of two variables: an x-coordinate and a y-coordinate,
where the value of the function indicates the brightness of the pixel at that
(x, y) coordinate. A video is a sequence of images, and thus can be viewed
as a function of three variables: an x-coordinate, a y-coordinate and a time-
instant. Chemical concentrations in the earth’s atmosphere can also be viewed
as functions of space and time.
In this course, we will primarily be focusing on signals that are functions of a
single independent variable (typically taken to be time). Based on the examples
above, we see that this class of signals can be further decomposed into two
subclasses:
• A continuous-time signal is a function of the form f (t), where t ranges

over all real numbers (i.e., t ∈ R).
• A discrete-time signal is a function of the form f [n], where n takes on only
a discrete set of values (e.g., n ∈ Z).
Note that we use square brackets to denote discrete-time signals, and round
brackets to denote continuous-time signals. Examples of continuous-time sig-
nals often include physical quantities, such as electrical currents, atmospheric
concentrations and phenomena, vehicle movements, etc. Examples of discrete-
time signals include the closing prices of stocks at the end of each day, population
demographics as measured by census studies, and the sequence of frames in a
digital video. One can obtain discrete-time signals by sampling continuous-time
signals (i.e., by selecting only the values of the continuous-time signal at certain
intervals).
Just as with signals, we can consider continuous-time systems and discrete-
time systems. Examples of the former include atmospheric, physical, electrical
and biological systems, where the quantities of interest change continuously over
time. Examples of discrete-time systems include communication and computing
systems, where transmissions or operations are performed in scheduled time-
slots. With the advent of ubiquitous sensors and computing technology, the
last few decades have seen a move towards hybrid systems consisting of both
continuous-time and discrete-time subsystems – for example, digital controllers
and actuators interacting with physical processes and infrastructure. We will
not delve into such hybrid systems in this course, but will instead focus on
systems that are entirely either in the continuous-time or discrete-time domain.
The term dynamical system loosely refers to any system that has an internal
state and some dynamics (i.e., a rule specifying how the state evolves in time).
1.1 Signals and Systems 3
This description applies to a very large class of systems, including individual ve-
hicles, biological, economic and social systems, industrial manufacturing plants,
electrical power grid, the state of a computer system, etc. The presence of dy-
namics implies that the behavior of the system cannot be entirely arbitrary; the
temporal behavior of the system’s state and outputs can be predicted to some
extent by an appropriate model of the system.
Example 1.1. Consider a simple model of a car in motion. Let the speed of
the car at any time t be given by v(t). One of the inputs to the system is the
acceleration a(t), applied by the throttle. From basic physics, the evolution of
the speed is given by
dv
= a(t). (1.1)
dt
The quantity v(t) is the state of the system, and equation (1.1) specifies the
dynamics. There is a speedometer on the car, which is a sensor that measures
the speed. The value provided by the sensor is denoted by s(t) = v(t), and this
is taken to be the output of the system.
Much of scientific and engineering endeavor relies on gathering, manipulating
and understanding signals and systems across various domains. For example,
in communication systems, the signal represents voice or data that must be
transmitted from one location to another. These information signals are often
corrupted en route by other noise signals, and thus the received signal must
be processed in order to recover the original transmission. Similarly, social,
physical and economic signals are of great value in trying to predict the current
and future state of the underlying systems. The field of signal processing studies
how to take given signals and extract desirable features from them, often via
the design of systems known as filters. The field of control systems focuses
on designing certain systems (known as controllers) that measure the signals
coming from a given system and apply other input signals in order to make
the given system behave in an desirable manner. Typically, this is done via a
feedback loop of the form
Desired Control
Output Input Output
Controller System
−
Sensor
Figure 1.2: Block Diagram of a feedback control system.
Example 1.2 (Inverted Pendulum). Suppose we try to balance a stick vertically

in the palm of our hand. The sensor, controller and actuator in this example
are our eyes, our brain, and our hand, respectively, which communicate using
signals of various forms. This is an example of a feedback control system.
4 Introduction
1.2 Outline of This Course

Since the concepts of signals and systems are prevalent across a wide variety of
domains, we will not attempt to discuss each specific application in this course.
Instead, we will deal with the underlying mathematical theory, analysis, and
design of signals and systems. In this sense, it will be more mathematical than
other engineering courses, but will be different from other math courses in that
it will pull together various branches of mathematics for a particular purpose
(i.e., to understand the nature of signals and systems).
The main components of this course will be as follows.
• Signal and systems classifications: develop terminology and identify useful

properties of signals and systems
• Time domain analysis of LTI systems: understand how the output of linear
time-invariant systems is related to the input
• Frequency domain analysis techniques and signal transformations (Fourier,
Laplace, z-transforms): study methods to study signals and systems from
a frequency domain perspective, gaining new ways to understand their
behavior
• Sampling and Quantization: study ways to convert continuous-time sig-
nals into discrete-time signals, along with associated challenges
The material in this course will lay the foundations for future courses in control
theory (ECE 382, ECE 483), communication systems (ECE 440) and signal
processing (ECE438, 445).
Chapter 2
Properties of Signals and

Systems
We will now identify certain useful properties and classes of signals and systems.
Recall that a continuous-time signal is denoted by f (t) (i.e., a function of the
real-valued variable t) and a discrete-time signal is denoted by f [n] (i.e., a
function of the integer-valued variable n). When drawing discrete-time signals,
we will use a sequence of dots to indicate the discrete nature of the time variable.
2.1 Signal Energy and Power

Suppose that we consider a resistor in an electrical circuit, and let v(t) denote
the voltage signal across the resistor i(t) denote the current. From Ohm’s law,
we know that v(t) = i(t)R, where R is the resistance. The power dissipated by
the resistor is then
v 2 (t)
p(t) = v(t)i(t) = i2 (t)R = .
R
Thus the power is a scaled multiple of the square of the voltage and current
signals.
Since the energy expended over a time-interval [t1 , t2 ] is given by the integral of
the power dissipated per-unit-time over that interval, we have
Z t2 Z t2
1 t2 2
Z
E= p(t)dt = R i2 (t)dt = v (t)dt.
t1 t1 R t1
The average power dissipated over the time-interval [t1 , t2 ] is then

Z t2 Z t2
1 1 1 R
E= v 2 (t)dt = i2 (t)dt.
t2 − t1 t2 − t1 R t1 t2 − t1 t1
6 Properties of Signals and Systems
We will find it useful to discuss the energy and average power of any continuous-
time or discrete-time signal. In particular, the energy of a general (potentially
complex-valued) continuous-time signal f (t) over a time-interval [t1 , t2 ] is de-
fined as Z t2
E[t1 ,t2 ] , |f (t)|2 dt,
t1
where |f (t)| denotes the magnitude of the signal at time t.
Similarly, the energy of a general (potentially complex-valued) discrete-time
signal f [n] over a time-interval [n1 , n2 ] is defined as
n2
X
E[n1 ,n2 ] , |f [n]|2 .
n=n1
Note that we are defining the energy of an arbitrary signal in the above way;
this will end up being a convenient way to measure the “size” of a signal, and
may not actually correspond to any physical notion of energy.
We will also often be interested in measuring the energy of a given signal over
all time. In this case, we define
Z ∞
E∞ , |f (t)|2 dt
−∞
for continuous-time signals, and

∞
X
E∞ , |f [n]|2
n=∞
for discrete-time signals. Note that the quantity E∞ may not be finite.
Similarly, we define the average power of a continuous-time signal as
Z T
1
P∞ , lim |f (t)|2 dt,
T →∞ 2T −T
and for a discrete-time signal as

N
1 X
P∞ , lim |f [n]|2 .
N →∞ 2N + 1
n=−N
Based on the above definitions, we have three classes of signals: finite energy
(E∞ < ∞), finite average power (P∞ < ∞), and those that have neither finite
energy nor average power. An example of the first class is the signal f (t) = e−t
for t ≥ 0 and f (t) = 0 for t < 0. An example of the second class is f (t) = 1 for
all t ∈ R. An example of the third class is f (t) = t for t ≥ 0. Note that any
signal that has finite energy will also have finite average power, since
E∞
P∞ = lim =0
T →∞ 2T
for continuous-time signals with finite energy, with an analogous characteriza-
tion for discrete-time signals.
2.2 Transformations of Signals 7
2.2 Transformations of Signals

Throughout the course, we will be interested in manipulating and transforming
signals into other forms. Here, we start by considering some very simple trans-
formations involving the time variable. For the purposes of introducing these
transformations, we will consider a continuous-time signal f (t) and a discrete-
time signal f [n].
Time-shifting: Suppose we define another signal g(t) = f (t−t0 ), where t0 ∈ R.
In other words, for every t ∈ R, the value of the signal g(t) at time t is the value
of the signal f (t) at time t − t0 . If t0 > 0, then g(t) is a “forward-shifted” (or
time-delayed) version of f (t), and if t0 < 0, then g(t) is a time-advanced version
of f (t) (i.e., the features in f (t) appear earlier in time in g(t)). Similarly, for a
discrete-time signal f [n], one can define the time-shifted signal f [n − n0 ], where
n0 is some integer.
Time-reversal: Consider the signal g(t) = f (−t). This represents a reversal
of the function f (t) in time. Similarly, f [−n] represents a time-reversed version
of the signal f [n].
Time-scaling: Define the signal g(t) = f (αt), where α is some real number.
When 0 < α < 1, this represents a stretching of f (t), and when α > 1, this
represents a compression of f (t). If α < 0, we get a time-reversed and stretched
(or compressed) version of f (t). Analogous definitions hold for the discrete-time
signal f [n].
The operations above can be combined to define signals of the form g(t) =
f (αt + β), where α and β are real numbers. To draw the signal g(t), we should
first apply the time-shift by β to f (t) and then apply the scaling α. To see why,
define h(t) = f (t + β), and g(t) = h(αt). Thus, we have g(t) = f (αt + β) as
required. If we applied the operations in the other order, we would first get the
signal h(t) = f (αt), and then g(t) = h(t + β) = f (α(t + β)) = f (αt + αβ). In
other words, the shift would be by αβ rather than β.
Examples of these operations can be found in the textbook (OW), such as
example 1.1.
2.3 Periodic, Even and Odd Signals

A continuous-time signal f (t) is said to be periodic with period T if f (t) =
f (t + T ) for all t ∈ R. Similarly, a discrete-time signal f [n] is periodic with
period N if f [n] = f [n + N ] for all n ∈ Z. The fundamental period of a signal
is the smallest period for which the signal is periodic.
A signal is even if f (t) = f (−t) for all t ∈ R (in continuous-time), or f [n] =
f [−n] for all n ∈ Z (in discrete-time). A signal is odd if f (t) = −f (−t) for all
t ∈ R, or f [n] = −f [−n] for all n ∈ Z. Note that if a signal is odd, it must
necessarily be zero at time 0 (since f (0) = −f (0)).
Given a signal f (t), define the signals

1 1
e(t) = (f (t) + f (−t)) , o(t) = (f (t) − f (−t)) .
2 2
It is easy to verify that o(t) is an odd signal and e(t) is an even signal. Further-
more, x(t) = e(t) + o(t). Thus, any signal can be decomposed as a sum of an
even signal and an odd signal.
2.4 Exponential and Sinusoidal Signals

2.4.1 Continuous-Time Complex Exponential Signals
Consider a signal of the form
f (t) = Ceat
where C and a are complex numbers. If both C and a are real, there are three
possible behaviors for this signal. If a < 0, then the signal goes to zero as
t → ∞, and if a > 0, the signal goes to ∞ as t → ∞. For a = 0, the signal is
constant.
Now suppose f (t) = ej(ω0 t+φ) for some positive real number ω0 and real number
φ (this corresponds to C = ejφ and a = jω0 in the signal given above). We first
note that
f (t + T ) = ej(ω0 (t+T )+φ) = ej(ω0 t+φ) ejω0 T .
If T is such that ω0 T is an integer multiple of 2π, we have ejω0 T = 1 and the
signal is periodic with period T . Thus, the fundamental period of this signal is
2π
T0 = .
ω0
Note that if ω0 = 0, then f (t) = 1 and is thus periodic with any period. The
fundamental period is undefined in this case. Also note that f (t) = e−jω0 T
is also periodic with period T0 . The quantity ω0 is called the fundamental
frequency of the signal.
Note that periodic signals (other than the one that is zero for all time) have
infinite energy, but finite average power. Specifically, let
Z T0
Ep = |f (t)|2 dt
0
be the energy of the signal over one period. The average power over that period
E
is then Pp = T0p , and since this extends over all time, this ends up being the
average power of the signal as well. For example, for the signal f (t) = ej(ω0 t+φ) ,
we have Z T Z T
1 1
P∞ = lim |f (t)|2 dt = lim 1dt = 1.
T →∞ 2T −T T →∞ 2T −T
2.4 Exponential and Sinusoidal Signals 9
Given a complex exponential with fundamental frequency ω0 , a harmonically

related set of complex exponentials is a set of periodic exponentials of the form
φk (t) = ejkω0 t , k ∈ Z.
In other words, it is the set of complex exponentials whose frequencies are

multiples of the fundamental frequency ω0 . Note that if ejω0 t is periodic with
period T0 (i.e, ω0 T0 = 2πm for some integer m), then φk (t) is also periodic with
period T0 for any k ∈ Z, since
φk (t + T0 ) = ejkω0 (t+T0 ) = ejkω0 T0 ejkω0 t = ejkm2π φk (t) = φk (t).
Although the signal f (t) given above is complex-valued in general, its real part
and imaginary part are sinusoidal. To see this, use Euler’s formula to obtain
Aej(ω0 t+φ) = A cos(ω0 t + φ) + jA sin(ω0 t + φ).
Similarly, we can write

A j(ω0 t+φ) A −j(ω0 t+φ)
A cos(ω0 t + φ) = e + e ,
2 2
i.e., a sinusoid can be written as a sum of two complex exponential signals.
Using the above, there are two main observations. First, continuous-time com-
plex exponential signals are periodic for any ω0 ∈ R (the fundamental period is
2π
ω0 for ω0 6= 0 and undefined otherwise). Second, the larger ω0 gets, the smaller
the period gets.
We will now look at discrete-time complex exponential signals and see that the
above two observations do not necessarily hold for such signals.
2.4.2 Discrete-Time Complex Exponential Signals

As in the continuous-time case, a discrete-time complex exponential signal is of
the form
f [n] = Cean
where C and a are general complex numbers. As before, let us focus on the case
where C = 1 and a = jω0 for some ω0 ∈ R in order to gain some intuition, i.e.,
f [n] = ejω0 n .
To see the differences in discrete-time signals from continuous-time signals, re-
call that a continuous-time complex exponential is always periodic for any ω0 .
The first difference between discrete-time complex exponentials and continuous-
time complex exponentials is that discrete-time complex exponentials are not
necessarily periodic. Specifically, consider the signal f [n] = ejω0 n , and suppose
it is periodic with some period N0 . Then by definition, it must be the case that
f [n + N0 ] = ejω0 (n+N0 ) = ejω0 n ejω0 N0 = f [n]ejω0 N0 .

Due to periodicity, we must have ejω0 N0 = 1, or equivalently, ω0 N0 = 2πk for

some integer k. However, N0 must be an integer, and thus we see that this can
be satisfied if and only of ω0 is a rational multiple of 2π. In other words, only
discrete-time complex exponentials whose frequencies are of the form
k
ω0 = 2π
N
for some integers k and N are periodic. The fundamental period N0 of a signal
is the smallest nonnegative integer for which the signal is periodic. Thus, for
discrete-time complex exponentials, we find the fundamental period by first
writing
ω0 k
=
2π N
where k and N have no factors in common. The value of N in this representation
is then the fundamental period.
2π 3π
Example 2.1. Consider the signal f [n] = ej 3 n + ej 4 n . Since both of the
exponentials have frequencies that are rational multiples of 2π, they are both
periodic. For the first exponential, we have
2π
3 1
= ,
2π 3
which cannot be reduced any further. Thus the fundamental period of the first
exponential is 3. Similarly, for the second exponential, we have
3π
4 3
= .
2π 8
Thus the fundamental period of the second exponential is 8. Thus f [n] is pe-
riodic with period 24 (the least common multiple of the periods of the two
signals).
The same reasoning applies to sinusoids of the form f [n] = cos(ω0 n). A nec-
essary condition for this function to be periodic is that there are two positive
integers n1 , n2 with n2 > n1 such that f [n1 ] = f [n2 ]. This is equivalent to
cos(ω0 n1 ) = cos(ω0 n2 ). Thus, we must either have
ω0 n2 = ω0 n1 + 2πk
or
ω0 n2 = −ω0 n1 + 2πk
for some positive integer k. In either case, we see that ω0 has to be a rational
multiple of 2π. In fact, when ω0 is not a rational multiple of 2π, the function
cos(ω0 n) never takes the same value twice for positive values of n.
The second difference from continuous-time complex exponentials pertains to
the period of oscillation. Specifically, even for periodic discrete-time complex
2.4 Exponential and Sinusoidal Signals 11
exponentials, increasing the frequency does not necessarily make the period
smaller. Consider the signal g[n] = ej(ω0 +2π)n , i.e., a complex exponential with
frequency ω0 + 2π. We have
g[n] = ejω0 n ej2πn = ejω0 n = f [n],
i.e., the discrete-time complex exponential with frequency ω0 + 2π is the same

as the discrete-time complex exponential with frequency ω0 , and thus they have
the same fundamental period. More generally, any two complex exponential
signals whose frequencies differ by an integer multiple of 2π are, in fact, the
same signal.
This shows that all of the unique complex exponential signals of the form ejω0 n
have frequencies that are confined to a region of length 2π. Typically, we will
consider this region to be 0 ≤ ω0 < 2π, or −π ≤ ω0 < π. Suppose we consider
the interval 0 ≤ ω0 < 2π. Note that
ejω0 n = cos(ω0 n) + j sin(ω0 n).
As ω0 increases from 0 to π, the frequencies of both the sinusoids increase.1 For

those ω0 between 0 and π that are also rational multiples of π, the sampled
signals will be periodic, and their period will decrease as ω0 increases.
Now suppose π ≤ ω0 < 2π. Consider the frequency 2π − ω0 , which falls between
0 and π. We have
ej(2π−ω0 )n = e−jω0 n = cos(ω0 n) − j sin(ω0 n).
Since ejω0 n = cos(ω0 n) + j sin(ω0 n), the frequency of oscillation of the discrete-
time complex exponential with frequency ω0 is the same as the frequency of
oscillation of the discrete-time complex exponential with frequency 2π − ω0 .
Thus, as ω0 crosses π and moves towards 2π, the frequency of oscillation starts
to decrease.
To illustrate this, it is again instructive to consider the sinusoidal signals f [n] =
cos(ω0 n) for ω0 ∈ {0, π2 , π, 3π
2 }. When ω0 = 0, the function is simply constant at
1 (and thus its period is undefined). We see that the functions with ω0 = π2 and
ω0 = 3π2 have the same period (in fact, they are exactly the same function).
2
The following table shows the differences between continuous-time and discrete-
time signals.
1 As we will see later in the course, the signals cos(ω n) and sin(ω n) correspond to
0 0
continuous-time signals of the form cos(ω0 t) and sin(ω0 t) that are sampled at 1 Hz. When
ω0
0 ≤ ω0 < π, this sampling rate is above the Nyquist frequency π , and thus the sampled
signals will be an accurate representation of the underlying continuous-time signal.
2 Note that cos(ω n) = cos((2π − ω )n) for any 0 ≤ ω ≤ π. The same is not true for
0 0 0
sin(ω0 n). In fact, one can show that for any two different frequencies 0 ≤ ω0 < ω1 ≤ 2π,
sin(ω0 n) and sin(ω1 n) are different functions.
ejω0 t ejω0 n
Distinct signals for different values of ω0 Identical signals for values of ω0 sepa-
rated by 2π
k
Periodic for any ω0 Periodic only if ω0 = 2π N for some in-
tegers k and N > 0.
Fundamental period: undefined for Fundamental period undefined for ω0 =
2π 2π
ω0 = 0 and ω0
otherwise 0 and k ω0
otherwise
Fundamental frequency ω0 Fundamental frequency ωk0
As with continuous-time signals, for any period N , we define the harmonic
family of discrete-time complex exponentials as
2π
φk [n] = ejk N n , k ∈ Z.
This is the set of all discrete-time complex exponentials that have a common
period N , and frequencies whose multiples of 2π
N . This family will play a role
in our analysis later in the course.
2.5 Impulse and Step Functions

2.5.1 Discrete-Time
The discrete-time unit impulse signal (or function) is defined as
(
0 if n 6= 0
δ[n] = .
1 if n = 0
The discrete-time unit step signal is defined as

(
0 if n < 0
u[n] = .
1 if n ≥ 0
Note that by the time-shifting property, we have
δ[n] = u[n] − u[n − 1]

X∞
u[n] = δ[n − k].
k=0
In other words, the unit step function can be viewed as a superposition of shifted
impulse functions.
Suppose we are given some arbitrary signal f [n]. If we multiply f [n] by the
time-shifted impulse function δ[n − k], we get a signal that is zero everywhere
except at n = k, where it takes the value f [k]. This is known as the sampling
or sifting property of the impulse function:
f [n]δ[n − k] = f [k]δ[n − k].

2.5 Impulse and Step Functions 13
More generally, for any signal f [n], we have

∞
X
f [n] = f [k]δ[n − k],
k=−∞
i.e., any function f [n] can be written as a sum of scaled and shifted impulse
functions.
2.5.2 Continuous-Time
The continuous-time unit step function is defined by
(
0 if t < 0
u(t) = .
1 if t ≥ 0
Note that u(t) is discontinuous at t = 0 (we will take it to be continuous from

the right).
To define the continuous-time analog of the discrete-time impulse function, we
first define the signal 
0 if t < 0

δ (t) = 1 if 0 ≤ t ≤ ,

0 if t >

R∞
where ∈ R>0 . Note that for any > 0, we have −∞ δ (t)dt = 1. As
gets smaller, the width of this function gets smaller and the height increases
proportionally. The continuous-time impulse function is defined as the limit of
the above function as approaches zero from the right:
δ(t) = lim δ (t).

↓0
This function is drawn with an arrow at the origin (since it has no width and
infinite height). We will often be interested in working with scaled and time-
shifted versions of the continuous-time impulse function. Just as we did with
discrete-time functions, we can take a continuous-time function f (t) and repre-
sent it as Z ∞
f (t) = f (τ )δ(t − τ )dτ.
−∞
In other words, if we take an infinite sequence of shifted impulse functions, scale

each of them by the value of the function f (t) at the value of the time-shift, and
add them together (represented by the integration), we get the function f (t).
For instance, we have
Z ∞ Z ∞
u(t) = u(τ )δ(t − τ )dτ = δ(t − τ )dτ.
−∞ 0
Just as the discrete-time impulse function could be viewed as a difference of

the discrete-time unit step and its time-shifted version, the continuous-time
impulse function can be viewed as the derivative of the continuous-time unit
step function.
2.6 Properties of Systems

As we discussed during the first lecture, a system can be viewed as an abstract
object that takes input signals and produces output signals. A continuous-time
system operates on continuous-time signals, and discrete-time systems operate
with discrete-time signals. Examples of the former include many physical sys-
tems such as electrical circuits, vehicles, etc. Examples of the latter include
computer systems, a bank account where the amount of money is incremented
with interest, deposits and withdrawals at the end of each day, etc.
2.6.1 Interconnections of Systems
We will often be interested in connecting different systems together in order to

achieve a certain objective. There are three basic interconnection patterns that
are used to build more complicated interconnections.
The first is a serial connection of systems:
Input Output
System 1 System 2
An example of a series interconnection of systems occurs in communication

systems; the signal to be transmitted is first passed through an encoder, which
transforms the signal into a form suitable for transmission. That transformed
signal is then sent through the communication channel (the second system in
the chain). The output of the communication channel is then passed through a
decoder, which is the third system in the chain. The output of the decoder is
an estimate of the signal that entered the encoder.
The second type of interconnection is a parallel interconnection:
System 1
Input Output
+
System 2
2.6 Properties of Systems 15
Input Output
+ System 1
System 2
The third type of interconnection is a feedback interconnection:

Feedback interconnections form the basis of control systems; in this case we are
given a specific system (System 1) that we wish to control (or make behave in
a certain way). The second system (System 2) is a controller that we design in
order to achieve the desired behavior. This controller takes the current output
of the system and uses that to decide what other inputs to apply to the original
system in order to change the output appropriately.
2.6.2 Properties of Systems
Systems With and Without Memory
A system is memoryless if the output each time-instant (either in discrete-

time or continuous-time) only depends on the input at that time-instant. For
example, the system
y[n] = cos(x[n])
is memoryless, as the output at each time-step n ∈ Z only depends on the input
at that time-step.
However, the system whose input and output are related by
Z t
y(t) = x(τ )dτ
−∞
is not memoryless, as the output depends on all of the input values from the
past.
Systems with memory are often represented as having some sort of state and
dynamics, which maintains the necessary information from the past. For exam-
ple, for the system given above, we can use the fundamental theorem of calculus
to obtain
dy(t)
= x(t)
dt
where the state of the system is y(t) (this is also the output), and the dynamics
of the state are given by the differential equation above. Similarly for the
discrete-time system
n
X
y[n] = x[k]
k=−∞
we have
n−1
X
y[n] = x[k] + x[n] = y[n − 1] + x[n]
k=−∞
which is a difference equation describing how the state y[n] evolves over time.
Invertibility
A system is said to be invertible if distinct inputs lead to distinct outputs. In

other words, by looking at the output of a system, one can uniquely identify
what the input was.
An example of an invertible system is y(t) = αx(t), where α is any nonzero
real number. Given the output y(t), we can uniquely identify the input as
x(t) = α1 y(t). However, if α = 0, then we have y(t) = 0 regardless of the input,
and there is no way to recover the input. In that case, the system would not be
invertible.
Another example of a noninvertible system is y(t) = x2 (t), as the sign of the
input is lost when converting to the output.
Pn
The system y[n] = k=−∞ x[k] is invertible; to see this, we use the equivalent
representation y[n] = y[n − 1] + x[n] to obtain x[n] = y[n] − y[n − 1] for all
n ∈ Z.
Causality
A system is causal if the output of the system at any time depends only on
the input at that time and from the past. In other words, for all t ∈ R, y(t)
depends only on x(τ ) for τ ≤ t. Thus, a causal system does not react to inputs
that will happen in the future. For a causal system, if two different inputs have
the same values up to a certain time, the output of the system due to those two
inputs will agree up to that time as well. All memoryless systems are causal.
There are various instances where we may wish to use noncausal systems. For
example, if we have time-series data saved offline, we can use the saved values
of the signal for k > n to process the signal at a given time-step n (this can be
used for music and video editing, for example). Alternatively, the independent
variable may represent space, rather than time. In this case, one can use the
values of the signal from points on either side of a given point in order to process
the signal.
Example 2.2. The signal y[n] = x[−n] is noncausal; for example, y[−1] = x[1],
and thus the output at negative time-steps depends on the input from positive
time-steps (i.e., in the future).
The signal y(t) = x(t) cos(t + 1) is causal; the t + 1 term does not appear in the
input, and thus the output at any time does not depend on values of the input
at future times.
Stability
The notion of stability is a critical system property. There are many different
notions of stability that can be considered, but for the purposes of this course,
we will say that a system is stable if a bounded input always leads to a bounded
output. In other words, for a continuous-time system, if there exists a constant
B1 ∈ R≥0 such that the input satisfies |x(t)| ≤ B1 for all t ∈ R, then there
should exist some other constant B2 ∈ R≥0 such that |y(t)| ≤ B2 for all t ∈
R. An entirely analogous definition holds for discrete-time systems. Loosely
speaking, for a stable system, the output cannot grow indefinitely when the
input is bounded by a certain value.
Example 2.3. The system y(t) = tx(t) is memoryless and causal, but not
stable. For example, if x(t) = 1 for all t ∈ R, we have y(t) = t which is not
bounded by any constant.
Similarly, the
Psystem y[n] = y[n − 1] + x[n] is not stable. This is seen by noting
n
that y[n] = k=−∞ x[k]. So, for example, if x[n] = u[n], we have y[n] = (n + 1)
if n ≥ 0, which is unbounded.
An example of a stable causal memoryless system is y(t) = cos(x(t)). Another
example of a stable and causal system is
(
0 if n < 0
y[n] = ,
αy[n − 1] + x[n] if n ≥ 0
where α ∈ R satisfies |α| < 1. Specifically, if |x[n]| ≤ B1 for all n ∈ Z, then we

B1
have |y[n]| ≤ 1−|α| for all n ∈ Z. To see this, we prove by induction. Clearly
B1 B1
|y[n]| ≤ 1−|α| for n ≤ 0. Suppose that |y[n]| ≤ 1−|α| for some n ≥ 0. Then we
have
|y[n + 1]| = |αy[n] + x[n + 1]| ≤ |α||y[n]| + |x[n + 1]|

B1
≤ |α| + B1
1 − |α|
B1
= .
1 − |α|
B1
Thus, by induction, we have |y[n]| ≤ 1−|α| for all n ∈ Z.
The above notion of stability is known as Bounded-Input-Bounded-Output (BIBO)

stability. There are also other notions of stability, such as ensuring that the in-
ternal state of the system remains stable as well. One of the main objectives of
control systems is to ensure that the overall system remains stable, as you will
see in your control systems courses in later semesters.
Time-Invariance
A system is said to be time-invariant if the system reacts in the same way to

an input signal, regardless of the time at which the input is applied. In other
words, if y(t) is the output signal when the input signal is x(t), then the output
due to x(t − t0 ) should be y(t − t0 ) for any time-shift t0 . Note that for a system
to be time-invariant, this should hold for every input signal.
Example 2.4. The system y(t) = cos(x(t)) is time-invariant. Suppose we
define the input signal w(t) = x(t − t0 ) (i.e., a time-shifted version of x(t)). Let
yw (t) be the output of the system when w(t) is applied. Then we have
yw (t) = cos(w(t)) = cos(x(t − t0 )) = y(t − t0 )
and thus the output due to x(t − t0 ) is time-shifted version of y(t), as required.
An example of a time-varying system is y[n] = nx[n]. For example, if x[n] =
δ[n], then we have the output signal y[n] = 0 for all time. However, if x[n] =
δ[n − 1], then we have y[n] = 1 for n = 1 and zero for all other times. Thus a
shift in the input did not result in a simple shift in the output.
Linearity
A system is linear if it satisfies the following two properties.
1. Additivity: Suppose the output is y1 (t) when the input is x1 (t), and the
output is y2 (t) when the input is x2 (t). Then the output to x1 (t) + x2 (t)
is y1 (t) + y2 (t).
2. Scaling: Suppose the output is y(t) when the input is x(t). Then for any
complex number α, the output should be αy(t) when the input is αx(t).
Both properties together define the superposition property: if the input to the
system is α1 x1 (t) + α2 x2 (t), then the output should be α1 y1 (t) + α2 y2 (t). Note
that this must hold for any inputs and scaling parameters in order for the system
to qualify as linear. An entirely analogous definition holds for discrete-time
systems.
For any linear system, the output must be zero for all time when the input is
zero for all time. To see this, consider any arbitrary input x(t), and let the
corresponding output be y(t). Then, using the scaling property, the output to
αx(t) must be αy(t) for any scalar complex number α. Simply choosing α = 0
yields the desired result that the output will be the zero signal when the input
is the zero signal.
Example 2.5. The system y(t) = tx(t) is linear. To see this, consider two
arbitrary input signals x1 (t) and x2 (t), and two arbitrary scalars α1 , α2 . Then
we have
t(α1 x1 (t) + α2 x2 (t)) = α1 tx1 (t) + α2 tx2 (t) = α1 y1 (t) + α2 y2 (t)
where y1 (t) and y2 (t) are the outputs due to x1 (t) and x2 (t), respectively.
The system y[n] = x2 [n] is nonlinear. Let y1 [n] = x21 [n] and y2 [n] = x22 [n].
Consider the input x3 [n] = x1 [n] + x2 [n]. Then the output due to x3 [n] is
y3 [n] = x23 [n] = (x1 [n] + x2 [n])2 6= x21 [n] + x22 [n]
in general. Thus the additivity property does not hold, and the system is
nonlinear.
The system y[n] = Re{x[n]} is nonlinear, where Re{·} denotes the real part
of the argument. To see this, let x[n] = a[n] + jb[n], where a[n] and b[n] are
real-valued signals. Consider a scalar α = j. Then we have
y[n] = Re{x[n]} = a[n].
However, Re{jx[n]} = Re{ja[n] − b[n]} = −b[n] 6= jy[n]. Thus, scaling the

input signal by the scalar j does not result in the output being jy[n], and so
the scaling property does not hold.
The system y(t) = 2x(t) + 5 is nonlinear; it violates the fact that the all-zero
input should cause the output to be zero for all time. One can also verify this by
applying two different constant input signals and checking that the output due
to the sum of the inputs is not equal to the sum of the corresponding outputs.
We will be focusing almost entirely on linear time-invariant systems in this

course; in practice, many systems are nonlinear and time-varying, but can of-
ten be approximated by linear time-invariant systems under certain operation
conditions.
Chapter 3
Analysis of Linear
Time-Invariant Systems
Reading: Signals and Systems, Sections 2.0-2.4.

In this part of the course, we will focus on understanding the behavior of linear
time-invariant (LTI) systems. As we will see, the linearity and time-invariance
properties provide a nice way to understand the input-output relationship of the
system. To develop this, let us start by considering discrete-time LTI systems.
3.1 Discrete-Time LTI Systems
Consider a discrete-time system with input x[n] and output y[n]. First, define
the impulse response of the system to be the output when x[n] = δ[n] (i.e.,
the input is an impulse function). Denote this impulse response by the signal
h[n].
Now, consider an arbitrary signal x[n]. Recall from the sifting property of the
impulse function that
∞
X
x[n] = x[k]δ[n − k],
k=−∞
i.e., x[n] can be written as a superposition of scaled and shifted impulse func-
tions.
Since the system is time-invariant, the response of the system to the input
δ[t − k] is h[t − k]. By linearity (and specifically the scaling property), the
response to x[k]δ[n − k] is x[k]δ[n − k]. By the additivity property, the response
22 Analysis of Linear Time-Invariant Systems
P∞
to k=−∞ x[k]δ[n − k] is then
∞
X
y[n] = x[k]h[n − k].
k=−∞
The above is called the convolution sum; the convolution of the signals x[n]
and h[n] is denoted by
∞
X
x[n] ∗ h[n] = x[k]h[n − k].
k=−∞
Thus we have the following very important property of discrete-time LTI sys-
tems: if x[n] is the input signal to an LTI system, and h[n] is the
impulse response of the system, then the output of the system is
y[n] = x[n] ∗ h[n].
Example 3.1. Consider an LTI system with impulse response
(
1 if 0 ≤ n ≤ 3,
h[n] =
0 otherwise.
Suppose the input signal is

(
1 if 0 ≤ n ≤ 3,
x[n] =
0 otherwise.
Then we have
∞
X
y[n] = x[n] ∗ h[n] = x[k]h[n − k].
k=−∞
Since both x[k] = 0 for k < 0 and h[n − k] = 0 for k > n,

n
X
y[n] = x[k]h[n − k].
k=0
Thus y[n] = 0 for n < 0. When n = 0 we have

0
X
y[0] = x[k]h[−k] = x[0]h[0] = 1.
k=0
When n = 1 we have
1
X
y[1] = x[k]h[1 − k] = x[0]h[1] + x[1]h[0] = 2.
k=0
Similarly, y[2] = 3, y[3] = 4, y[4] = 3, y[5] = 2, y[6] = 1 and y[n] = 0 for n ≥ 7.

3.2 Continuous-Time LTI Systems 23
Example 3.2. Consider an LTI system with impulse response h[n] = u[n].
Suppose the input signal is x[n] = αn u[n] with 0 < α < 1. Then we have
∞
X
y[n] = x[n] ∗ h[n] = x[k]h[n − k].
k=−∞
Since both x[k] = 0 for k < 0 and h[n − k] = 0 for k > n, we have
n n
X X 1 − αn+1
y[n] = x[k]h[n − k] = αk =
1−α
k=0 k=0
for n ≥ 0, and y[n] = 0 for n < 0.
3.2 Continuous-Time LTI Systems

The analysis and intuition that we developed for discrete-time LTI systems
carries forward in an entirely analogous way for continuous-time LTI systems.
Specifically, recall that for any signal x(t), we can write
Z ∞
x(t) = x(τ )δ(t − τ )dτ.
−∞
The expression on the right hand side is a superposition of scaled and shifted
impulse functions. Thus, when this signal is applied to an LTI system, the
output will be a superposition of scaled and shifted impulse responses. More
specifically, if h(t) is the output of the system when the input is x(t) = δ(t),
then the output for a general input x(t) is given by
Z ∞
y(t) = x(τ )h(t − τ )dτ.
−∞
This is the convolution integral and is denoted by y(t) = x(t) ∗ h(t).

Example 3.3. Suppose x(t) = e−at u(t) with a ∈ R>0 and h(t) = u(t). Then
the output of the LTI system with impulse response h(t) is given by
Z ∞
y(t) = x(t) ∗ h(t) = x(τ )h(t − τ )dτ
−∞
Z ∞
= e−aτ u(τ )u(t − τ )dτ
−∞
Z t
= e−aτ dτ
0
if t ≥ 0, and y(t) = 0 otherwise. Evaluating the above expression, we have

(
1
(1 − e−at ) if t ≥ 0
y(t) = a
0 if t < 0.
Example 3.4. Consider the signals

( (
1 if 0 < t < T t if 0 < t < 2T
x(t) = , h(t) = ,
0 otherwise 0 otherwise
where T > 0 is some constant. The convolution of these signals is easiest to do
graphically and by considering different regions of the variable t. The result is



 0 t<0
1
2t 2
0<t<T



y(t) = T t − 12 T 2 T < t < 2T .
1 2 3 2

−2t + Tt + 2T 2T < t < 3T





0 t > 3T
3.3 Properties of Linear Time-Invariant Systems

In this section we will study some useful properties of the convolution operation;
based on the previous section, this will have implications for the input-output
behavior of linear time-invariant systems (h[n] for discrete-time systems and
h(t) for continuous-time systems).
3.3.1 The Commutative Property

The first useful property of convolution is that it is commutative:
x[n] ∗ h[n] = h[n] ∗ x[n]
x(t) ∗ h(t) = h(t) ∗ x(t).
To see this, start with the definition of convolution and perform a change of
variable by setting r = n − k. This gives
∞
X ∞
X
x[n] ∗ h[n] = x[k]h[n − k] = x[n − r]h[r] = h[n] ∗ x[n].
k=−∞ r=−∞
The same holds for the continuous-time convolution. Thus it does not matter
which of the signals we choose to flip and shift in the convolution operation.
3.3.2 The Distributive Property

The second useful property of convolution is that it is distributive:
x[n] ∗ (h1 [n] + h2 [n]) = x[n] ∗ h1 [n] + x[n] ∗ h2 [n]
x(t) ∗ (h1 (t) + h2 (t)) = x(t) ∗ h1 (t) + x(t) ∗ h2 (t).
3.3 Properties of Linear Time-Invariant Systems 25
This property is easy to verify:

∞
X
x[n] ∗ (h1 [n] + h2 [n]) = x[k](h1 [n − k] + h2 [n − k])
k=−∞
X∞ ∞
X
= x[k]h1 [n − k] + x[k]h2 [n − k]
k=−∞ k=−∞
= x[n] ∗ h1 [n] + x[n] ∗ h2 [n].
The distributive property has implications for LTI systems connected in parallel:
y1 [n]
h1 [n]
x[n] y[n]
+
h2 [n]
y2 [n]
Let h1 [n] be the impulse response of System 1, and let h2 [n] be the impulse
response for System 2. Then we have y1 [n] = x[n]∗h1 [n] and y2 [n] = x[n]∗h2 [n].
Thus,
y[n] = y1 [n] + y2 [n] = x[ n] ∗ h1 [n] + x[n] ∗ h2 [n] = x[n] ∗ (h1 [n] + h2 [n]).
The above expression indicates that the parallel interconnection can equivalently
be viewed as x[n] passing through a single system whose impulse response is
h1 [n] + h2 [n]:
x[n] y[n]
h1 [n] + h2 [n]
3.3.3 The Associative Property
A third useful property of convolution is that it is associative:
x[n] ∗ (h1 [n] ∗ h2 [n]) = (x[n] ∗ h1 [n]) ∗ h2 [n]

x(t) ∗ (h1 (t) ∗ h2 (t)) = (x(t) ∗ h1 (t)) ∗ h2 (t).
In other words, it does not matter which order we do the convolutions. The
above relationships can be proved by manipulating the summations (or inte-
grals); we won’t go into the details here.
x[n] y1 [n] y[n]

h1 [n] h2 [n]
Figure 3.1: A series interconnection of systems.
Just as the distributive property had implications for parallel interconnections

of systems, the associative property has implications for series interconnections
of systems. Specifically, consider the series interconnection shown in Fig. 3.1.
We have
y[n] = y1 [n] ∗ h2 [n] = (x[n] ∗ h1 [n]) ∗ h2 [n] = x[n] ∗ (h1 [n] ∗ h2 [n]).
Thus, the series interconnection is equivalent to a single system with impulse

response h1 [n] ∗ h2 [n], as shown in Fig. 3.2.
x[n] y[n]
h1 [n] ∗ h2 [n]
Figure 3.2: The equivalent representation of a series interconnection.
Further note that since h1 [n] ∗ h2 [n] = h2 [n] ∗ h1 [n], we can also interchange the
order of the systems in the series interconnection as shown in Fig. 3.3, without
changing the overall input-output relationship between x[n] and y[n].
x[n] y[n]
h2 [n] h1 [n]
Figure 3.3: An equivalent series representation of the interconnection shown in

Fig. 3.1.
3.3.4 Memoryless LTI Systems

Let us now see the implications of the memoryless property for LTI systems.
Specifically, let h[n] (or h(t)) be the impulse response of a given LTI system.
Since we have
∞
X ∞
X
y[n] = x[k]h[n − k] = x[n − k]h[k],
k=−∞ k=−∞
we see that y[n] will depend on a value of the input signal other than at time-
step n unless h[k] = 0 for all k 6= 0. In other words, for an LTI system to
be memoryless, we require h[n] = Kδ[n] for some constant K. Similarly, a
continuous-time LTI system is memoryless if and only if h(t) = Kδ(t) for some
constant K. In both cases, all LTI memoryless systems have the form
y[n] = Kx[n] or y(t) = Kx(t)
for some constant K.
3.3.5 Invertibility of LTI Systems
Consider an LTI system with impulse response h[n] (or h(t)). Recall that the
system is said to be invertible if the output of the system uniquely specifies the
input. If a system is invertible, there is another system (known as the “inverse
system”) that takes the output of the original system and outputs the input to
the original system, as shown in Fig. 3.4.
x[n] y[n] x[n]

h[n] Inverse system
Figure 3.4: A system in series with its inverse.
Suppose the second system is LTI and has impulse response hI [n]. Then, by
the associative property discussed earlier, we see that the series interconnection
of the system with its inverse is equivalent (from an input-output sense) to a
single system with impulse response h[n] ∗ hI [n]. In particular, we require
x[n] = x[n] ∗ (h[n] ∗ hI [n])
for all input signals x[n], from which we have
h[n] ∗ hI [n] = δ[n].
In other words, if we have an LTI system with impulse response h[n], and
another LTI system with impulse response hI [n] such that h[n] ∗ hI [n] = δ[n],
then those systems are inverses of each other. The analogous statement holds
in continuous-time as well.
Example 3.5. Consider the LTI system with impulse response h[n] = αn u[n].
One can verify that this impulse response corresponds to the system
n
X
y[n] = x[k]αn−k = αy[n − 1] + x[n].
k=−∞
Now consider the system yI [n] = xI [n] − αxI [n − 1], with input signal xI [n]
and output signal yI [n]. The impulse response of this system is hI [n] = δ[n] −
αδ[n − 1]. We have
h[n] ∗ hI [n] = αn u[n] ∗ (δ[n] − αδ[n − 1])

= αn u[n] ∗ δ[n] − (αn u[n]) ∗ (αδ[n − 1])
= αn u[n] − α(αn−1 u[n − 1])
= αn (u[n] − u[n − 1])
= αn δ[n]
= δ[n].
Thus, the system with impulse response hI [n] is the inverse of the system with
impulse response h[n].
3.3.6 Causality of LTI Systems
Recall that a system is causal if its output at time t depends only on the inputs
up to (and potentially including) t. To see what this means for LTI systems,
consider the convolution sum
∞
X ∞
X
y[n] = x[k]h[n − k] = x[n − k]h[k],
k=−∞ k=−∞
where the second expression follows from the commutative property of the con-
volution. In order for y[n] to not depend on x[n + 1], x[n + 2], . . ., we see that
h[k] must be zero for k < 0. The same conclusion holds for continuous-time
systems, and thus we have the following: A continuous-time LTI system is
causal if and only if its impulse response h(t) is zero for all t < 0. A
discrete-time LTI system is causal if and only if its impulse response
h[n] is zero for all n < 0.
Note that causality is a property of a system; however we will sometimes refer
to a signal as being causal, by which we simply mean that its value is zero for
n or t less than zero.
3.3.7 Stability of LTI Systems
To see what the LTI property means for stability of systems, consider again the
convolution sum
∞
X
y[n] = x[k]h[n − k].
k=−∞
Note that

∞
X ∞
X
|y[n]| = x[k]h[n − k] ≤ |x[k]h[n − k]|

k=−∞ k=−∞
X∞
= |x[k]||h[n − k]|.
k=−∞
Now suppose that x[n] is bounded, i.e., there exists some B ∈ R≥0 such that
|x[n]| ≤ B for all n ∈ Z. Then the above expression becomes
∞
X
|y[n]| ≤ B |h[n − k]|.
k=−∞
P∞
Thus, if k=−∞ |h[n−k]| < ∞ (which means that h[n] is absolutely summable),
then |y[n]| will also P
be bounded for all n. It turns out that this is a necessary
∞
condition as well: if k=−∞ |h[n − k]| = ∞, then there is a bounded input that
causes the output to be unbounded.
The same conclusion holds in continuous-time as well. Thus,
R ∞ we have: A
continuous-time LTI system is stable if and only if −∞ |h(τ )|dτ < ∞.
P∞
A discrete-time LTI system is stable if and only if k=−∞ |h[k]| < ∞.
Example 3.6. Consider the LTI system with impulse response h[n] = αn u[n],
where α ∈ R. We have
∞ ∞
(
1
X X if |α| < 1
|h[k]| = |α|k = 1−|α| .
k=−∞ k=0
∞ if |α| ≥ 1
Thus, the system is stable if and only if |α| < 1.

Similarly, consider the continuous-time LTI system with impulse response h(t) =
eαt u(t), where α ∈ R. We have
Z ∞ Z ∞ ∞ ( 1
1 − α if α < 0
eατ dτ = (eατ ) =

|h(τ )|dτ = .
−∞ 0 α 0 ∞ if α ≥ 0
Thus, the system is stable if and only if α < 0.
3.3.8 Step Response of LTI Systems

Just as we defined the impulse response of a system to be the output of the
system when the input is an impulse function, we define the step response of
a system to be the output when the input is a step function u[n] (or u(t) in
continuous-time). We denote the step response as s[n] for discrete-time systems
and s(t) for continuous-time systems.
To see how the step response is related to the impulse response, note that
∞
X ∞
X n
X
s[n] = u[k]h[n − k] = u[n − k]h[k] = h[k].
k=−∞ k=−∞ k=−∞
This is equivalent to s[n] = s[n − 1] + h[n]. Thus, the step response of a discrete-
time LTI system is the running sum of the impulse response.
Note that this could also have been seen by noting that δ[n] = u[n] − u[n − 1].
If the impulse is applied to an LTI system, we get the impulse response h[n].
However, by the linearity property, this output must be the superposition of the
outputs due to u[n] and u[n − 1]. By the time-invariance property, the output
due to u[n−1] is s[n−1], and thus for LTI systems we have h[n] = s[n]−s[n−1],
which corroborates what we obtained above.
For continuous-time systems, we have the same idea:
Z ∞ Z ∞ Z t
s(t) = u(τ )h(t − τ )dτ = u(t − τ )h(τ )dτ = h(τ )dτ.
−∞ −∞ −∞
Differentiating both sides and applying the fundamental theorem of calculus,

we have
ds
= h(t),
dt
i.e., the impulse response is the derivative of the step response.
3.4 Differential and Difference Equation Models

for Causal LTI Systems
As we have already seen in a few examples, many systems can be described us-
ing differential equation (in continuous-time) or difference-equation (in discrete-
time) models, capturing the relationship between the input and the output. For
example, for a vehicle with velocity v(t) and input acceleration a(t), we have
dv
= a(t).
dt
If we included wind resistance or friction (which produces a force that is pro-
portional to the velocity in the opposite direction of travel), we have
dv
= −αv(t) + a(t),
dt
where α > 0 is the coefficient of friction. Similarly, given an RC circuit, if we
define the voltage across the capacitor as the output, and the source voltage as
the input, then the input and output are again related via a differential equation
of the above form.
3.4 Differential and Difference Equation Models for Causal LTI
Systems 31
In discrete-time, consider a bank-account where earnings are deposited at the

end of each month. Let the amount in the account at the end of month n be
denoted by s[n]. Then we have
s[n] = (1 + r)s[n − 1] + x[n]
where r is the interest rate and x[n] is the new amount deposited into the
account at the end of month n.
Since such differential and difference equations play a fundamental role in the
analysis of LTI systems, we will now review some methods to solve such equa-
tions.
3.4.1 Linear Constant-Coefficient Differential Equations

To illustrate the solution of linear differential equations, we consider the follow-
ing example.
Example 3.7. Consider the differential equation
dy
+ 2y(t) = x(t),
dt
where x(t) = Ke3t u(t) (K is some constant). The solution to such differential
equations is given by y(t) = yh (t) + yp (t), where yp (t) is a particular solu-
tion to the above equation, and yh (t) is a homogeneous solution satisfying the
differential equation
dyh
+ 2yh (t) = 0.
dt
The above differential equation is called homogeneous as it has no driving
function x(t).
Let us first solve the homogeneous equation. For equations of this form (where
a sum of derivatives of yh (t) have to sum to zero), a reasonable guess would be
that yh (t) takes the form
yh (t) = Aemt
for some m ∈ C. Substituting this into the homogeneous equation gives
mAemt + 2Aemt = 0 ⇒ m + 2 = 0 ⇒ m = −2.
Thus, the homogeneous solution is yh (t) = Ae−2t , for any constant A.
Next, we search for a particular solution to the equation
dyp
+ 2yp (t) = Ke3t u(t).
dt
It seems reasonable to try yp (t) = Be3t , for some constant B. Substituting and
evaluating for t > 0, we have
K
3Be3t + 2Be3t = Ke3t ⇒ B = .
5
K 3t
Thus, the particular solution is given by yp (t) = 5 e for t > 0.
−2t K 3t
Together, we have y(t) = yh (t) + yp (t) = Ae + 5 e
for t > 0. Note that
the coefficient A has not been determined yet; in order to do so, we need more
information about the solutions to the differential equation, typically in the
form of initial conditions. For example, if we know that the system is at rest
until the input is applied (i.e., y(t) = 0 until x(t) becomes nonzero), we have
y(t) = 0 for t < 0. Suppose we are given the initial condition y(0) = 0. Then,
K K
y(0) = A + =0⇒A=− .
5 5
Thus, with the given initial condition, we have y(t) = K 3t
5 (e − e−2t )u(t).
The above example illustrates the general approach to solving linear differential
equations of the form
N M
X dk y X dk x
ak k = bk k .
dt dt
k=0 k=0
First find the homogeneous solution to the equation

N
X d k yh
ak =0
dtk
k=0
by hypothesizing that yh (t) = Aemt for some m ∈ C. If there are N different

values of m, denoted m1 , m2 , . . . , mN for which the proposed form holds, then
we take the homogeneous solution to be yh (t) = A1 em1 t +A2 em2 t +· · ·+AN emN t ,
where the coefficients A1 , . . . , AN are to be determined from initial conditions.
If there are not N different values of m, then further work is required; we will
see a more general way to solve these cases later in the course.
Next, find a particular solution to the equation
N M
X dk yp X dk x
ak = bk k
dtk dt
k=0 k=0
where x(t) is some given function. The idea will be to make yp (t) a linear
combination of terms that, when differentiated, yield terms that appear in x(t)
and its derivatives. Typically this only works when x(t) involves terms like
et , sin(t), cos(t), polynomials in t, etc. Let’s try another example.
y 00 (t) + y 0 (t) − 6y(t) = x0 (t) + x(t), (3.1)
where x(t) = e4t u(t).

We first search for a homogeneous solution yh (t) = Aemt satisfying
yh00 (t)+yh0 (t)−6yh (t) = 0 ⇒ m2 Aemt +mAemt −6Aemt = 0 ⇒ (m2 +m−6) = 0.

3.4 Differential and Difference Equation Models for Causal LTI
Systems 33
This yields m = −3 or m = 2. Thus, the homogeneous solution is of the form
yh (t) = A1 e−3t + A2 e2t
for some constants A1 and A2 that will be determined from the initial conditions.
To find a particular solution, note that for t > 0, we have x0 (t) + x(t) = 4e4t +
e4t = 5e4t . Thus we search for a particular solution of the form yp (t) = Be4t
for t > 0. Substituting into the differential equation (3.1), we have
5
yp00 (t)+yp0 (t)−6yp (t) = x0 (t)+x(t) ⇒ 16Be4t +4Be4t −6Be4t = 5e4t ⇒ B = .
14
5 4t
Thus, yp (t) = 14 e for t > 0 is a particular solution.
The overall solution is then of the form y(t) = yh (t) + yp (t) = A1 e−3t + A2 e2t +
5 4t
14 e for t > 0. If we are told that the system is at rest until the input is
applied, and that y(0) = y 0 (0) = 0, we have
5
y(0) = A1 + A2 + =0
14
20
y 0 (0) = −3A1 + 2A2 + = 0.
14
1
Solving these equations, we obtain A1 = and A2 = − 21 . Thus, the solution is
7

1 −3t 1 2t 5 4t
y(t) = e − e + e u(t).
7 2 14
3.4.2 Linear Constant Coefficient Difference Equations

The same general idea that we used to solve differential equations in the previous
section apply to solving difference equations of the form
N
X M
X
ak y[n − k] = bk x[n − k]. (3.2)
k=0 k=0
The overall solution will be of the form y[n] = yh [n] + yp [n], where yh [n] is a
homogeneous solution to
XN
ak yh [n − k] = 0,
k=0
and yp [n] is a particular solution satisfying the difference equation (3.2) for the
given function x[n]. In this case, we seek homogeneous solutions of the form
yh [n] = Aβ n for some A, β ∈ C, and seek particular solutions that have the
same form as the quantities that appear in x[n]. Let’s do an example.
Example 3.9. Suppose we have the difference equation

1
y[n] − y[n − 1] = x[n], (3.3)
2
1 n

with x[n] = 3 u[n].
The solution to this difference equation will be of the form y[n] = yh [n] + yp [n],
where yh [n] is the homogeneous solution satisfying
1
yh [n] − yh [n − 1] = 0
2
and yp [n] is a particular solution to the given difference equation with the given
input signal x[n].
To find the homogeneous solution, we try yh [n] = Aβ n for some constants A
and β. Substituting into the homogeneous difference equation, we obtain
1 A 1
yh [n] − yh [n − 1] = Aβ n − β n−1 = 0 ⇒ β = .
2 2 2
Thus, the homogeneous solution is yh [n] = Aβ n for some A that we will identify
based on initial conditions (after we have found the particular solution).
To find the particular solution, we attempt to mimic the input. Thus, we seek
n
a particular solution of the form yp [n] = B 31 for n ≥ 0 and some constant
B. Substituting this into the difference equation (3.3), we have
n n−1
1 B 1 1n
B − =
3 2 3 3
for n ≥ 1 (note that we don’t look at n = 0 here because we have not defined
n Solving this, we get B = −2. Thus the particular solution is yp [n] =
yp [−1]).
−2 31 for n ≥ 0.
n n
Now, we have y[n] = yh [n] + yp [n] = A 12 − 2 13 for n ≥ 0. Suppose we are
told that the system is at rest for n < 0, i.e., y[n] = 0 for n < 0. Looking at
equation (3.3), we have
1
y[0] − y[−1] = 1 ⇒ y[0] = 1.
2
Substituting the expression for y[n], we have
1 = y[0] = A − 2 ⇒ A = 3.
Thus, the solution is given by

n n
1 1
y[n] = 3 −2 u[n].
2 3
3.5 Block Diagram Representations of Linear Differential and
Difference Equations 35
An alternative method to solve difference equations is to write them in recur-

sive form, and then iteratively solve, as shown by the following example.
Example 3.10. Suppose
1
y[n] − y[n − 1] = x[n].
2
We can rewrite this as
1
y[n − 1] + x[n].
y[n] =
2
Suppose that x[n] = δ[n] and the system is initially at rest (i.e., y[n] = 0 for
n < 0). Then we have
1
y[−1] + δ[0] = 1
y[0] =
2
1 1
y[1] = y[0] + δ[1] =
2 2
1 1
y[2] = y[1] + δ[2] =
2 4
..
.
n
1
y[n] =
2
n
Thus, the impulse response is h[n] = 12 u[n].1
3.5 Block Diagram Representations of Linear Dif-

ferential and Difference Equations
It is often useful to represent linear differential and difference equations using
block diagrams; this provides us with a way to implement such equations using
primitive computational elements (form form the components of the block dia-
gram), and to derive alternative representations of systems. Here, we will focus
on differential and difference equations of the form
dN y(t) dN −1 y(t)
+ aN −1 + · · · + a0 y(t) = b0 x(t)
dtN dtN −1
y[n + N ] + aN −1 y[n + N − 1] + · · · + a0 y[n] = b0 x[n].
Drawing block diagrams for more general differential and difference equations
(involving more than just x[n] on the right hand side) is easier using Laplace
1 One can also calculate this using the homogeneous and particular solutions; in this case,
the particular solution would

n have the form yp [n] = Bδ[n] and B would be found to be zero,
so that y[n] = yh [n] = A 12 . Under the condition of initial rest and y[0] = 1 (obtained from
the difference equation), we obtain A = 1, thus matching the impulse response calculated
recursively above.
and z-transform techniques, and so we will defer a study of such equations until
then.
For the above equations, we start by writing the highest derivative of y (or the
most advanced version of y) in terms of all of the other quantities:
dN y(t) dN −1 y(t)
= −aN −1 − · · · − a0 y(t) + b0 x(t) (3.4)
dtN dtN −1
y[n + N ] = −aN −1 y[n + N − 1] − · · · − a0 y[n] + b0 x[n]. (3.5)
Next, we use a key building block: the integrator block (for continuous-time)
or the delay block (for discrete-time). Specifically, the integrator block is a
system whose output is the integral of the input, and the delay block is a system
whose output is a delayed version of the input. Thus, if we feed dy dt into the
integrator block, we get y(t) out, and if we feed y[n + 1] into the delay block,
we get y[n] out, as shown in Fig. 3.5.
dy(t)
dt R y(t) y[n + 1] y[n]
D
Figure 3.5: Integrator and Delay blocks.
To use these blocks to represent differential and difference equations, we simply

chain a sequence of these blocks in series, and feed the highest derivative into
the first block in the chain, as shown in Fig. 3.6.
dN −1 y(t) dN −2 y(t)
dN y(t) dtN −1 dtN −2 dy(t)
dtN R R dt R y(t)
···
y[n + N − 1] y[n + N − 2]
y[n + N ] y[n + 1] y[n]
D D ··· D
Figure 3.6: Chained integrator and delay blocks.
This series chain of integrator (or delay) blocks provides us with all of the signals
needed to represent (3.4) and (3.5). Specifically, from equation (3.4), we see that
N −1
dN y(t)
dtN
is a linear combination of the signals d dtN −1y(t)
, · · · , y(t), x(t). Thus, to
N
generate the signal d dty(t)
N , we simply take the signals from the corresponding
integrator blocks, multiply them by the coefficients, and add them all together.
The same holds true for the signal y[n + N ] in (3.5).
Chapter 4
Fourier Series
Representation of Periodic
Signals
Reading: Signals and Systems, Chapter 3.
In the last part of the course, we decomposed signals into sums of scaled and
time-shifted impulse functions. For LTI systems, we could then write the output
as a sum of scaled and time-shifted impulse responses (using the superposition
property). In this part of the course, we will consider alternate (and very useful)
decompositions of signals as sums of scaled complex exponential functions.
As we will see, such functions exhibit some nice behavior when applied to LTI
systems. This particular chapter will focus on decomposing periodic signals into
complex exponentials (leading to the Fourier Series), and subsequent chapters
will deal with the decomposition of more general signals.
4.1 Applying Complex Exponentials to LTI Sys-

tems
Recall that a complex exponential has the form x(t) = est (in continuous-time),
and x[n] = z n (in discrete-time), where s and z are general complex numbers.
Let’s start with a continuous-time LTI system with impulse response h(t). When
we apply the complex exponential x(t) = est to the system, the output is given
38 Fourier Series Representation of Periodic Signals
by
Z ∞ Z ∞
y(t) = x(t) ∗ h(t) = h(τ )x(t − τ )dτ = h(τ )es(t−τ ) dτ
−∞ −∞
Z ∞
=e st
h(τ )e−sτ dτ.
−∞
Let us define
Z ∞
H(s) = h(τ )e−sτ dτ.
−∞
If, for the given complex number s, the above integral exists (i.e., is finite), then
H(s) is just some complex number. Thus, we see that for an LTI system, if we
apply the complex exponential x(t) = est as an input, we obtain the quantity
y(t) = H(s)est
as an output. In other words, we get the same complex exponential out of the
system, just scaled by the complex number H(s). Thus, the signal est is called
an eigenfunction of the system, with eigenvalue H(s).
The same reasoning applies for discrete-time LTI systems. Consider an LTI
system with impulse response h[n], and input x[n] = z n . Then,
∞
X ∞
X
y[n] = x[n] ∗ h[n] = h[k]x[n − k] = h[k]z n−k
k=−∞ k=−∞
∞
X
=z n
h[k]z −k .
k=−∞
Let us define
∞
X
H(z) = h[k]z −k .
k=−∞
If this sum converges for the given choice of complex number z, then H(z) is just
some complex number. Thus, we see again that for a discrete-time LTI system
with the complex exponential x[n] = z n as an input, we obtain the quantity
y[n] = H(z)z n
as an output. In this case z n is an eigenfunction of the system, and H(z) is the

eigenvalue.
So, to summarize, we have the following:
4.1 Applying Complex Exponentials to LTI Systems 39
• If the signal x(t) = est is applied to an LTI system with

impulse response R ∞h(t), the output is y(t) = H(s)est ,
−sτ
where H(s) = −∞ h(τ )e dτ (assuming the integral
exists).
• If the signal x[n] = z n is applied to an LTI system with

impulse responseP∞h[n], the −k output is y[n] = H(z)z n ,
where H(z) = k=−∞ h[k]z (assuming the sum con-
verges).
As we will see later in the course, the quantities H(s) and H(z) are the Laplace
Transform and z-Transform of the impulse response of the system, respec-
tively.
Note that the above translates to superpositions P of complex exponentials in an
n
natural way. Specifically, if the input is x(t) = i=1 ai esi t for some complex
numbers a1 , . . . , an and s1 , . . . , sn , we have
n
X
y(t) = ai H(si )esi t .
i=1
An essentially identical relationship is true for discrete-time systems.

Example 4.1. Consider the signal x(t) = cos(ωt). We can write this as
1 jωt 1 −jωt
x(t) = e + e .
2 2
Thus, the output will be
1 1
y(t) = H(jω)ejωt + H(−jω)e−jωt .
2 2
Now, suppose that the impulse response of the system is real-valued (i.e., h(t) ∈
R for all t). Then, we have
Z ∞ Z ∞
∗ jωτ ∗
h(τ )e−jωτ dτ = H(jω).

H(−jω) = h(τ )e dτ =
−∞ −∞
Thus, for real-valued impulse responses, we have H(−jω) = H(jω)∗ . We can

equivalently write these in polar form as
H(jω) = |H(jω)|ej∠H(jω) , H(−jω) = |H(jω)|e−j∠H(jω) .
Thus,
1 1
y(t) = H(jω)ejωt + H(−jω)e−jωt
2 2
1 1
= |H(jω)|e e + |H(jω)|e−j∠H(jω) e−jωt
j∠H(jω) jωt
2 2
= |H(jω)| cos(ωt + ∠H(jω)).
Example 4.2. Consider the system y(t) = x(t−t0 ), where t0 ∈ R. The impulse
response if this system is h(t) = δ(t − t0 ), and thus
Z ∞ Z ∞
H(s) = h(t)e−st dt = δ(t − t0 )e−st dt = e−st0 .
−∞ −∞
Suppose we apply the signal x(t) = cos(ω0 t) to the system. We expect the
output to be cos(ω0 (t − t0 )), based on the definition of the system. Let’s verify
this using the identities we derived earlier. We have
|H(jω0 )| = |e−jω0 t0 | = 1, ∠H(jω0 ) = −ω0 t0 .
Thus, when we apply the input cos(ω0 t), the output is given by
y(t) = |H(jω0 )| cos(ω0 t + ∠H(jω0 )) = cos(ω0 t − ω0 t0 ),
matching what we expect.
4.2 Fourier Series Representation of Continuous-

Time Periodic Signals
Consider the complex exponential signal
x(t) = ejω0 t .
2π
Recall that this signal is periodic with fundamental period T = ω0
(assuming
ω0 > 0). Based on this complex exponential, we can define an entire harmonic
family of complex exponentials, given by
φk (t) = ejkω0 t , k ∈ Z.
In other words, for each k ∈ Z, φk (t) is a complex exponential whose funda-

mental frequency is kω0 (i.e., k times the fundamental frequency of x(t)). Thus,
each of the signals φk (t) is periodic with period T , since
φk (t + T ) = ejkω0 (t+T ) = ejkω0 t ejkω0 T = ejkω0 t ejk2π = ejkω0 t .
Note that T may not be the fundamental period of the signal φk (t), however.
Since each of the signals in the harmonic family is periodic with period T , a
linear combination of signals from that family is also periodic. Specifically,
consider the signal
∞ ∞
2π
X X
x(t) = ak ejkω0 t = ak ejk T t .
k=−∞ k=−∞
The terms corresponding to k = 1 and k = −1 are known as the first harmonic

of the signal x(t). The terms corresponding to k = 2 and k = −2 are known as
the second harmonic and so forth.
4.2 Fourier Series Representation of Continuous-Time Periodic
Signals 41
Suppose we are given a certain periodic signal x(t) with fundamental period T .
Define ω0 = 2π
T and suppose that we can write x(t) as
∞ ∞
!
jk 2π
X X
x(t) = ak e T t = ak ejkω0 t
k=−∞ k=−∞
for some sequence of coefficients ak , k ∈ Z. Then the above representation is

called the Fourier Series representation of x(t). The quantities ak , k ∈ Z are
called the Fourier Series coefficients.
Example 4.3. Consider the signal x(t) = cos(ω0 t), where ω0 > 0. We have
1 jω0 t 1 −jω0 t
x(t) = e + e .
2 2
This is the Fourier Series representation of x(t); it has only first harmonics, with
coefficients a1 = a−1 = 21 .
Similarly, consider the signal x(t) = sin(ω0 t). We have
1 jω0 t 1
x(t) = e − e−jω0 t .
2j 2j
1
Once again, the signal has only first harmonics, with coefficients a1 = 2j and
1
a−1 = − 2j .
Suppose that we have a periodic signal that has a Fourier Series representation
∞
X
x(t) = ak ejkω0 t . (4.1)
k=−∞
Now suppose that x(t) is real, i.e., x∗ (t) = x(t). Taking the complex conjugate
of both sides of the above expression, we have
∞
X
x∗ (t) = a∗k e−jkω0 t .
k=−∞
Equating the expressions for x(t) and x∗ (t), we have

∞
X ∞
X
ak ejkω0 t = a∗k e−jkω0 t .
k=−∞ k=−∞
Comparing the terms, we see that for any k ∈ Z, the coefficient of ejkω0 t is ak
on the left hand side, and is a∗−k on the right hand side. Thus, for real signals
x(t), the Fourier Series coefficients satisfy
a−k = a∗k
for all k ∈ Z. Substituting this into the Fourier Series representation (4.1) we
have
∞
X
ak ejkω0 t + a−k e−jkω0 t

x(t) = a0 +
k=1
X∞
ak ejkω0 t + a∗k e−jkω0 t

= a0 +
k=1
X∞
2Re ak ejkω0 t ,

= a0 +
k=1
where Re is the real part of the given complex number. If we write ak in polar
form as rk ejθk , the above expression becomes
∞
X n o ∞
X
x(t) = a0 + 2Re rk ej(kω0 t+θk ) = a0 + 2 rk cos(kω0 t + θk ).
k=1 k=1
This is an alternate representation of the Fourier Series for real-valued signals

(known as the trigonometric representation).
4.3 Calculating the Fourier Series Coefficients

Suppose that we are given a periodic signal x(t) with period T , and that this
signal has a Fourier Series representation
∞
X
x(t) = ak ejkω0 t .
k=−∞
We will soon see conditions under which a signal will have such a representation,
but for now, suppose that we are just interested in finding the coefficients ak ,
k ∈ Z. To do this, multiply x(t) by e−jnω0 t , where n is some integer. This gives
∞
X ∞
X
−jnω0 t jkω0 t −jnω0 t
x(t)e = ak e e = ak ej(k−n)ω0 t .
k=−∞ k=−∞
Now suppose that we integrate both sides of the above equation from t0 to t0 +T
for any t0 :
Z t0 +T ∞
Z t0 +T X ∞
X Z t0 +T
x(t)e−jnω0 t dt = ak ej(k−n)ω0 t dt = ak ej(k−n)ω0 t dt.
t0 t0 k=−∞ k=−∞ t0
Now note that if n = k, we have

Z t0 +T Z t0 +T
j(k−n)ω0 t
e dt = 1dt = T.
t0 t0
4.3 Calculating the Fourier Series Coefficients 43
Otherwise, if n 6= k, we have
Z t0 +T t0 +T
1
ej(k−n)ω0 t dt = ej(k−n)ω0 t

t0 j(k − n)ω0 t0
1
= ej(k−n)ω0 (t0 +T ) − ej(k−n)ω0 t0
j(k − n)ω0
= 0.
Thus, we have
Z t0 +T ∞
X Z t0 +T
−jnω0 t
x(t)e dt = ak ej(k−n)ω0 t dt = an T,
t0 k=−∞ t0
or equivalently,
1 t0 +T
Z
an =x(t)e−jnω0 t dt
T t0
where t0 is any arbitrary starting point. In other words, we obtain the Fourier
coefficient an by multiplying the signal x(t) by e−jnω0 t and then integrating the
resulting product over any period.
Example 4.4. Consider the signal

0
 − T2 ≤ t < T1
x(t) = 1 −T1 ≤ t ≤ T1 ,
T1 < t < T2

0

where T1 ≤ T and x(t) is T -periodic.

2π
Define ω0 = T . We have
T
Z Z T1
1 2 1 2T1
a0 = x(t)dt = x(t)dt = .
T − T2 T T1 T
For k 6= 0, we have
Z T2
1 T1 −jkω0 t
Z
1 −jkω0 t 1 1 T1
ak = x(t)e dt = e dt = e−jkω0 t −T1
T − T2 T −T1 T −jkω0
2
= sin(kω0 T1 )
T kω0
sin(kω0 T1 )
= ,
kπ
where we used the fact that T ω0 = 2π.
Note that in this case, ak is real for all k ∈ Z, and satisfies ak = a−k . Thus we
can also write the Fourier series as
∞
X X∞
ak ejkω0 t + e−jkω0 t = a0 + 2

x(t) = a0 + ak cos(kω0 t).
k=1 k=1
4.3.1 A Vector Analogy for the Fourier Series
In the derivation of the Fourier series coefficients, we saw that

Z t0 +T
ejkω0 t e−jnω0 t dt = 0
t0
if k 6= n, and is equal to T otherwise. The functions ejkω0 and ejnω0 (for k 6= n)

are said to be orthogonal. More generally, a set of functions φk (t), k ∈ Z, are
said to be orthogonal on an interval [a, b] if
Z b
φk (t)φ∗n (t) = 0
a
if k 6= n, and nonzero otherwise. Note that φ∗n (t) is the complex conjugate
of φn (t). We then derived the expressions for the coefficients by using the
orthogonality property. However, that derivation assumed that the signal could
be written as a linear combination of the functions in the harmonic family, and
then derived the coefficient expressions. Here will justify this by first trying to
approximate a given signal by a finite number of functions from the harmonic
family and then taking the number of approximating functions to infinity. We
will start by reviewing how to approximate a given vector by other vectors, and
then explore the analogy to the approximation of functions.
Review of Approximation of Vectors
The above definition of orthogonality of functions is exactly analogous to the

definition of orthogonal vectors in a vector space. Recall that two vectors
v1 , v2 ∈ Rn are said to be orthogonal if v10 v2 = 0. Suppose we are given the
vectors
     
1 1 0
x = 2 , v1 = 0 , v2 = 1 .
3 0 1
Note that v10 v2 = 0 and thus v1 and v2 are orthogonal. Suppose we wish to
approximate the vector x using a linear combination of the vectors v1 and v2 .
In other words, we wish to find coefficients a and b so that the approximation
x̂ = av1 + bv2
is “close” to x. A typical metric of “closeness” is taken to be the square of the

approximation error. Specifically, the approximation error is given by
e = x − x̂ = x − av1 − bv2 .
Note that e is a vector, where the i-th component is the approximation error
for the i-th component of x. We try to minimize e21 + e22 + e23 , which is given by
e21 + e22 + e23 = e0 e = (x − av1 − bv2 )0 (x − av1 − bv2 )
= x0 x − ax0 v1 − bx0 v2 − av10 x + a2 v10 v1 + abv10 v2
− bv20 x + abv20 v1 + b2 v22 .
Noting that v1 and v2 are orthogonal, we have
e0 e = x0 x − ax0 v1 − bx0 v2 − av10 x + a2 v10 v1 − bv20 x + b2 v22 .
This is a convex function of the scalars a and b. If we wish to minimize e0 e, we
take the derivative with respect to these scalars and set it equal to zero. This
yields
∂e0 e
= −x0 v1 − v10 x + 2av10 v1 = 0
∂a
v0 x
⇒ a = 01
v1 v1
0
∂e e
= −x0 v2 − v20 x + 2bv20 v2 = 0
∂b
v0 x
⇒ b = 02 ,
v2 v2
where we used the fact that x0 v1 = v10 x and x0 v2 = v20 x (since these quantities
are all scalars). In terms of the vectors given above, we obtain
1 5
a= = 1, b = .
1 2
Application to Approximation of Functions
Entirely analogous ideas hold when we are trying to approximate one function
as a linear combination of a set of orthogonal functions (as in the Fourier series).
Given a set of orthogonal functions φk (t) over an interval [a, b], suppose we wish
to approximate a given function x(t) as a linear combination of some finite
number of these functions. Specifically, suppose that we are given some positive
integer N , and wish to find the best coefficients αk ∈ C such that the estimate
N
X
x̂(t) = ak φk (t)
k=−N
is “close” to x(t). Mathematically, we will use the notion of squared error to

measure “closeness.” Specifically, the approximation error at any given point in
time t is given by
N
X
e(t) = x(t) − x̂(t) = x(t) − ak φk (t),
k=−N
and the squared error over the entire interval [a, b] is then defined as
Z b
|e(t)|2 dt.
a
Here, we will allow e(t) to a general complex valued function, so the absolute
value in the integral is interpreted as the magnitude of the complex number
e(t), i.e., the square error over the interval [a, b] is given by
Z b
e∗ (t)e(t)dt.
a
Consider the harmonic family φk (t) = ejkω0 t , and suppose that we wish to find
the best approximation of a given T -periodic signal x(t) as a linear combination
of φk (t) for −N ≤ k ≤ N , i.e.,
N
X
x̂(t) = ak ejkω0 t ,
k=−N
with error
N
X
e(t) = x(t) − x̂(t) = x(t) − ak ejkω0 t .
k=−N
We evaluate the squared error over any interval of length T (since the functions
φk (t) are orthogonal over such intervals):
Z t0 +T Z t0 +T
Squared Error = |e(t)|2 dt = e∗ (t)e(t)dt
t0 t0
N
! N
!
Z t0 +T X X
∗
= x (t) − a∗k e−jkω0 t x(t) − ak e jkω0 t
dt
t0 k=−N k=−N
N N
!
Z t0 +T X X
∗ ∗
= x (t)x(t) − x (t) ak e jkω0 t
− x(t) a∗k e−jkω0 t dt
t0 k=−N k=−N
N N
!
Z t0 +T X X
+ a∗k e−jkω0 t ak e jkω0 t
dt
t0 k=−N k=−N
N N
!
Z t0 +T Z t0 +T X X
∗
= 2
|x(t)| dt + −x (t) ak e jkω0 t
− x(t) a∗k e−jkω0 t dt
t0 t0 k=−N k=−N
N N
!
Z t0 +T X X
+ a∗k an e−jkω0 t ejnω0 t dt
t0 k=−N k=−N
Z t0 +T N
X Z t0 +T N
X Z t0 +T
∗
= 2
|x(t)| dt − ak x (t)e jkω0 t
dt − a∗k x(t)e−jkω0 t dt
t0 k=−N t0 k=−N t0
N N
!
X X Z t0 +T
+ a∗k an e−jkω0 t ejnω0 t dt
k=−N n=−N t0
Z t0 +T N
X Z t0 +T N
X Z t0 +T
∗
= 2
|x(t)| dt − ak x (t)e jkω0 t
dt − a∗k x(t)e−jkω0 t dt
t0 k=−N t0 k=−N t0
N
X
+T |ak |2 ,
k=−N
R t0 +T
where we used the fact that t0
e−jkω0 t ejnω0 t dt = 0 if k 6= n and T otherwise.
Our job is to find the best coefficients ak , −N ≤ k ≤ N to minimize the square
error. Thus, we first write ak = bk +jck , where bk , ck ∈ R, and then differentiate
the above expression with respect to bk and ck and set the result to zero. After
some algebra, we obtain the optimal coefficient as
Z t0 +T
1
ak = bk + jck = x(t)e−jkω0 t dt,
T t0
which is exactly the same expression we found for the Fourier series coefficients
earlier. Note again why we bothered to go through this exercise. Here, we
did not assume that a signal x(t) had a Fourier series representation; we sim-
ply asked how to best approximate a given signal by a linear combination of
complex exponentials, and found the resulting coefficients. These coefficients
match exactly the coefficients that we obtained by assuming that the signal had
a Fourier series representation, and lends some justification for the validity of
the earlier analysis.
As N gets larger, the approximation error will get smaller and smaller. The
R t +T
question is then: will t00 |e(t)|2 dt go to zero as N goes to infinity? If so,
then the signal would, in fact, have a Fourier series representation (in the sense
of having asymptotically zero error between the true signal and the approxima-
tion). It turns out that most periodic signals of practical interest will satisfy
this property.
When Will a Periodic Signal Have a Fourier Series Representation?
There are various different sufficient conditions that guarantee that a given
signal x(t) will have a Fourier series representation. One commonly used set of
conditions are known as the Dirichlet conditions stated as follows.
A periodic signal x(t) has a Fourier series representation if all three of the
following conditions are satisfied:
• The signal is absolutely integrable over one period:

Z t0 +T
|x(t)|dt < ∞.
t0
• In any finite interval of time x(t) has bounded variation, meaning that it
has only a finite number of minima and maxima during any single period
of the signal.
• In any finite interval of time, there are only a finite number of discontinu-
ities, and each of these discontinuities are finite.
We won’t go into the proof of why these conditions are sufficient here, but it
suffices to note that signals that violate the above conditions (the last two in
particular) are somewhat pathological. The first condition guarantees that the
Fourier series coefficients are finite, since
Z
1 t0 +T 1 t0 +T
Z
1 t0 +T
Z
−jkω0 t
x(t)e−jkω0 t dt =

|ak | = x(t)e dt ≤ |x(t)| dt.

T t0 T t0 T t0
Thus if the signal is absolutely integrable over a period, then |ak | < ∞ for all
k ∈ Z.
Gibbs Phenomenon
If x(t) has a Fourier series representation, then that representation will exactly
equal x(t) at all points t where x(t) is continuous. At points of discontinuity in
x(t), the value of the Fourier series representation will be equal to the average of
the values of x(t) on either side of the discontinuity. One particularly interest-
ing phenomenon occurs at points of discontinuity: the Fourier series typically
overshoots the signal x(t). The height of the overshoot stays constant as the
number of terms N in the approximation increases, but the width shrinks. Thus,
asymptotically, the error goes to zero (although technically the two signals are
not exactly the same at the discontinuity).
4.4 Properties of Continuous-Time Fourier Se-

ries
We will now derive some useful properties of the Fourier series coefficients.
Throughout we will use the notation
FS
x(t) ←→ ak
to denote that the T -periodic signal x(t) has Fourier series coefficients ak , k ∈ Z.
4.4 Properties of Continuous-Time Fourier Series 49
4.4.1 Linearity
Suppose we have two signals x1 (t) and x2 (t), each of which is periodic with
period T . Let
FS FS
x1 (t) ←→ ak , x2 (t) ←→ bk .
For any complex scalars α, β, let g(t) = αx1 (t) + βx2 (t). Then
FS
g(t) ←→ αak + βbk .
The above property follows immediately from the definition of the Fourier series
coefficients (since integration is linear).
Example 4.5. Consider x1 (t) = cos(ω0 t) and x2 (t) = sin(ω0 t). We have
α jω0 t α −jω0 t β β
g(t) = α cos(ω0 t) + β sin(ω0 t) = e + e + ejω0 t − e−jω0 t
2 2 2j 2j

α β α β
= + e jω0 t
+ − e−jω0 t .
2 2j 2 2j
Thus, we see that each Fourier series coefficient of g(t) is indeed given by a
linear combination of the corresponding Fourier series coefficients of x1 (t) and
x2 (t).
4.4.2 Time Shifting

Define g(t) = x(t − τ ), where τ ∈ R is some delay. Let the Fourier series
coefficients of g(t) be given by bk , k ∈ Z. Then
Z t0 +T Z t0 +T
1 −jkω0 t 1
bk = g(t)e dt = x(t − τ )e−jkω0 t dt.
T t0 T t0
dt̄
Define t̄ = t − τ , so that dt = 1. Then we have
Z t0 −τ +T Z t0 −τ +T
1 −jkω0 (t̄+τ ) −jkω0 τ 1
bk = x(t̄)e dt̄ = e x(t̄)e−jkω0 t̄ dt̄ = e−jkω0 τ ak .
T t0 −τ T t0 −τ
Thus
FS
x(t − τ ) ←→ e−jkω0 τ ak .
Example 4.6. Consider x(t) = cos(ω0 t), and let g(t) = x(t − τ ) We have
1 jω0 (t−τ ) 1 −jω0 (t−τ ) 1 1

g(t) = cos(ω0 (t−τ )) = e + e = e−jω0 τ ejω0 t + ejω0 τ e−jω0 t .
2 2 2 2
Thus, we see that the coefficient of ejω0 t is 12 e−jω0 τ , and the coefficient of e−jω0 t
is 21 ejω0 τ , as predicted by the expressions derived above.
4.4.3 Time Reversal

Define y(t) = g(−t). Let the Fourier series coefficients of g(t) be given by bk ,
k ∈ Z. Note that
∞
X ∞
X
g(t) = x(−t) = ak e−jkω0 t = a−k ejkω0 t .
k=−∞ k=−∞
Thus we have
FS
x(−t) ←→ a−k ,
i.e., the Fourier series coefficients for a time-reversed signal are just the time-
reversal of the Fourier series coefficients for the original signal. Note that this
is not true for the output of LTI systems (a time reversal of the input to an
LTI system does not necessarily mean that the output is a time-reversal of the
original output).
Example 4.7. Consider x(t) = sin(ω0 t) and define g(t) = x(−t). First, note
that
1 1
x(t) = sin(ω0 t) = ejω0 t − e−jω0 t = a1 ejω0 t + a−1 e−jω0 t .
2j 2j
We have
1 1
g(t) = sin(−ω0 t) = e−jω0 t − ejω0 t = b1 ejω0 t + b−1 e−jω0 t ,
2j 2j
1 1
and thus we see that b1 = − 2j = a−1 and b−1 = 2j = a1 .
4.4.4 Time Scaling

Consider the signal g(t) = x(αt), where α ∈ R>0 . Thus, g(t) is a time-scaled
version of x(t). Note that the period of g(t) is T2 , where T is the period of x(t).
We have
X∞
g(t) = x(αt) = ak ejkω0 αt .
k=−∞
Thus, g(t) has the same Fourier series coefficients as x(t), but the Fourier series
representation has changed: the frequency is now ω0 α rather than ω0 , to
reflect the fact that the harmonic family is in terms of the new period Tα rather
than T .
Example 4.8. Consider x(t) = cos(ω0 t), which has series representation
1 jω0 t 1 −jω0 t
x(t) = e + e .
2 2
Then we have
1 jω0 αt 1 −jω0 αt
g(t) = x(αt) = cos(ω0 αt) = e + e .
2 2
4.4 Properties of Continuous-Time Fourier Series 51
4.4.5 Multiplication
Let x1 (t) and x2 (t) be T -periodic signals with Fourier series ak and bk re-
spectively. Consider the signal g(t) = x1 (t)x2 (t), and note that g(t) is also
T -periodic. We have
∞
X ∞
X ∞
X ∞
X
g(t) = x1 (t)x2 (t) = al ejlω0 t bn ejnω0 t = al bn ej(l+n)ω0 t .
l=−∞ n=−∞ l=−∞ n=−∞
Define k = l + n, so that
∞ ∞ ∞ ∞
!
X X X X
g(t) = al bk−l ejlω0 t = al bk−l ejkω0 t .
l=−∞ l=−∞ k=−∞ l=−∞
Thus the Fourier series coefficients of g(t) are given by

∞
FS
X
g(t) = x1 (t)x2 (t) ←→ al bk−l .
l=−∞
In other words, the Fourier series coefficients of a product of two signals are
given by the convolution of the corresponding Fourier series coefficients.
Example 4.9. Consider the signals x1 (t) = cos(ω0 t) and x2 (t) = sin(ω0 t).
Define g(t) = x1 (t)x2 (t). The Fourier series representations of x1 (t) and x2 (t)
are given by
1 jω0 t 1 −jω0 t 1 jω0 t 1

x1 (t) = e + e , x2 (t) = e − e−jω0 t .
2 2 2j 2j
Denote the Fourier series coefficients of x1 (t) by the sequence ak , with a−1 =
a1 = 21 , and ak = 0 otherwise. Similarly, denote the Fourier series coefficients
1 1
of x2 (t) by the sequence bk , with b−1 = − 2j , b(1) = 2j , and bk = 0 otherwise.
Denote the Fourier series coefficients of g(t) by ck , k ∈ Z. Then we have
∞
X
ck = ak ∗ bk = al bk−l .
l=−∞
1 1
Convolving the two sequences given above, we see that c−2 = − 4j , c2 = 4j , and
ck = 0 otherwise. Thus
1 j2ω0 t 1
g(t) = x1 (t)x2 (t) = cos(ω0 t) sin(ω0 t) = e − e−j2ω0 t .
4j 4j
1
Noting that cos(ω0 t) sin(ω0 t) = 2 sin(2ω0 t), we see that the above expression is,
in fact, correct.
4.4.6 Parseval’s Theorem

Consider a T -periodic signal x(t), and let ak , k ∈ Z be its Fourier series coeffi-
cients. Now let us consider the average power of x(t) over one period, defined
as
1 t0 +T
Z
|x(t)|2 dt.
T t0
Substituting |x(t)|2 = x∗ (t)x(t) and the Fourier series for x(t) we have
∞ ∞
! !
1 t0 +T 1 t0 +T
Z Z X X
2 ∗ −jkω0 t ∗ jnω0 t
|x(t)| dt = ak e an e dt
T t0 T t0 n=−∞
k=−∞
∞ ∞
1 t0 +T −jkω0 t jnω0 t
X X Z
∗
= ak an e e dt.
n=−∞
T t0
k=−∞
jkω0 t
Using the fact that e and ejnω0 t are orthogonal for k 6= n, we have
∞ ∞
1 t0 +T
Z X X
|x(t)|2 dt = a∗k ak = |ak |2 .
T t0
k=−∞ k=−∞
This leads to Parseval’s Theorem: for a T -periodic signal x(t) with Fourier
series coefficients ak , k ∈ Z, we have
Z t0 +T ∞
1 X
|x(t)|2 dt = |ak |2 .
T t0 k=−∞
Example 4.10. Consider the signal x(t) = cos(ω0 t), with Fourier coefficients
a1 = a−1 = 12 and ak = 0 otherwise. We have
1 t0 +T 1 t0 +T 1 t0 +T 1
Z Z Z
1
|x(t)|2 dt = cos2 (ω0 t)dt = (1 + cos(2ω0 t))dt = .
T t0 T t0 T t0 2 2
We also have
∞
X 1 1 1
|ak |2 = a21 + a2−1 = + = ,
4 4 2
k=−∞
which agrees with the direct calculation of average power, as indicated by Par-
seval’s Theorem.
4.5 Fourier Series for Discrete-Time Periodic Sig-

nals
We now turn our attention to the discrete-time Fourier series. Specifically,
consider the discrete-time signal x[n], n ∈ Z, and suppose it is N -periodic. Let
4.5 Fourier Series for Discrete-Time Periodic Signals 53
ω0 = 2πN , and note that the signal φk [n] = e

jkω0 n
is also N -periodic for any
k ∈ Z. Thus, as in the case for continuous-time signals, we would like to write
x[n] as a linear combination of signals from the harmonic family φk [n], k ∈ Z,
i.e.,
∞ ∞
2π
X X
x[n] = ak ejkω0 n = ak ejk N n .
k=−∞ k=−∞
At this point, we encounter the first main difference between the discrete-time
and continuous-time Fourier series. Recall that a discrete-time complex expo-
nential with frequency ω0 is the same as a discrete-time complex exponential
with frequency ω0 + 2π. Specifically, for any k ∈ Z, consider
φk+N [n] = ej(k+N )ω0 n = ejkω0 n ejN ω0 n = ejkω0 n ,
since N ω0 = 2π. Thus, there are only N different complex exponentials in the
discrete-time harmonic family for the fundamental frequency ω0 , and so we have
the following.
The discrete-time Fourier series for an N -periodic signal

x[n] is given by
X−1
n0 +N X−1
n0 +N
2π
x[n] = ak ejkω0 n = ak ejk N n (4.2)
k=n0 k=n0
where n0 is any integer.
In other words, the discrete-time signal can be written in terms of any N con-
tiguous multiples of the fundamental frequency ω0 . Since there are only N
coefficients ak in this representation, and since it does not matter which con-
tiguous N members of the harmonic family we choose, the Fourier series
coefficients are N -periodic as well, i.e., ak = ak+N for all k ∈ Z.
4.5.1 Finding the Discrete-Time Fourier Series Coefficients

To find the Fourier series coefficients in (4.2), we use a similar trick as in the
continuous-time case. Specifically, first multiply both sides of (4.2) by e−jrω0 n ,
where r is any integer, and sum both sides over N terms. This gives
X−1
n1 +N X−1 n0 +N
n1 +N X−1
x[n]e−jrω0 n = ak ej(k−r)ω0 n ,
n=n1 n=n1 k=n0
where n1 is any integer. Interchanging the summations, we have

X−1
n1 +N X−1
n0 +N X−1
n1 +N
x[n]e−jrω0 n = ak ej(k−r)ω0 n .
n=n1 k=n0 n=n1
Now suppose that k − r is a multiple of N . In this case we obtain

X−1
n1 +N X−1
n1 +N
ej(k−r)ω0 n = 1 = N.
n=n1 n=n1
On the other hand, if r − k is not a multiple of N , we use the finite sum formula
to obtain
X−1
n1 +N
ej(k−r)ω0 n1 − ej(k−r)ω0 (n1 +N )
ej(k−r)ω0 n = = 0.
n=n1
ej(k−r)ω0 − 1
Thus, we have the following.
The discrete-time Fourier series coefficients are given by
1 X−1
n1 +N
ak = x[n]e−jkω0 n
N n=n1
where n1 is any integer. The Fourier series coefficients are

N -periodic, i.e., ak = ak+N .
Example 4.11. Consider the N -periodic signal x[n], which is equal to 1 for
−N1 ≤ n ≤ N1 and zero otherwise (modulo the periodic constraints).
We have
N1
1 X
ak = e−jkω0 n .
N
n=−N1
2N1 +1
If k = 0, we have a0 = N . For k ∈ {1, 2, . . . , N − 1}, we have (via the finite
sum formula)
1 1 1
1 ejkω0 N1 − ejkω0 (N1 +1) 1 e−jkω0 2 ejkω0 (N1 + 2 ) − ejkω0 (N1 + 2 )
ak = =
N 1 − e−jkω0 N e−jkω0 12 1 1
ejkω0 2 − e−jkω0 2
1 sin(kω0 (N1 + 21 ))
= .
N sin(k ω20 )
It is of interest to note that we do not have to worry about convergence con-

ditions for discrete-time Fourier series, as we did in the continuous-time case.
Specifically, for an N -periodic discrete-time signal x[n], we only require N num-
bers to completely specify the entire signal. The Fourier series coefficients ak ,
k ∈ {0, 1, . . . , N − 1} thus contain as much information as the signal itself, and
form a perfect representation of the signal. In other words, for discrete-time sig-
nals, the Fourier series representation is just a transformation of the signal into
another form; we do not encounter discrepancies like the Gibbs phenomenon in
discrete-time.
4.5.2 Properties of the Discrete-Time Fourier Series
The discrete-time Fourier series has properties that can be derived in almost
the same way as the properties for the continuous-time Fourier series (linearity,
time-shifting, etc.). Here we will just discuss the multiplication property and
Parseval’s theorem.
Multiplication of Discrete-Time Periodic Signals
First, let’s start with an example. Consider two 2-periodic signals x1 [n] and
x2 [n]. We know that the Fourier series respresentations can be uniquely specified
by two coefficients. Specifically,
x1 [n] = a0 + a1 ejω0 n , x2 [n] = b0 + b1 ejω0 n .
Now consider the product
g[n] = x1 [n]x2 [n] = (a0 + a1 ejω0 n )(b0 + b1 ejω0 n )

= a0 b0 + (a0 b1 + a1 b0 )ejω0 n + a1 b1 ej2ω0 n .
2π
Now, note that ω0 = 2 , and thus ej2ω0 n = 1. This gives
g[n] = (a0 b0 + a1 b1 ) + (a0 b1 + a1 b0 )ejω0 n .
Thus, the Fourier series coefficients of g[n] are given by c0 = a0 b0 + a1 b1 and

c1 = a0 b1 + a1 b0 . We can write these in a unform way as follows:
1
X
c0 = a0 b0 + a1 b1 = a0 b0 + a1 b−1 = al b−l
l=0
1
X
c1 = a0 b1 + a1 b0 = al b1−l ,
l=0
where we used the fact that b1 = b−1 by the periodic nature of the discrete-time
Fourier series coefficients. The above expressions show that the Fourier series
coefficients of the product of the two signals are given by a form of convolution of
the coefficients of those signals; however, the convolution is over a finite number
of terms, as opposed to over all time-indices.
Let us generalize this to functions with a larger period. Let x1 [n] and x2 [n]
be two N -periodic discrete-time signals, with discrete-time Fourier series coef-
ficients ak and bk , respectively. We have
N
X −1 N
X −1
x1 [n] = ak ejkω0 n , x2 [n] = bk ejkω0 n .
k=0 k=0
Define the signal g[n] = x1 [n]x2 [n]. We have

−1 −1
N
! N
!
X X
g[n] = x1 [n]x2 [n] = al ejlω0 n br ejrω0 n
l=0 r=0
N
X −1 N
X −1
= al br ej(l+r)ω0 n ,
l=0 r=0
where we have used l and r as the indices in the Fourier series in order to keep
the terms in the two sums distinct.
Define the new variable k = l + r. Substituting into the above expression, this
gives
N
X −1 l+N
X−1
g[n] = al bk−l ejkω0 n
l=0 k=l
−1 −1 X−1
N N l+N
!
X X
jkω0 n jkω0 n
= al bk−l e + al bk−l e
l=0 k=l k=N
−1 −1
N N l−1
!
X X X
jkω0 n j(k+N )ω0 n
= al bk−l e + al bk+N −l e
l=0 k=l k=0
−1 −1
N N l−1
!
X X X
jkω0 n jkω0 n
= al bk−l e + al bk−l e
l=0 k=l k=0
N
X −1 N
X −1
= al bk−l ejkω0 n
l=0 k=0
−1 −1
N N
!
X X
= al bk−l ejkω0 n
k=0 l=0
Thus, the Fourier series coefficients of g[n] are given by

N
X −1
ck = al bk−l
l=0
for 0 ≤ k ≤ N − 1. The above convolution is known as the periodic convolu-

tion of two periodic signals; for any k, there are only N terms in the summation.
We saw an example of this with the 2-periodic signals that we had above. In
essence, the above calucations are simply multiplying together the discrete-time
Fourier series representations of the two signals (each of which has N terms),
and then using the periodicity of the discrete-time complex exponentials in the
frequencies to combine terms together. Note that we can actually perform the
sum over any continguous N values of l, since all of the signals involved are
periodic.
Parseval’s Theorem for Discrete-Time Signals
Let x[n] be an N -periodic discrete-time signal, with Fourier series coefficients

ak , 0 ≤ k ≤ N − 1. The average power of x[n] over one period is
1 X−1
n0 +N
|x[n]|2 ,
N n=n0
where n0 is any integer. Parseval’s theorem for discrete-time signals states the
following.
1 X−1
n0 +N N
X −1
2
|x[n]| = |ak |2 .
N n=n0 k=0
The following example illustrates the application of the various facts that we
have seen about the discrete-time Fourier series.
Example 4.12. Suppose we are told the following facts about a discrete-time
signal x[n].
• x[n] is periodic with period N = 6.

P5
• n=0 x[n] = 2.
P7 n
• n=2 (−1) x[n] = 1.
• x[n] has the minimum power per period of all signals satisfying the pre-
ceding three conditions.
The above facts are sufficient for us to uniquely determine x[n]. First, note that
the Fourier series representation of x[n] is
5 5
π
X X
jkω0 n
x[n] = ak e = ak ejk 3 n ,
k=0 k=0
where we used the fact that the signal is 6-periodic.

From the second fact, we have
N −1
1 X 2 1
a0 = x[n] = = .
N 6 3
k=0
π
To use the third fact, note that (−1)n = e−jπn = e−j3 3 n . Thus, the third fact
seems to be related to the Fourier series coefficient for k = 3. Specifically, we
have
7
1 X 1
a3 = x[n]e−jπn = .
N 6
k=2
To use the last fact, note from Parseval’s theorem that the power of the signal
over one period is given by
5 5
1 X X
|x[n]|2 = |ak |2
N n=0
k=0
= |a0 |2 + |a1 |2 + |a2 |2 + |a3 |2 + |a4 |2 + |a5 |2 .
We are told that x[n] has the minimum average power over all signals that
satisfy the other three conditions. Since the other three conditions have already
set a0 and a1 , the average power is given by setting all of the other Fourier series
coefficients to 0. Thus, we have
5
X π 1 1
x[n] = ak ejk 3 n = a0 + a3 ejπn = + (−1)n .
3 6
k=0
Chapter 5
The Continuous-Time
Fourier Transform
Reading: Signals and Systems, Chapter 4.1-4.6.

In the last part of the course, we decomposed periodic signals into superpositions
of complex exponentials, where each complex exponential is a member of the
harmonic family corresponding to the fundamental period of the signal. We
now turn our attention to the case where the signal of interest is not periodic.
As we will see, the main idea will be to view an aperiodic signal as a periodic
signal whose period goes to ∞.
5.1 The Fourier Transform

Suppose that we are given a signal x(t) that is aperiodic. As a concrete example,
suppose that x(t) is a solitary square pulse, with x(t) = 1 if −T1 ≤ t ≤ T1 , and
zero elsewhere. Clearly x(t) is not periodic.
Now define a new signal x̃(t) which is a periodic extension of x(t) with period
T . In other words, x̃(t) is obtained by repeating x(t), where each copy is shifted
T units in time. This x̃(t) has a Fourier series representation, which we found
in the last chapter to be
2T1 2 sin(kω0 T1 )
a0 = , ak = ,
T kω0 T
for k ∈ Z.
Now recall that the Fourier series coefficients are calculated as follows:
Z T2
1
ak = x̃(t)e−jkω0 t dt.
T − T2
60 The Continuous-Time Fourier Transform
However, we note that x(t) = x̃(t) in the interval of integration, and thus
Z T
1 2
ak = x(t)e−jkω0 t dt.
T − T2
Furthermore, since x(t) is zero for all t outside the interval of integration, we
can expand the limits of the integral to obtain
Z ∞
1
ak = x(t)e−jkω0 t dt.
T −∞
Let us define
Z ∞
X(jω) = x(t)e−jωt dt.
−∞
This is called the Fourier transform of the signal x(t), and the Fourier series
coefficients can be viewed as samples of the Fourier transform, scaled by T1 , i.e.,
1
ak = X(jkω0 ), k ∈ Z.
T
Now consider the fact that

∞ ∞
X 1 X
x̃(t) = ak ejkω0 t = X(jkω0 )ejkω0 t .
T
k=−∞ k=−∞
2π
Since ω0 = T , this becomes
∞
1 X
x̃(t) = X(jkω0 )ejkω0 t ω0 .
2π
k=−∞
Now consider what happens as the period T gets bigger. In this case, x̃(t)
approaches x(t), and so the above expression becomes a representation of x(t).
As T → ∞, we have ω0 → 0. Since each term in the summand can be viewed
as the area of the rectangle whose height is X(jkω0 )ejkω0 t and whose base goes
from kω0 to (k + 1)ω0 , we see that as ω0 → 0, the sum on the right hand side
approaches the area underneath the curve X(jω)ejωt (where t is held fixed).
Thus, as T → ∞ we have
Z ∞
1
x(t) = X(jω)ejωt dω.
2π −∞
Thus we have the following.

5.1 The Fourier Transform 61
Given a continuous-time signal x(t), the Fourier

Transform of the signal is given by
Z ∞
X(jω) = x(t)e−jωt dt.
−∞
The Inverse Fourier Transform of the signal is given by

Z ∞
1
2π −∞
The Fourier transform X(jω) is also called the spectrum of the signal, as it
represents the contribution of the complex exponential of frequency ω to the
signal x(t).
Example 5.1. Consider the signal x(t) = e−at u(t), a ∈ R>0 . The Fourier
transform of this signal is
Z ∞ Z ∞
X(jω) = x(t)e−jωt dt = e−at e−jωt dt
−∞ 0
1 ∞
=− e−(a+jω)t

a + jω 0
1
= .
a + jω
To visualize X(jω), we plot its magnitude and phase on separate plots (since
X(jω) is complex-valued in general). We have
1 ω
|X(jω)| = √ , ∠X(jω) = − tan−1 .
a + ω2
2 a
The plots of these quantities are show in Fig. 4.5 of the text.
Example 5.2. Consider the signal x(t) = δ(t). We have
Z ∞
X(jω) = δ(t)e−jωt dt = 1.
−∞
In other words, the spectrum of the impulse function has an equal contribution
at all frequencies.
Example 5.3. Consider the signal x(t) which is equal to 1 for −T1 ≤ t ≤ T1
and zero elsewhere. We have
Z T1
1 2 sin(ωT1 )
X(jω) = e−jωt dt = ejωT1 − e−jωT1 = .
−T1 jω ω
Example 5.4. Consider the signal whose Fourier transform is

(
1, |ω| ≤ W
X(jω) = .
0 |ω| > W
We have
Z ∞ Z W
1 1
x(t) = X(jω)ejωt dω = ejωt dω
2π −∞ 2π −W
1 1 jωt W
= e −W
2π jt
sin(W t)
= .
πt
The previous two examples showed the following. When x(t) is a square pulse,
then X(jω) = 2 sin(ωT
ω
1)
and when X(jω) is a square pulse, x(t) = sin(W
πt
t)
. This
is an example of the duality property of Fourier transforms, which we will see
later.
sin(W t)
Functions of the form πt will show up frequently, and are called sinc func-
tions. Specifically
sin(πθ)
sinc(θ) = .
πθ
2 sin(ωT1 ) ωT1
sin(W t) W Wt

Thus ω = 2T1 sinc π and πt = π sinc π .
5.1.1 Existence of Fourier Transform
Just as we saw with the Fourier series for periodic signals, there are some rather
mild conditions under which a signal x(t) is guaranteed to have a Fourier trans-
form (such that the inverse Fourier transform converges to the true signal).
Specifically, there are a set of sufficient conditions (also called Dirichlet condi-
tions) under which a continuous-time signal x(t) is guaranteed to have a Fourier
transform:
R∞
1. x(t) is absolutely integrable: −∞
|x(t)|dt < ∞.
2. x(t) has a finite number of maxima and minima in any finite interval.
3. x(t) has a finite number of discontinuities in any finite interval, and each
of these discontinuities is finite.
If all of the above conditions are satisfied, x(t) is guaranteed to have a Fourier
transform. Note that this only a sufficient set of conditions, and not necessary.
5.2 Fourier Transform of Periodic Signals 63
An alternate sufficient condition is that the signal have finite energy (i.e., that
it be square integrable):
Z ∞
|x(t)|2 dt < ∞.
−∞
For example, the signal x(t) = 1t u(t − 1) is square integrable, but not absolutely
integrable. Thus the finite energy condition guarantees that x(t) will have a
Fourier transform, whereas the Dirichlet conditions do not apply.
5.2 Fourier Transform of Periodic Signals

The Fourier transform can also be applied to certain periodic signals (although
such signals will not be absolutely integrable or square integrable over the entire
time-axis). A direct application of the Fourier transform equation to such signals
will not necessarily yield a meaningful answer, due to the fact that periodic
signals do not die out. Instead, we will work backwards by starting with a
frequency domain signal and doing an inverse Fourier transform to see what
pairs arise.
Thus, consider the signal x(t) whose Fourier transform is
X(jω) = 2πδ(ω − ω0 ),
i.e., the frequency domain signal is a single impulse at ω = ω0 , with area 2π.
Using the inverse Fourier transform, we obtain
Z ∞ Z ∞
1
x(t) = X(jω)ejωt dω = δ(ω − ω0 )ejωt dω
2π −∞ −∞
= ejω0 t .
Thus, the Fourier transform of x(t) = ejω0 t is X(jω) = 2πδ(ω − ω0 ). Similarly,

if
X∞
X(jω) = ak 2πδ(ω − kω0 ),
k=−∞
then an application of the inverse Fourier transform gives

∞
X
x(t) = ak ejkω0 t .
k=−∞
In other words, if x(t) is a periodic signal with Fourier series coefficients ak ,

then the Fourier transform of x(t) consists of a sequence of impulse functions,
each spaced at multiples of ω0 ; the area of the impulse at kω0 will be 2πak .
Example 5.5. Consider the signal x(t) = cos(ω0 t). The Fourier series coeffi-
cients are a1 = a−1 = 21 . Thus, the Fourier transform of this signal is given
by
X(jω) = a1 2πδ(ω − ω0 ) + a−1 2πδ(ω + ω0 ) = πδ(ω − ω0 ) + πδ(ω + ω0 ).
Example 5.6. Consider the periodic signal

∞
X
x(t) = δ(t − nT ).
n=−∞
The Fourier series coefficients for this signal are given by

Z T ∞
1 2 X 1
a0 = δ(t − nT )dt =
T − T2 n=−∞
T
Z T ∞
1 2 X 1
ak = δ(t − nT )e−jkω0 t dt = .
T − T2 n=−∞
T
Thus, the Fourier transform of this signal is given by

∞ ∞
X 2π X 2kπ
X(jω) = ak 2πδ(ω − kω0 ) = δ ω− .
T T
k=−∞ k=−∞
Thus, if x(t) is an impulse train with period T , its Fourier transform is also an
impulse train in the frequency domain, except with period 2π T . Once again, we
see that if T increases (i.e., the period increases in the time-domain) we obtain
a time-shrinking in the frequency domain.
5.3 Properties of the Continuous-Time Fourier

Transform
We will now discuss various properties of the Fourier transform. As with the
Fourier series, we will find it useful to introduce the following notation. Suppose
x(t) is a time-domain signal, and X(jω) is its Fourier transform. We then say
X(jω) = F {x(t)}
x(t) = F −1 {X(jω)}.
We will also use the notation

F
x(t) ←→ X(jω)
to indicate that x(t) and X(jω) are Fourier transform pairs.

5.3 Properties of the Continuous-Time Fourier Transform 65
5.3.1 Linearity
The first property of Fourier transforms is easy to show:
F {αx1 (t) + βx2 (t)} = αF {x1 (t)} + β F {x2 (t)},

which follows immediately from the definition of the Fourier transform.
5.3.2 Time-Shifting
Suppose x(t) is a signal with Fourier transform X(jω). Define g(t) = x(t − τ )
where τ ∈ R. Then we have
Z ∞ Z ∞
−jωt
G(jω) = g(t)e dt = x(t − τ )e−jωt dt = e−jωτ X(jω).
−∞ −∞
Thus
F {x(t − τ )} = e−jωτ X(jω).
Note the implication: if we time-shift a signal, the magnitude of its Fourier
transform is not affected. Only the phase of the Fourier transform gets shifted
by −ωτ at each frequency ω.
5.3.3 Conjugation
Consider a signal x(t). We have
Z ∞ ∗ Z ∞
X ∗ (jω) = x(t)e−jωt dt = x∗ (t)ejωt dt.
−∞ −∞
Thus, Z ∞
X ∗ (−jω) = x∗ (t)e−jωt dt = F {x∗ (t)}.
−∞
The above is true for any signal x(t) that has a Fourier transform. Now suppose
additionally that x(t) is a real-valued signal. Then we have x∗ (t) = x(t) for all
t ∈ R. Thus
X ∗ (−jω) = F {x∗ (t)} = F {x(t)} = X(jω).
Based on the above relationship between X ∗ (−jω) and X(jω) for real-valued
signals, we see the following. Write X(jω) in polar form as
X(jω) = |X(jω)|ej∠X(jω) .
Then we have
X(−jω) = X ∗ (jω) = |X(jω)|e−j∠X(jω) .
Thus, for any ω, X(−jω) has the same magnitude as X(jω), and the phase of
X(−jω) is the negative of the phase of X(jω). Thus, when plotting X(jω), we
only have to plot the magnitude and phase for positive values of ω, as the plots
for negative values of ω can be easily recovered according to the relationships
described above.
Example 5.7. Consider again the signal x(t) = e−at u(t); we saw earlier that
the Fourier transform of this signal is
1
X(jω) = .
a + jω
It is easy to verify that
1
X(−jω) = = X ∗ (jω),
a − jω
as predicted. Furthermore, we can see from the plots of the magnitude and
phase of X(jω) that the magnitude is indeed an even function, and the phase
is an odd function.
Suppose further that x(t) is even (in addition to being real-valued). Then we
have x(t) = x(−t). Then we have
Z ∞ Z ∞ Z ∞
X(−jω) = jωt
x(t)e dt = jωt
x(−t)e dt = x(t)e−jωt dt
−∞ −∞ −∞
= X(jω).
This, together with the fact that X(−jω) = X ∗ (jω) for real-valued signals
indicates that X(jω) is real-valued and even.
Similarly, if x(t) is real-valued and odd, we have X(jω) is purely imaginary and
odd.
Example 5.8. Consider the signal x(t) = e−a|t| , where a is a positive real
number. This signal is real-valued and even. We have
Z ∞ Z 0 Z ∞
−jωt at −jωt
X(jω) = x(t)e dt = e e dt + e−at e−jωt dt
−∞ −∞ 0
1 1
= +
a − jω a + jω
2a
= 2 .
a + ω2
As predicted, X(jω) is real-valued and even.
5.3.4 Differentiation
Consider the inverse Fourier transform
Z ∞
1
2π −∞
Differentiating both sides with respect to t, we obtain

Z ∞
dx(t) 1
= jωX(jω)ejωt dω.
dt 2π −∞
Thus, we see that

dx(t)
F{ } = jωX(jω).
dt
5.3.5 Time and Frequency Scaling

Let a be a nonzero real number and consider the signal g(t) = x(at) (i.e., a
time-scaling of x(t)). We have
Z ∞
F {g(t)} = x(at)e−jωt dt.
−∞
If we perform the substitution τ = at, we have

( R∞ ω
1
x(τ )e−j a τ dτ, a>0
F {g(t)} = a 1−∞ R∞ −j ω τ
.
− a −∞ x(τ )e a dτ, a < 0
This can be written in a uniform way as

1 ω
F {x(at)} = X j .
|a| a
Thus we see again that shrinking a signal in the time-domain corresponds to

expanding it in the frequency domain, and vice versa.
5.3.6 Duality
We have already seen a few examples of the duality property: suppose x(t) has
Fourier transform X(jω). Then if we have a time-domain signal that has the
same form as X(jω), the Fourier transform of that signal will have the same
form as x(t). For example, the square pulse in the time-domain had a Fourier
transform in the form of a sinc function, and a sinc function in the time-domain
had a Fourier transform in the form of a square pulse.
We can consider another example. Suppose x(t) = e−|t| . Then one can verify
that
2
F {e−|t| } = .
1 + ω2
Specifically we have
Z ∞
−|t| 1 2
e = e−jωt dω
2π −∞ 1 + ω 2
If we multiply both sides by 2π and interchange ω and t, we obtain

Z ∞
−|ω| 2
2πe = 2
e−jωt dt
−∞ 1 + t
Thus, we have
2
F{ } = 2πe−|ω| .
1 + t2
Duality also applies to properties of the Fourier transform. For example, recall
that differentiation in the time-domain corresponds to multiplication by jω in
the frequency domain. We will now see that differentiation the frequency domain
corresponds to multiplication by a certain quantity of t in the time-domain. We
have Z ∞
dX(jω)
= x(t)(−jt)e−jωt dt.
dω −∞
Thus, differentiation in the frequency domain corresponds to multiplication by
−jt in the time-domain.

Just as with periodic signals, we have the following.
Z ∞ Z ∞
1
|x(t)|2 dt = |X(jω)|2 dω
−∞ 2π −∞
To derive this, note that

Z ∞ Z ∞ Z ∞ Z ∞
1
|x(t)|2 dt = x(t)x∗ (t)dt = x(t) X ∗ (jω)e−jωτ dωdt
−∞ −∞ 2π −∞ −∞
Z∞ Z ∞
1 ∗
= X (jω) x(t)e−jωt dtdω
2π −∞ −∞
Z∞
1 ∗
= X (jω)X(jω)dω
2π −∞
Z ∞
1
= |X(jω)|2 dω
2π −∞
5.3.8 Convolution
Consider a signal x(t) with Fourier transform X(jω). From the inverse Fourier
transform, we have Z ∞
1
2π −∞
This has the interpretation that x(t) can be written as a superposition of com-
plex exponentials (with frequencies spanning the entire real axis). From earlier
in the course, we know that if the input to an LTI system is ejωt , then the
output will be H(jω)ejωt , where
Z ∞
H(jω) = h(t)e−jωt dt.
−∞
In other words, H(jω) is the Fourier transform of the impulse response h(t).
This, together with the LTI property of the system, implies that
Z ∞ Z ∞
1 jωt 1
x(t) = X(jω)e dω ⇒ X(jω)H(jω)ejωt dω = y(t).
2π −∞ 2π −∞
Thus, we see that the Fourier transform of the output y(t) is given by
Y (jω) = H(jω)X(jω).
In other words:
The Fourier transform of the output of an LTI system is

given by the product of the Fourier transforms of the input
and the impulse response.
This is potentially the most important fact pertaining to LTI systems and fre-
quency domain analysis. Let’s derive this another way just to reinforce the
fact.
Suppose that we have two signals x(t) and h(t), and define
Z ∞
y(t) = x(t) ∗ h(t) = x(τ )h(t − τ )dτ.
−∞
We have
Z ∞ Z ∞ Z ∞
−jωt
Y (jω) = y(t)e dt = x(τ )h(t − τ )dτ e−jωt dt
−∞ −∞ −∞
Z ∞ Z ∞
−jωt
= x(τ ) h(t − τ )e dt dτ
−∞ −∞
Z∞
= x(τ )e−jωτ H(jω)dτ
−∞
Z ∞
= H(jω) x(τ )e−jωτ dτ
−∞
= H(jω)X(jω).
In the third line, we used the time-shifting property of the Fourier transform.
Thus we see that convolution of two signals in the time-domain corresponds to
multiplication of the signals in the frequency domain, i.e.,
F {x(t) ∗ h(t)} = F {x(t)}F {h(t)}.
One thing to note here pertains to the existence of the Fourier transform of h(t).
Specifically, recall that the LTI system is stable if and only if
Z ∞
|h(t)|dt < ∞.
−∞
This is precisely the first condition in the Dirichlet conditions; thus, as long as
the system is stable and the impulse response satisfies the other two conditions
(which almost all real systems would), the Fourier transform is guaranteed to
exist. If the system is unstable, we will need the machinery of Laplace transforms
to analyze the input-output behavior, which we will defer to a later discussion.
The convolution - multiplication property is also very useful for analysis of
interconnected linear systems. For example, consider the series interconnection
shown in Fig. 5.1.
x(t) y1 (t) y(t)

h1 (t) h2 (t)
Figure 5.1: A series interconnection of systems.
We have
y(t) = y1 (t) ∗ h2 (t) = (x(t) ∗ h1 (t)) ∗ h2 (t) = x(t) ∗ (h1 (t) ∗ h2 (t)).
Taking Fourier transforms, we obtain
Y (jω) = X(jω)H1 (jω)H2 (jω).
This reinforces what we saw earlier, that the series interconnection of LTI sys-
tems can be lumped together in a single LTI system whose impulse response
is the convolution of the impulse responses of the individual systems. In the
frequency domain, their Fourier transforms get multiplied together.
One of the important implications of the convolution property is that it allows
us to investigate the effect of systems on signals in the frequency domain. For
example, this facilitates the design of appropriate filters for signals, as illustrated
in the following example.
Example 5.9. Consider a signal v(t) which represents a measurement of some
relatively low frequency content (such as a voice signal). Suppose that we mea-
sure this signal via a microphone, whose output is given by
x(t) = v(t) + n(t)

where n(t) is high-frequency noise. Note that X(jω) = V (jω) + N (jω). We

would like to take the measured signal x(t) and remove the noise; unfortunately
we do not have access to n(t) to subtract it out. Instead, we can work in the
frequency domain. Suppose we design a filter (an LTI system) whose impulse
response as the following Fourier transform:
(
1 |ω| ≤ W
H(jω) = ,
0 |ω| > W
where W is the highest frequency of the underlying signal v(t). If we feed x(t)
into this filter, the output will have Fourier transform given by
Y (jω) = X(jω)H(jω) = V (jω)H(jω) + N (jω)H(jω).
If all of the frequency content of the noise n(t) occurs at frequencies larger than
W , then we see that N (jω)H(jω) = 0, and thus
Y (jω) = V (jω)H(jω) = V (jω).
In other words, we have recovered the voice signal v(t) by passing x(t) through
the low-pass filter.
Recall that the inverse Fourier transform of the given H(jω) is
sin(W t)
h(t) = .
πt
However, there are various challenges with implementing an LTI system with
this impulse response. One is that this is noncausal, and thus one must poten-
tially include a sufficiently large delay (followed by a truncation of the signal) in
order to apply it. Another problem is that it contains many oscillations, which
may not be desirable for an impulse response.
Instead of the above filter, suppose consider another filter whose impulse re-
sponse is
h2 (t) = e−at u(t).
This filter can be readily implemented with an RC circuit (with the input signal
being applied as an input voltage, and the output signal being the voltage across
the capacitor). The Fourier transform of this impulse response is
1
H2 (jω) = .
jω + a
The magnitude plot of this Fourier transform has content at all frequencies,
and thus this filter will not completely eliminate all of the high frequency noise.
However, by tuning the value of a, one can adjust how much of the noise affects
the filtered signal. Note that this filter will also introduce phase shifts at dif-
ferent frequencies, which will also cause some distortion of the recovered signal.
We just saw that multiplication in the time domain corresponds to convolu-
tion in the frequency domain. By duality, we obtain that multiplication in the
frequency domain corresponds to convolution in the time-domain. Specifically,
consider two signals x1 (t) and x2 (t), and define g(t) = x1 (t)x2 (t). Then we have
Z ∞ Z ∞
G(jω) = g(t)e−jωt dt = x1 (t)x2 (t)e−jωt dt
−∞ −∞
Z ∞ Z ∞
1
= x2 (t) X1 (jθ)ejθt dθe−jωt dt
2π −∞ −∞
Z ∞ Z ∞
1
= X1 (jθ) x2 (t)e−j(ω−θ)t dtdθ
2π −∞ −∞
Z ∞
1
= X1 (jθ)X2 (j(ω − θ))dθ.
2π −∞
Thus,
Z ∞
1 1
F {x1 (t)x2 (t)} = (X1 (jω) ∗ X2 (jω)) = X1 (jθ)X2 (j(ω − θ))dθ.
2π 2π −∞
Multiplication of one signal x1 (t) by another signal x2 (t) can be viewed as

modulating the amplitude of one signal by the other. This plays a key role in
communication systems.
Example 5.10. Consider a signal s(t) whose frequency spectrum lies in some
interval [−W, W ]. Consider the signal p(t) = cos(ω0 t). The Fourier transform
of p(t) is given by
P (jω) = πδ(ω − ω0 ) + πδ(ω + ω0 ).
Now consider the signal x(t) = s(t)p(t), with Fourier transform given by
Z ∞
1
X(jω) = S(jθ)P (j(ω − θ))dθ
2π −∞
Z ∞
1
= S(jθ)δ(ω − θ − ω0 )dθ
2 −∞
Z ∞
1
+ S(jθ)δ(ω − θ + ω0 )dθ
2 −∞
1 1
= S(j(ω − ω0 )) + S(j(ω + ω0 )).
2 2
Thus, multiplying the signal s(t) by p(t) results in a signal x(t) whose frequency
spectrum consists of two copies of the spectrum of s(t), cenetered at the fre-
quencies ω0 and −ω0 and scaled by 21 .
The above example illustrates the principle behind amplitude modulation (AM)
in communication and radio systems. A low frequency signal (such as voice) is
amplitude modulated to a higher frequency that is reserved for that signal. It
is then transmitted at that frequency to the receiver. The following example
illustrates how the receiver can recover the transmitted signal.
Example 5.11. Consider the signal x(t) = s(t)p(t) from the previous example.
Its frequency spectrum has two copies of the spectrum of s(t), located at ±ω0 .
We want to recover the original signal s(t) from x(t). To do this, suppose we
multiply x(t) by cos(ω0 t) again, to obtain
y(t) = x(t) cos(ω0 t).
As above, we have
1 1
Y (jω) = X(j(ω − ω0 )) + X(j(ω + ω0 )).
2 2
By drawing this, we see that the frequency spectrum of y(t) contains three
copies of the spectrum of s(t): one copy centered at ω = 0 (and scaled by 21 ),
one copy at 2ω0 scaled by 14 , and one copy at −2ω0 , scaled by 41 . Thus, if we
want to recover s(t) from y(t), we simply apply a low pass filter to it (and scale
it by 2).
Chapter 6
The Discrete-Time Fourier

Transform
Reading: Signals and Systems, Chapter 5.1-5.5.

We now turn our attention to discrete-time aperiodic signals. We saw that
the Fourier transform for continuous-time aperiodic signals can be obtained by
taking the Fourier series of an appropriately defined periodic signal (and letting
the period go to ∞); we will follow an identical argument for discrete-time
aperiodic signals. The differences between the continuous-time and discrete-
time Fourier series (e.g., that the latter only involves a finite number of complex
exponentials) will be reflected as differences between the continuous-time and
discrete-time Fourier transforms as well.
6.1 The Discrete-Time Fourier Transform

Consider a general signal x[n] which is nonzero on some interval −N1 ≤ n ≤ N2
and zero elsewhere. We create a periodic extension x̃[n] of this signal with
period N (where N is large enough so that there is no overlap). As N → ∞,
x̃[n] becomes equal to x[n] for each finite value of n.
Since x̃[n] is periodic, it has a discrete-time Fourier series representation given
by
N
X −1
x̃[n] = ak ejkω0 n ,
k=0
2π
where ω0 = N . The Fourier series coefficients are given by
1 X−1
n0 +N
ak = x̃[n]e−jω0 kn ,
N n=n0
76 The Discrete-Time Fourier Transform
where n0 is any integer. Suppose we choose n0 so that the interval [−N1 , N2 ] is

contained in [n0 , n0 + N − 1]. Then since x̃[n] = x[n] in this interval, we have
1 X−1
n0 +N
1 X
∞
ak = x[n]e−jω0 kn = x[n]e−jω0 kn .
N n=n0
N n=−∞
Let us now define the discrete-time Fourier transform as
∞
X
X(ejω ) , x[n]e−jωn .
n=−∞
From this, we see that ak = N1 X(ejkω0 ), i.e., the discrete-time Fourier series
coefficients are obtained by sampling the discrete-time Fourier transform at
periodic intervals of ω0 . Also note that X(ejω ) is periodic in ω with period 2π
(since e−jωn is 2π-periodic).
Using the Fourier series representation of x̃[n], we now have
N −1 N −1 N −1
X 1 X 1 X
x̃[n] = ak ejkω0 n = X(ejkω0 )ejkω0 n = X(ejkω0 )ejkω0 n ω0 .
N 2π
k=0 k=0 k=0
Once again, we see that each term in the summand represents the area of a
rectangle of width ω0 obtained from the curve X(ejω )ejω . As N → ∞, we have
ω0 → 0. In this case, the sum of the areas of the rectangles approaches the
integral of the curve X(ejω )ejωn , and since the sum was over only N samples
of the function, the integral is only over one interval of length 2π. Since x̃[n]
approaches x[n] as N → ∞, we have
Z
1
x[n] = X(ejω )ejωn dω.
2π 2π
This is the inverse discrete-time Fourier transform, or the synthesis

equation.
The main differences between the discrete-time and continuous-time Fourier
transforms are the following. (1) The discrete-time Fourier transform X(ejω ) is
periodic in ω with period 2π, whereas the continuous-time Fourier transform is
not necessarily periodic. (2) The synthesis equation for the discrete-time Fourier
transform only involves an integral over an interval of length 2π, whereas the one
for the continuous-time Fourier transform is over the entire ω axis. Both of these
are due to the fact that ejωn is 2π-periodic in ω, whereas the continuous-time
complex exponential is not.
6.1 The Discrete-Time Fourier Transform 77
Since the frequency spectrum of X(ejω ) is only uniquely specified over an in-
terval of length 2π, we have to be careful about what we mean by “high” and
“low” frequencies. Recalling the discussion of discrete-time complex exponen-
tials, high-frequency signals in discrete-time have frequencies close to odd multi-
ples of π, whereas low-frequency signals have frequencies close to even multiples
of π.
x[n] = an u[n], |a| < 1.
We have
∞
X ∞
X
X(ejω ) = x[n]e−jωn = an e−jωn
n=−∞ n=0
X∞
= (ae−jω )n
n=0
1
= .
1 − ae−jω
If we plot the magnitude of X(ejω ), we see an illustration of the “high” versus
“low” frequency effect. Specifically, if a > 0 then the signal x[n] does not have
any oscillations and |X(ejω )| has its highest magnitude around even multiples
of π. However, if a < 0, then the signal x[n] oscillates between positive and
negative values at each time-step; this “high-frequency” behavior is captured
by the fact that |X(ejω )| has its largest magnitude near odd multiples of π. See
Figure. 5.4 in OW for an illustration.
x[n] = a|n| , |a| < 1.
We have
∞
X ∞
X
X(ejω ) = x[n]e−jωn = a|n| e−jωn
n=−∞ n=−∞
−1
X ∞
X
= a−n e−jωn + an e−jωn
n=−∞ n=0
X∞ ∞
X
= n jωn
a e + an e−jωn
n=1 n=0
jω
ae 1
= +
1 − aejω 1 − ae−jω
1 − a2
= .
1 − 2a cos(ω) + a2
6.2 The Fourier Transform of Discrete-Time Pe-

riodic Signals
In the last chapter, we saw that if we take the Fourier transform of a continuous-
time periodic signal, we obtain scaled impulses located at the harmonic frequen-
cies. We will see something similar here for discrete-time periodic signals.
First, consider the signal
x[n] = ejω0 n .
We claim that the Fourier transform of this signal is
∞
X
X(ejω ) = 2πδ(ω − ω0 − 2πl),
l=−∞
i.e., a set of impulse functions spaced 2π apart on the frequency axis. To verify
this, note that the inverse Fourier transform is given by
Z
1
X(ejω )ejωn dω.
2π 2π
The integral is only over an interval of length 2π, and there is at most one
impulse function from X(ejω ) in any such interval. Let that impulse be located
at ω0 + 2πr for some r ∈ Z. Then we have
Z Z
1 1
X(ejω )ejωn dω = 2πδ(ω −ω0 −2πr)ejωn dω = ej(ω0 +2πr)n = ejω0 n .
2π 2π 2π 2π
Thus consider a periodic discrete-time signal x[n], with Fourier series

N
X −1
x[n] = ak ejkω0 n = a0 + a1 ejω0 n + a2 ej2ω0 n + · · · + aN −1 ej(N −1)ω0 n ,
k=0
where ω0 = 2π N . The Fourier transform of each term of the form ak e

jkω0 n
is
a set of impulses spaced 2π apart, with one located at ω = kω0 . Furthermore
each of these impulses is scaled by ak 2π. Since ak = ak+N l for any l (by the
periodicity of the discrete-time Fourier series coefficients), when we add up the
Fourier transforms of all of the terms in the Fourier series expansion of x[n], we
obtain
∞
X 2πk
X(ejω ) = 2πak δ ω − .
N
k=−∞
Thus, the Fourier transform of a discrete-time periodic signal is indeed a se-

quence of impulses located at multiples of ω0 , with a period of 2π.
6.3 Properties of the Discrete-Time Fourier Transform 79
Example 6.3. Consider the impulse train

∞
X
x[n] = δ[n − kN ].
k=−∞
The Fourier series coefficients of x[n] are given by

N −1
1 X 1
ak = x[n]e−jkω0 n = .
N N
k=0
Thus the Fourier transform of x[n] is

∞ ∞
X 2πk 2π X 2πk
X(ejω ) = ak 2πδ ω − = δ ω− .
N N N
k=−∞ k=−∞
6.3 Properties of the Discrete-Time Fourier Trans-

form
6.3.1 Periodicity
The Fourier transform of a signal x[n] is periodic in frequency, with period 2π:
X(ejω ) = X(ej(ω+2π) ).
This comes out of the fact that discrete-time complex exponentials are periodic
in frequency with period 2π.
6.3.2 Linearity
It is easy to see that
F {αx1 [n] + βx2 [n]} = αX1 (ejω ) + βX2 (ejω ).
6.3.3 Time and Frequency Shifting

We have
F {x[n − n0 ]} = e−jωn0 X(ejω ),
and
F {ejω0 n x[n]} = X(ej(ω−ω0 ) ).
The first property is easily proved using the inverse Fourier transform equation,
and the second property is proved using the Fourier transform equation.
Example 6.4. Consider a discrete-time low-pass filter, whose Fourier transform

Hlp (ejω ) is a square pulse centered at even multiples of π. Now consider the high
pass filter Hhp (ejω ) which consists of square pulses centered at odd multiples of
π. We see that Hhp (ejω ) = Hlp (ej(ω−π) ). Thus we have
hhp [n] = ejπn hlp [n] = (−1)n hlp [n].
6.3.4 First Order Differences

Consider the discrete-time analog of differentiation, which is to take the differ-
ences between subsequent samples of the signal. By applying the linearity and
time-shifting properties, we have
F {x[n] − x[n − 1]} = X(ejω ) − e−jω X(ejω ) = (1 − e−jω )X(ejω ).
6.3.5 Conjugation
For any discrete-time signal (that has a Fourier transform), we have
F {x∗ [n]} = X ∗ (e−jω ).
Furthermore, if x[n] is real, we have x∗ [n] = x[n] and thus
X(ejω ) = X ∗ (e−jω ).
6.3.6 Time-Reversal
Consider the time-reversed signal x[−n]. We have
Z 2π Z 2π
1 −jωn 1
x[−n] = jω
X(e )e dω = X(e−jω )ejωn dω
2π 0 2π 0
which is obtained by performing a change of variable ω → −ω (note that the

negative sign introduced by this change is canceled out by the reversal of the
bounds of integration that arise because of the negation). Thus, we have
F {x[−n]} = X(e−jω ).
Together with the conjugation property, we see that for real-valued even signals
(where x[n] = x[−n]), we have
X(ejω ) = F {x[n]} = F {x[−n]} = X(e−jω ).
Thus, the Fourier transform of real, even signals is also real and even.
6.3.7 Time Expansion

Recall that for a continuous-time signal x(t) and a scalar a 6= 0, we had

1 jω
F {x(at)} = X .
|a| a
Thus, an expansion in the time-domain led to a compression in the frequency

domain and vice versa.
In discrete-time, expansion and contraction of time-domain signals is not achieved
simply by scaling the time-variable. First, since the time index must be an inte-
ger, it does not make sense to consider the signal x[an], where a < 1. Similarly,
if we consider integer values of a larger than 1, then the signal x[an] only con-
siders the values of x[n] at integer multiples of a, and all information between
those values is lost.
A different expansion of signals that preserves all their values is as follows. For
a given signal x[n] and positive integer k, define the signal
(
x[n/k] if n is a multiple of k
x(k) [n] = .
0 otherwise
Thus, the signal x(k) [n] is obtained by spreading the points of x[n] apart by k
samples and placing zeros between the samples. We have
∞
X ∞
X
F {x(k) [n]} = x(k) [n]e−jωn = x(k) [rk]e−jωrk
n=−∞ r=−∞
since x(k) [n] is nonzero only at integer multiples of k. Since x(k) [rk] = x[r], we
have
X∞
F {x(k) [n]} = x[r]e−jωrk = X(ejkω )
r=−∞


1
 n ∈ {0, 2, 4}
x[n] = 2 n ∈ {1, 3, 5} .

0 otherwise

We note that we can write x[n] as
x[n] = g[n] + 2g[n − 1],
where (
1 n ∈ {0, 2, 4}
g[n] = .
0 otherwise
This g[n] can be viewed as an expansion of the signal

(
1 0≤n≤2
h[n] = .
0 otherwise
Specifically, g[n] = h(2) [n]. The Fourier transform of h[n] is given by
2
X 1 − e−3jω sin( 23 ω)
H(ejω ) = e−jωn = = e−jω .
n=0
1−e −jω sin( ω2 )
Thus, the Fourier transform of g[n] is given by
sin(3ω)
G(ejω ) = H(ej2ω ) = e−2jω
sin(ω)
Finally,
sin(3ω)
X(ejω ) = G(ejω ) + 2e−jω G(ejω ) = e−2jω (1 + 2e−jω ) .
sin(ω)
6.3.8 Differentiation in Frequency

Suppose x[n] has Fourier transform X(ejω ). Then we have
∞ ∞
d d X X
X(ejω ) = x[n]e−jωn = (−jn)x[n]e−jωn .
dω dω n=−∞ n=−∞
Thus
dX(ejω )
F {nx[n]} = j .
dω

We have
∞ Z
X
2 1
|x[n]| = |X(ejω )|2 dω.
n=−∞
2π 2π
6.3.10 Convolution
Just as in continuous time, the discrete time signal ejωn is an eigenfunction
of discrete-time LTI systems. Specifically, if ejωn is applied to a (stable) LTI
system with impulse response h[n], the output of the system will be H(ejω )ejωn .
Thus consider a signal x[n] written in terms of its Fourier transform as

Z
1
x[n] = X(ejω )ejωn dω.
2π 2π
This is a linear combination of complex exponentials (where the scaling factor

1
on the complex exponential ejωn is 2π X(ejω ). By the LTI property, we thus
have
Z Z
1 1
x[n] = X(ejω )ejωn dω → X(ejω )H(ejω )ejωn dω = y[n].
2π 2π 2π 2π
The expression on the right hand side is the output y[n] of the system when the
input is x[n]. Thus we have
Y (ejω ) = H(ejω )X(ejω ).
As in the continuous-time case, convolution in the time-domain is given by

multiplication in the frequency domain.
Example 6.6. Consider the system shown in Fig. 5.18a in OW. Let us analyze
the relationship between y[n] and x[n] for that system.
First, we have w1 [n] = (−1)n x[n] = ejπn x[n]. By the frequency shifting prop-
erty, we see that W1 (ejω ) = X(ej(ω−π) ). Next, we have
W2 (ejω ) = Hlp (ejω )W1 (ejω ) = Hlp (ejω )X(ej(ω−π) ).
The signal w3 [n] is given by w3 [n] = (−1)n w2 [n], and thus W3 (ejω ) = W2 (ej(ω−π) ).
Putting this together with the expression for W2 (ejω ), we obtain
W3 (ejω ) = Hlp (ej(ω−π) )X(ej(ω−2π) ) = Hlp (ej(ω−π) )X(ejω ).
From the bottom path, we have W4 (ejω ) = Hlp (ejω )X(ejω ). Thus, we have

Y (ejω ) = W3 (ejω ) + W4 (ejω ) = Hlp (ejω ) + Hlp (ej(ω−π) ) X(ejω ).
Recall that Hlp (ej(ω−π) ) is a high-pass filter centered at π. Thus, this system
acts as a bandstop filter, blocking all frequencies in a certain range and letting
all of the low and high frequency signals through.
Consider two signals x1 [n] and x2 [n], and define g[n] = x1 [n]x2 [n]. The discrete-
time Fourier transform of g[n] is given by
∞
X
G(ejω ) = x1 [n]x2 [n]e−jωn .
n=−∞
Replacing x1 [n] by the synthesis equation

Z
1
x1 [n] = X1 (ejθ )ejθn dθ,
2π 2π
where we simply replaced the dummy variable ω with θ to avoid confusion, we
obtain
∞ Z
X 1
jω
G(e ) = X1 (ejθ )ejθn dθx2 [n]e−jωn
n=−∞
2π 2π
Z ∞
1 X
= X1 (ejθ ) x2 [n]e−j(ω−θ)n dθ
2π 2π n=−∞
Z
1
= X1 (ejθ )X2 (ej(ω−θ) )dθ.
2π 2π
This resembles the typical convolution of the signals X1 (ejω ) and X2 (ejω ), ex-
cept that the integral is over only an interval of length 2π as opposed over the
entire frequency axis. This is called the periodic convolution of the two signals.
Recall that we saw the same thing when we considered the discrete-time Fourier
series of the product of two periodic discrete-time signals.
Example 6.7. Consider the two signals
sin( π2 n) sin( π4 n)
x1 [n] = , x2 [n] = .
πn πn
The Fourier transforms of these signals are square pulses, where the pulse cen-
tered at 0 extend from − π2 to π2 (for X1 (ejω )) and from − π4 to π4 (for X2 (ejω )).
The Fourier transform of g[n] = x1 [n]x2 [n] is given by
Z
1
G(ejω ) = X1 (ejθ )X2 (ej(ω−θ) )dθ.
2π 2π
Since we can choose any interval of length 2π to integrate over, let’s choose the
interval [−π, π) for convenience. We also only need to determine the values of
the Fourier transform for values of ω between −π and π, since the transform is
periodic. Depending on the value of ω, there are different cases that occur:
• If −π ≤ ω < − 3π jθ
4 , then there is no overlap in the signals X1 (e ) and
j(ω−θ)
X2 (e ), and thus G(jω) is zero.
• If − 3π π
4 ≤ ω < − 4 , then there is partial overlap in the signals; the product
is a rectangle with support from − π2 to ω + π4 , and thus G(jω) evaluates
1
to 2π (ω + 3π
4 ).
• If − π4 ≤ ω < π
4, there is full overlap and G(jω) is 1 π
2π ( 2 ) = 14 .
π 3π 1 3π
• If 4 ≤ω< 4 , there is partial overlap and G(jω) is 2π ( 4 − ω).
3π
• If 4 ≤ ω < π, there is no overlap and G(jω) is zero.
Note that since we are only integrating over θ between −π and π, the values of
X(ejθ ) outside of that interval does not matter. Thus, we could also create a
new signal X̂1 (ejθ ) which is equal to X1 (ejθ ) over the interval [−π, π) and zero
everywhere else. The Fourier transform can then be written as
Z Z ∞
jω 1 jθ j(ω−θ) 1
G(e ) = X1 (e )X2 (e )dθ = X̂1 (ejθ )X2 (ej(ω−θ) )dθ,
2π 2π 2π −∞
i.e., it is the usual convolution of the signals X̂1 (ejω ) and X2 (ejω ).
Chapter 7
Sampling
Thus far, we have considered continuous-time signals and discrete-time signals

(and their associated Fourier transforms) as two parallel tracks. In this part of
the course, we will bring these two threads together and relate their frequency
spectra. The main tool that we will leverage is sampling continuous-time signals
to yield discrete-time signals. In particular, it is often desirable to process signals
using digital systems (e.g., computers or embedded devices). Thus, we take a
continuous-time signal, sample it at a sufficiently fast rate, process it using a
digital filter, and then convert the processed signal back to a continuous-time
signal.
7.1 The Sampling Theorem

Consider a continuous-time signal x(t). A sampled version of this signal is
obtained by considering the values of the signal only at certain discrete points
in time. In particular, periodic or uniform sampling occurs when we pick some
positive real number Ts , and consider the values x(nTs ), n ∈ Z. We will often
denote this discrete sequence as x[n] (dropping the sampling period Ts ), which
was the notation that we used in our analysis of discrete-time signals. The
sampling frequency is denoted by ωs = 2π Ts .
One can always sample a signal this way. However, the main question is how
fast one needs to sample (i.e., how small Ts needs to be) in order for the samples
to be a faithful representation of the underlying continuous-time signal. We will
study this question here.
First, it will be useful for us to have a mathematical representation of the
sampled signal. Define the impulse train
∞
X
p(t) = δ(t − nTs ).
n=−∞
88 Sampling
Then, sampling a signal x(t) at a sampling period of Ts can be represented as

multiplying that signal by an impulse train, i.e.,
∞
X ∞
X
xp (t) = x(t)p(t) = x(t)δ(t − nTs ) = x(nTs )δ(t − nTs ).
n=−∞ n=−∞
Note that the values of the signal x(t) are irrelevant outside of the points where
the impulse functions in p(t) occur (i.e., at the sampling instants). Let us con-
sider the frequency spectra of these signals. Specifically, by the multiplication
property of Fourier transforms, we have
Z ∞
1
Xp (jω) = X(jθ)P (j(ω − θ))dθ.
2π −∞
Furthermore, since p(t) is periodic, we saw that the Fourier transform of p(t)
will be given by
∞
2π X
P (jω) = δ(ω − nωs ),
Ts n=−∞
as the Fourier series coefficients of p(t) are each T1s . Thus,
∞ Z ∞
1 X
Xp (jω) = X(jθ)δ(ω − θ − nωs )dθ
Ts n=−∞ −∞
∞
1 X
= X(j(ω − nωs )).
Ts n=−∞
Thus, the frequency spectrum of xp (t) consists of copies of the frequency spec-
trum of x(t), where each copy is shifted (in frequency) by an integer multiple of
the sampling frequency ωs and scaled by T1s (see Fig. 7.1).
If we want to be able to reconstruct x(t) from its sampled version xp (t), we would
like to make sure that there is an exact copy of X(jω) that can be extracted
from Xp (jω). Based on the above discussion, we see that this will be the case if
no two copies of X(jω) overlap in Xp (jω). Looking at Fig. 7.1, this will occur
as long as
ωs − ωM > ωM ,
or equivalently,
ωs > 2ωM ,
where ωM is the largest frequency at which x(t) has nonzero content. This leads
to the sampling theorem.
If the sampling frequency ωs is larger than twice the largest

frequency of the signal x(t), then we can reconstruct the
signal x(t) from its sampled version xp (t) by passing xp (t)
through an ideal low-pass filter, with cutoff ωc = ω2s .
The frequency 2ωM is called the Nyquist rate.

7.2 Reconstruction of a Signal From Its Samples 89
X(jω)
ω
ωM ωM
Xp (jω)
1
Ts X(0)
−ωs ωs ω
−ωs − ωM −ωs + ωM −ωM ωM ωs − ωM ωs + ωM
Figure 7.1: The frequency spectrum of the signal x(t) and the signal xp (t).
7.2 Reconstruction of a Signal From Its Samples

In general, it is not possible to implement an ideal low-pass filter: obtaining
sharp cutoffs is difficult, and furthermore, an ideal low-pass filter is noncausal
(as it corresponds to a sinc function in the time-domain). There are various
other options that are frequently used to reconstruct sampled signals.
7.2.1 Zero-Order Hold

The simplest option to reconstruct a signal is to simply hold the value of the
signal constant at the value of the previous sample. This is called a zero-order-
hold (ZOH). To compare this strategy to the ideal low-pass-filter, let’s consider
the transfer function of the ZOH. Specifically, note that if we put an impulse
function into the ZOH, the output h0 (t) (i.e., impulse response) will be a square
pulse from t = 0 to t = Ts (because the ZOH will keep the value of the sample
at t = 0 constant at 1 until the next sample at t = Ts , after which point all
samples are zero). This is shown in Fig. 7.2. It is easy to check that the transfer
function is given by
Z ∞ Ts

−jωt −jω T2s sin ω 2
H0 (jω) = h0 (t)e dt = e ω .
−∞ 2
This has magnitude Ts at ω = 0 (like the ideal reconstructor), and the first
frequency at which it is equal to zero is at ωs (unlike the ideal reconstructor
that cuts off at ω2s ). Furthermore, this frequency spectrum is not bandlimited,
and thus the copies of X(jω) in the spectrum of Xp (jω) will leak into the
reconstructed signal under the ZOH.
90 Sampling
h0 (t) h1 (t)
1 1
t t
Ts −Ts Ts
Figure 7.2: The impulse response of a zero-order-hold (left) and a first-order-

hold (right).
7.2.2 First-Order Hold

A slightly more sophisticated reconstruction mechanism is to create a continuous-
time signal by joining each consecutive pair of samples by a line. This is called
a first-order-hold (FOH). The impulse response h1 (t) of an FOH is shown in
the right plot of Fig. 7.2 (again, imagine how the FOH reacts to an impulse
coming into it: it sees a value of 0 at t = −Ts , a value of 1 at t = 0 and a value
of 0 at t = Ts ). The transfer function is given by
Z ∞
H1 (jω) = h1 (t)e−jωt dt
−∞
Z0 Z Ts
1 1
= t + 1 e−jωt dt + − t + 1 e−jωt dt.
−Ts Ts 0 Ts
After some algebra (integration by parts, etc.), we obtain

!2
1 sin ω T2s 1
H1 (jω) = ω = |H0 (jω)|2 .
Ts 2 Ts
The magnitude of this filter is smaller than that of H0 (jω) outside of ω2s , al-
though it is still not bandlimited. Furthermore, the FOH is noncausal, but can
be made causal with a delay of Ts .
Higher order filters are also possible, and can be defined as a natural extension
of zero and first order holds.
7.3 Undersampling and Aliasing

If the sampling frequency ωs is not strictly larger than twice the largest fre-
quency, we will not be able to perfect reconstruct the original signal. To illus-
trate this, it is easiest to consider sampled sinusoids.
Consider the signal x(t) = cos(t) which has frequency ω0 = 1. The sampling
theorem indicates that as long as the sampling frequency ωs is larger than
7.4 Discrete-Time Processing of Continuous-Time Signals 91
Xp (jω)
1
2Ts
ω
−9 −8 −7 −6 −5 −4 −3 −2 −1 0 1 2 3 4 5 6 7 8 9
Figure 7.3: Sampling x(t) = cos(t) at a frequency of ωs = 4.
2ω0 = 2, we can reconstruct x(t) from its samples. Let us choose ωs = 4. The
frequency spectrum of xp (t) = x(t)p(t) is shown in Fig. 7.3.
Now consider another signal x1 (t) = cos(ω1 t), and suppose that we sample this
signal at ωs = 4. Let xp1 (t) be the resulting (continuous-time) sampled signal.
For what value of ω1 will the frequency spectrum Xp1 (jω) look exactly the same
as Xp (jω)?
To answer this, note that X1 (jω) has impulses located at ±ω1 , and Xp1 (jω)
will have impulses located at kωs ± ω1 , for k ∈ Z. Looking at the frequency
spectrum of Xp (jω) in Fig. 7.3, we see that ω1 should be odd (otherwise, Xp1 (jω)
will have impulses at some even frequencies, whereas all of the impulses are at
odd frequencies in Xp (jω)). Suppose we try ω1 = 3. Then Xp1 (jω) will have
impulses at ±3, which matches two of the impulses in Xp (jω). We should check
the copies of the signals in Xp1 (jω) as well. Specifically, there will be a copy
centered at ωs = 4, with one impulse three units to the left (at ω = 1) and
one impulse three units to the right (at ω = 7). Similarly, the copy centered at
2ωs will have one impulse at 5 and one impulse at 11. The same is true for the
negative frequencies. Thus, we see that if ω1 = 3, then Xp1 (jω) looks exactly
the same as Xp (jω), and thus the signals x(t) = cos(t) and x1 (t) = cos(3t) look
exactly the same if sampled at ωs = 4.
7.4 Discrete-Time Processing of Continuous-Time

Signals
Let’s take a closer look at taking a signal from continuous-time, operating on
it, and converting it back to discrete-time. Specifically, given a signal x(t), let
xp (t) = x(t)p(t) be the continuous-time representation of the sampled signal,
and let xd [n] = x(nTs ) be the sequence of samples.
We have
Z ∞ ∞
X ∞
X
Xp (jω) = x(t)δ(t − nTs )e−jωt dt = x(nTs )e−jωnTs .
−∞ n=−∞ n=−∞
92 Sampling
One the other hand, if we take the discrete-time Fourier transform of the se-
quence xd [n], we have
∞
X ∞
X
Xd (ejω ) = xd [n]e−jωn = x(nTs )e−jωn .
n=−∞ n=−∞
Comparing the two expressions, we see that

ω
Xd (ejω ) = Xp j .
Ts
In other words, the frequency spectrum of xp (t) (given by the continuous-time
Fourier transform) is just a frequency-scaled version of the frequency spectrum
of xd [n] (given by the discrete-time Fourier transform). Specifically, Xp (jω) is
ωs = 2π jω
Ts periodic, whereas Xd (e ) is 2π periodic. This scaling is essentially due
to the fact that the discrete-time signal xd [n] is “normalized” with respect to
the sampling period; it only operates on the sequence of samples, and does not
explicitly consider how far apart those samples are. However, xp (t) explicitly
contains the sampling period Ts , as the impulses are spaced that far apart.
The above result has the following implication for the digital processing of sig-
nals. Suppose that we wish to implement a filter that has a continuous-time
Fourier transform H(jω), but using a discrete-time system. Suppose H(jω) is
bandlimited, with highest frequency ωM . Then we simply
design the discrete-
jω ω
time filter to have frequency reponse Hd (e ) = H j Ts for −ωM Ts ≤ ω ≤
ωM Ts (and Hd (ejω ) being 2π-periodic otherwise). The inverse Fourier trans-
form of Hd (ejω ) can then be found to obtain the impulse response of the digitial
filter. After the sampled signal is processed with this digital filter, it can then
be transformed back into continuous-time via a ZOH, FOH, etc.
Example 7.1. Consider a bandlimited differentiator
(
jω |ω| ≤ ωc
H(jω) = .
0 otherwise
The magnitude and phase are shown in Fig. 7.4.
To implement this in discrete-time, we create a discrete-time filter with transfer
function Hd (ejω ) to have the same shape as H(jω) (except for the fact that Hd
is periodic), with frequencies scaled by Ts , i.e.,
( ω
jω ω j Ts |ω| ≤ ωc Ts
Hd (e ) = H j = .
Ts 0 ωc Ts < |ω| ≤ 2π
The impulse response of the corresponding filter is
(−1)n
(
nTs n 6= 0
hd [n] = .
0 n=0
7.4 Discrete-Time Processing of Continuous-Time Signals 93
|H(jω)| ∠H(jω)
π
ωc 2
ω ω
−ωc ωc −ωc ωc
Figure 7.4: The frequency response of a bandlimited differentiator.

94 Sampling
Chapter 8
The Laplace Transform
Reading: Signals and Systems, Chapter 9.1-9.3, 9.5-9.7.
Thus far, we have seen ways to take time-domain signals and transform them
into frequency-domain signals, by identifying the amount of contribution of
complex exponentials of given frequencies to the signal. Specifically, for pe-
riodic signals, we started with the Fourier series representation of a signal in
terms of its harmonic family. For more general absolutely integrable signals,
we generalized the Fourier series to the Fourier transform, where the signal is
represented in terms of complex exponentials of all frequencies (not just those
from the harmonic family).
8.1 The Laplace Transform
To develop this, first recall that complex exponentials of the form est are eigen-
functions of LTI systems, even when s is a general complex number. Specifically,
if x(t) = est is the input to an LTI system with impulse response h(t), we have
Z ∞ Z ∞
y(t) = x(t) ∗ h(t) = x(t − τ )h(t)dτ = est h(t)e−sτ dτ.
−∞ −∞
Based on the above,

R ∞ we see that the output is the input signal est , multiplied
−sτ
by the quantity −∞ h(t)e dτ . We will call this the Laplace transform of the
signal h(t).
96 The Laplace Transform
The Laplace transform of a signal x(t) is given by

Z ∞
X(s) = x(t)e−st dt,
−∞
where s ∈ C. We will also denote the Laplace transform of

x(t) by L{x(t)}.
Note that the limits of the integration go from −∞ to ∞, and thus this is called
the bilateral Laplace transform. When the limits only go from 0 to ∞, it
is called the unilateral Laplace transform. For the purposes of this course,
if we leave out the qualifier, we mean the bilateral transform. Note that when
s = jω, then X(s) is just the Fourier transform of x(t) (assuming the transform
exists). However, the benefit of the Laplace transform is that it also applies to
signals that do not have a Fourier transform. Specifically, note that s can be
written as s = σ + jω, where σ and ω are real numbers. Then we have
Z ∞
X(s) = X(σ + jω) = x(t)e−σt e−jωt dt.
−∞
Thus, for a given s = σ+jω, we can think of the Laplace transform as the Fourier
transform of the signal x(t)e−σt . Even if x(t) is not absolutely integrable, it may
be possible that x(t)e−σt is absolutely integrable if σ is large enough (i.e., the
complex exponential can be chosen to cancel out the growth of the signal in the
Laplace transform).
Example 8.1. Consider the signal x(t) = e−at u(t) where a is some real number.
The Laplace transform is given by
Z ∞ Z ∞
X(s) = x(t)e−st dt = e−(s+a)t dt
−∞ 0
∞
1 −(s+a)t
=− e
s+a
0
1
= ,
s+a
as long as Re{s + a} > 0, or equivalently Re{s} > −a. Note that if a is positive,
then the integral converges for Re{s} = 0 as well, in which case we get the
1
Fourier transform X(jω) = jω+a . However, if a is negative, then the signal
does not have a Fourier transform (but it does have a Laplace transform for s
with a sufficiently large real part).
Example 8.2. Consider the signal x(t) = −e−at u(−t) where a is a real number.
8.1 The Laplace Transform 97
We have
Z ∞ Z 0
−st
X(s) = x(t)e dt = − e−(s+a)t dt
−∞ −∞
1 0
= e−(s+a)t

s+a −∞
1
= ,
s+a
as long as Re{s + a} < 0, or equivalently, Re{s} < −a.
Comparing the above examples, we notice that both the signals e−at u(t) and
−e−at u(−t) had the same Laplace transform s+a1
, but that the ranges of s for
which each had a Laplace transform was different.
Consider a signal x(t). The range of values of s for which

the Laplace transform integral converges is called the
Region of Convergence (ROC) of the Laplace transform.
Thus, in order to specify the Laplace transform of a signal, we have to specify

1
both the algebraic expression (e.g., s+a ) and the region of convergence for which
this expression is valid. A convenient way to visualize the ROC is as a shaded
region in the complex plane. For example, the ROC Re{s} > −a can be repre-
sented by shading all of the complex plane to the right of the line Re{s} = −a.
Similarly, the ROC Re{s} < −a is represented by shading the complex plane to
the left of the line Re{s} = −a.
Example 8.3. Consider the signal x(t) = 3e−2t u(t) − 2e−t u(t). It is easy to
see that the Laplace transform is a linear operation, and thus we can find the
Laplace transform of x(t) as a sum of the Laplace transform of the two signals
on the right hand side.
The Laplace transform of 3e−2t u(t) is s+2
3
, with ROC Re{s} > −2. The Laplace
−t 2
transform of −2e u(t) is − s+1 , with ROC Re{s} > −1. Thus, for the Laplace
transform of x(t) to exist, we need s to fall in the ROC of both of its constituent
parts, which means Re{s} > −1. Thus,
3 2 s−1
X(s) = − = 2 ,
s+2 s+1 s + 3s + 2
with ROC Re{s} > −1.
In the above examples, we saw that the Laplace transform was of the form
N (s)
X(s) = ,
D(s)
where N (s) and D(s) are polynomials in s. The roots of the polynomial N (s)
are called the zeros of X(s) (since X(s) will be zero when s is equal to one
of those roots), and the roots of D(s) are called the poles of X(s) (evaluating
X(s) at a pole will yield ∞). We can draw the poles and zeros in the s-plane
using ◦ for zeros and × for poles.
4 1
x(t) = δ(t) − e−t u(t) + e2t u(t).
3 3
The Laplace transform of δ(t) is
Z ∞
L{δ(t)} = δ(t)e−st dt = 1
−∞
for any value of s. Thus the ROC for δ(t) is the entire s-plane. Putting this
with the other two terms, we have
4 1 1 1 (s − 1)2
X(s) = 1 − + = ,
3s+1 3s−2 (s + 1)(s − 2)
with ROC Re{s} > 2.
8.2 The Region of Convergence

Let us dig a little deeper into the region of convergence for Laplace transforms.
Recall that for a given signal x(t), the ROC is the set of values s ∈ C such that
the Laplace transform integral converges. More specifically, writing s = σ + jω,
we see that Z ∞ Z ∞
x(t)e−st dt = x(t)e−σt e−jωt dt.

−∞ −∞
Thus, as long as x(t)e−σt is absolutely integrable, this integral exists. Note that
this does not depend on the value of ω. Thus, we have the following fact about
the ROC.
Property 1. The ROC consists of strips parallel to the

jω-axis in the s-plane.
For the next property of the ROC, suppose that the signal x(t) has a Laplace
transform given by a rational function. We know that the poles of this function
are the set of complex s such that X(s) is infinite. Since X(s) is given by the
Laplace transform integral, we see that the ROC cannot contain any poles of
X(s).
Property 2. For rational Laplace transforms, the ROC

does not contain any poles of X(s).
8.2 The Region of Convergence 99
The third property pertains to signals that are of finite duration (and absolutely
integrable). Specifically, suppose that x(t) is nonzero only between two finite
times T1 and T2 . Then we have
Z T2
X(s) = x(t)e−st dt
T1
which is finite for any finite s. Thus we have the following.
Property 3. If x(t) is of finite duration and absolutely

integrable, then the ROC is the entire s-plane.
Another way to think of the above property is as follows. No matter what σ

we pick, the signal x(t)e−σt will be absolutely integrable as long as x(t) is of
finite duration and absolutely integrable. The fact that x(t) is of finite duration
allows us to overcome the fact that the signal e−σt may be growing unboundedly
outside of the interval [T1 , T2 ].
While the previous property considered the case where the signal is of finite
duration, we will also be interested in signals that are only zero either before or
after some time. First, a signal x(t) is right-sided if there exists some T1 ∈ R
such that x(t) = 0 for all t < T1 . A signal x(t) is left-sided if there exists some
T2 ∈ R such that x(t) = 0 for all t > T2 . A signal x(t) is two-sided if it extends
infinitely far in both directions.
Property 4. If x(t) is right-sided and if the line

Re{s} = σ0 is in the ROC, then the ROC contains all
values s such that Re{s} ≥ σ0 .
To see why this is true, first note that since x(t) is right-sided, there exists some
T1 such that x(t) = 0 for all t < T1 . If s with Re{s} = σ0 is in the ROC, then
x(t)e−σ0 t is absolutely integrable, i.e.,
Z ∞
|x(t)|e−σ0 t dt < ∞.
T1
Now suppose that we consider some σ1 > σ0 . If T1 > 0, then e−σ1 t is always
smaller than e−σ0 t over the region of integration, and thus x(t)e−σ1 t will also
be absolutely integrable. If T1 < 0, then
Z ∞ Z 0 Z ∞
|x(t)|e−σ1 t dt = |x(t)|e−σ1 t dt + |x(t)|e−σ1 t dt
T1 T1 0
Z0 Z ∞
≤ |x(t)|e−σ1 t dt + |x(t)|e−σ0 t dt.
T1 0
The first term is finite (since it is integrating some signal of finite duration),
and the second term is finite since x(t)e−σ0 t is absolutely integrable. Thus, once
again, x(t)e−σ1 t is absolutely integrable, and thus s with Re{s} ≥ σ0 also falls
within the ROC of the signal.
The same reasoning applies to show the following property.
Property 5. If x(t) is left-sided and if the line Re{s} = σ0

is in the ROC, then the ROC contains all values s such that
Re{s} ≤ σ0 .
If x(t) is two-sided, we can write x(t) as x(t) = xR (t) + xL (t), where xR (t) is a
right-sided signal and xL (t) is a left-sided signal. The former has an ROC that
is the region to the right of some line in the s-plane, and the latter has an ROC
that is the region to the left of some line in the s-plane. Thus, the ROC for x(t)
contains the intersection of these two regions (if there is no intersection, x(t)
does not have a Laplace transform).
Property 6. If x(t) is two-sided and contains the line

Re{s} = σ0 in its ROC, then the ROC consists of a strip in
the s-plane that contains the line Re{s} = σ0 .
Example 8.5. Consider the signal x(t) = e−b|t| . We write this as
x(t) = e−bt u(t) + ebt u(−t).
Note that we modify the definition of u(t) in this expression so that u(0) = 21 ,
so that x(0) = 1 as required. As this modification is only at a single point
(of zero width and finite height), it will not make a difference to the quantities
calculated by integrating the signals.
The signal e−bt u(t) has Laplace transform
1
L{e−bt u(t)} = ,
s+b
with ROC Re{s} > −b. The signal ebt u(−t) has Laplace transform
−1
L{ebt u(−t)} = ,
s−b
with ROC Re{s} < b. If b ≤ 0, then these two ROCs do not overlap, in which
case x(t) does not have a Laplace transform. However, if b > 0, then x(t) has
the Laplace transform
1 1
L{x(t)} = − ,
s+b s−b
with ROC −b < Re{s} < b.
8.3 The Inverse Laplace Transform 101
As we will see soon, a rational Laplace transform X(s) can be decomposed into
a sum of terms, each of which correspond to an exponential signal. The ROC
for X(s) consists of the intersection of the ROCs for each of those terms, and
since none of the ROCs can contain poles, we have the following property.
Property 7. If X(s) is rational, then its ROC is bounded

by poles or extends to infinty.
Example 8.6. Consider

1
X(s) = .
s(s + 1)
There are three possible ROCs for this Laplace transform: the region to the
right of the line Re{s} = 0, the region between the lines Re{s} = −1 and
Re{s} = 0, or the region to the left of the line Re{s} = −1.
8.3 The Inverse Laplace Transform

Consider again the Laplace transform evaluated at s = σ + jω:
Z ∞
X(σ + jω) = x(t)e−σt e−jωt dt.
−∞
Since this is just the Fourier transform of x(t)e−σt , we can use the inverse Fourier
transform formula to obtain
Z ∞
−σt 1
x(t)e = X(σ + jω)ejωt dω.
2π −∞
If we multiply both sides by eσt , we get

Z ∞
1
x(t) = X(σ + jω)e(σ+jω)t dω.
2π −∞
Doing a change of variable s = σ + jω, we get
Z σ+∞
1
x(t) = X(s)est dω.
2π σ−j∞
This is the inverse Fourier transform formula. It involves an integration
over the line in the complex plane consisting of points satisfying Re{s} = σ.
There are actually simpler ways to calculate the inverse Fourier transform, using
the notion of partial fraction expansion, which we will consider here.
1
Example 8.7. Consider X(s) = s(s+1) . First, we note that
1 1 1
= − .
s(s + 1) s s+1
Now each of these terms is of a form that we know (they correspond to complex
exponentials). So, for example, if the ROC for X(s) is the region to the right of
the imaginary axis, since the ROC consists of the intersection of the ROCs of
both of the terms, we know that both terms must be right-sided signals. Thus,
x(t) = u(t) − e−t u(t).
Similarly, if the ROC is between the lines Re{s} = −1 and Re{s} = 0, the first
term is left-sided and the second term is right-sided, which means
x(t) = −u(−t) − e−t u(t).
Finally, if the ROC is to the right of the line Re{s} = −1, both terms are
left-sided and thus
x(t) = −u(−t) + e−t u(−t).
1
In the above example, we “broke up” the function s(s+1) into a sum of simpler
functions, and then applied the inverse Laplace Transform to each of them. This
is a general technique for inverting Laplace Transforms, which we now study.
8.3.1 Partial Fraction Expansion

Suppose we have a rational function
bm sm + bm−1 sm−1 + · · · + b1 s + b0 N (s)

X(s) = = ,
sn + an−1 sn−1 + · · · + a1 s + a0 D(s)
where the ai ’s and bi ’s are constant real numbers.
Definition 8.1. If m ≤ n, the rational function is called

proper. If m < n, it is strictly proper.
By factoring N (s) and D(s), we can write
(s + z1 )(s + z2 ) · · · (s + zm )
X(s) = K .
(s + p1 )(s + p2 ) · · · (s + pn )
Recall that the zeros of X(s) are given by −z1 , −z2 , . . . , −zm , and the poles are
−p1 , −p2 , . . . , −pn . First, suppose each of the poles are distinct and that X(s)
is strictly proper. We would like to write
k1 k2 kn
X(s) = + + ··· + ,
s + p1 s + p2 s + pn
8.4 Some Properties of the Laplace Transform 103
for some constants k1 , k2 , . . . , kn , since the inverse Laplace Transform of X(s)

is easy in this form. How do we find k1 , k2 , . . . , kn ?
Heaviside’s Cover-up Method. To find the constant ki , multiply both sides
of the expansion of X(s) by (s + pi ):
k1 (s + pi ) k2 (s + pi ) kn (s + pi )
(s + pi )X(s) = + + · · · + ki + · · · + .
s + p1 s + p2 s + pn
Now if we let s = −pi , then all terms on the right hand side will be equal to
zero, except for the term ki . Thus, we obtain
ki = (s + pi )X(s)|s=−pi .
s+5
Example 8.8. Consider X(s) = s3 +3s2 −6s−8 . The denominator is factored as
s3 + 3s2 − 6s − 8 = (s + 1)(s − 2)(s + 4).
We would thus like to write

s+5 k1 k2 k3
X(s) = = + + .
(s + 1)(s − 2)(s + 4) s+1 s−2 s+4
Using the Heaviside coverup rule, we obtain
4 4
k1 = (s + 1)X(s)|s=−1 = =−
(−3)(3) 9
7 7
k2 = (s − 2)X(s)|s=2 = =
(3)(6) 18
1 1
k3 = (s + 4)X(s)|s=−4 = = .
(−3)(−6) 18
The partial fraction expansion when some of the poles are repeated is obtained
by following a similar procedure, but it is a little more complicated. We will
not worry too much about this scenario here. One can also do a partial fraction
expansion of nonstrictly proper functions by first dividing the denominator into
the numerator to obtain a constant and a strictly proper function, and then
applying the above partial fraction expansion.
8.4 Some Properties of the Laplace Transform

The Laplace transform has various properties that are quite similar to those for
Fourier transforms (linearity, time-shifting, etc.) We will focus on two important
ones here.
8.4.1 Convolution
Consider two signals x1 (t) and x2 (t) with Laplace transforms X1 (s) and X2 (s)
and ROCs R1 and R2 , respectively. Then
L{x1 (t) ∗ x2 (t)} = X1 (s)X2 (s),
with ROC containing R1 ∩ R2 . Thus, convolution in the time-domain maps to
multiplication in the s-domain (as was the case with Fourier transforms).
1
Example 8.9. Consider the convolution u(t) ∗ u(t). Since L{u(t)} = s with
ROC Re{s} > 0, we have
1
L{u(t) ∗ u(t)} = ,
s2
with ROC containing the region Re{s} > 0.
8.4.2 Differentiation
Consider a signal x(t), with Laplace transform X(s) and ROC R. We have
Z
1
x(t) = X(s)est ds.
2π
Differentiating both sides with respect to t, we have
Z
dx(t) 1
= sX(s)est ds.
dt 2π
Thus, we see that
dx
L{ } = sX(s),
dt
with ROC containing R. More generally,
dm x
L{ } = sm X(s) .
dtm
8.4.3 Integration
Given a signal x(t) whose Laplace transform has ROC R, consider the integral
Rt
−∞
x(τ )dτ . Note that
Z t
x(τ )dτ = u(t) ∗ x(t),
−∞
and thus using the convolution property, we have

Z t
1
L{ x(τ )dτ } = X(s),
−∞ s
with ROC containing R ∩ {Re{s} > 0}.
8.5 Finding the Output of an LTI System via Laplace Transforms105
8.5 Finding the Output of an LTI System via

Laplace Transforms
The Laplace transform properties (time-domain convolution corresponding to
frequency-domain multiplication, in particular) are very useful in analyzing the
output of LTI systems to inputs. Specifically, consider an LTI system with
impulse response h(t) and input y(t). We know that
Y (s) = H(s)X(s),
assuming all Laplace transforms exist. Using the expressions for H(s) and X(s),
we can thus calculate Y (s) (and its ROC), and then use an inverse Laplace
transform to determine y(t).
Example 8.10. Consider an LTI system with impulse response h(t) = e−2t u(t).
Suppose the input is x(t) = e−3t u(t). The Laplace transforms of h(t) and x(t)
are
1 1
H(s) = , X(s) = ,
s+2 s+3
with ROCs Re{s} > −2 and Re{s} > −3, respectively. Thus we have
1 1
Y (s) = H(s)X(s) = ,
s+2s+3
with ROC Re{s} > −2. Using partial fraction expansion, we have
1 1
Y (s) = − ,
s+2 s+3
and thus y(t) = e−2t u(t) − e−3t u(t).
Example 8.11. Consider an LTI system with impulse response h(t) = −e4t u(−t)
and input x(t) = e2t u(t), where we interpret u(−t) as being 1 for t < 0. The
Laplace transforms are
1 1
H(s) = , X(s) = ,
s−4 s−2
with ROCs Re{s} < 4 and Re{s} > 2, respectively. Since there is a nonempty
intersection, we have
1 1 1 1 1 1
Y (s) = H(s)X(s) = = − ,
s−4s−2 2s−4 2s−2
with ROC 2 < Re{s} < 4. Thus, y(t) is two-sided, and given by
1 1
y(t) = − e4t u(−t) + e2t u(t).
2 2
8.6 Finding the Impulse Response of a Differen-

tial Equation via Laplace Transforms
The differentiation property of Laplace transforms is also extremely useful for
analyzing differential equations. Specifically, suppose that we have a constant-
coefficient differential equation of the form
n m
X dk y(t) X dk x(t)
ak = bk
dtk dtk
k=0 k=0
Taking Laplace transforms, we obtain

n
X m
X
( ak sk )Y (s) = ( bk sk )X(s)
k=0 k=0
or equivalently Pm
bk sk
Y (s) = Pnk=0 k
X(s).
k=0 ak s
| {z }
H(s)
Thus, the impulse response of the differential equation is just the inverse Laplace
transform of H(s) (corresponding to an appropriate region of convergence).
d3 y(t) d2 y(t) dy(t)

3
+2 − − 2y(t) = x(t).
dt dt2 dt
Taking Laplace transforms, we have
Y (s) 1 1
H(s) = = 3 = .
X(s) s + 2s2 − s − 2 (s − 1)(s + 1)(s + 2)
In this case, we have the partial fraction expansion

1 1 1 1 1 1
H(s) = − +
6s−1 2s+1 6s+2
Suppose we are told the impulse response is causal (which implies it is right-
sided). Thus, the ROC would be to the right of the furthest pole and we have

1 t 1 −t 1 −2t
h(t) = e − e + e u(t).
6 2 6
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ECE 301: Signals and Systems

Prof. Shreyas Sundaram

School of Electrical and Computer Engineering

• Linear Systems and Signals by B. P. Lathi

I claim credit for all typos and mistakes in the notes.

2 Properties of Signals and Systems 5

3 Analysis of Linear Time-Invariant Systems 21

3.3.4 Memoryless LTI Systems . . . . . . . . . . . . . . . . . . 26

4 Fourier Series Representation of Periodic Signals 37

5 The Continuous-Time Fourier Transform 59

6 The Discrete-Time Fourier Transform 75

8 The Laplace Transform 95

1.1 Signals and Systems

Figure 1.1: An abstract representation of a system.

Examples of systems and associated signals:

• Electrical circuits: voltages, currents, temperature,...

• Computer systems: Internet traffic, user input, ...

From a mathematical perspective, signals can be regarded as functions of one

• A continuous-time signal is a function of the form f (t), where t ranges

Figure 1.2: Block Diagram of a feedback control system.

Example 1.2 (Inverted Pendulum). Suppose we try to balance a stick vertically

1.2 Outline of This Course

• Signal and systems classifications: develop terminology and identify useful

Properties of Signals and

2.1 Signal Energy and Power

The average power dissipated over the time-interval [t1 , t2 ] is then

for continuous-time signals, and

and for a discrete-time signal as

2.2 Transformations of Signals

2.3 Periodic, Even and Odd Signals

Given a signal f (t), define the signals

2.4 Exponential and Sinusoidal Signals

Given a complex exponential with fundamental frequency ω0 , a harmonically

In other words, it is the set of complex exponentials whose frequencies are

φk (t + T0 ) = ejkω0 (t+T0 ) = ejkω0 T0 ejkω0 t = ejkm2π φk (t) = φk (t).

Aej(ω0 t+φ) = A cos(ω0 t + φ) + jA sin(ω0 t + φ).

Similarly, we can write

2.4.2 Discrete-Time Complex Exponential Signals

f [n + N0 ] = ejω0 (n+N0 ) = ejω0 n ejω0 N0 = f [n]ejω0 N0 .

Due to periodicity, we must have ejω0 N0 = 1, or equivalently, ω0 N0 = 2πk for

g[n] = ejω0 n ej2πn = ejω0 n = f [n],

i.e., the discrete-time complex exponential with frequency ω0 + 2π is the same

ejω0 n = cos(ω0 n) + j sin(ω0 n).

As ω0 increases from 0 to π, the frequencies of both the sinusoids increase.1 For

ej(2π−ω0 )n = e−jω0 n = cos(ω0 n) − j sin(ω0 n).

2.5 Impulse and Step Functions

The discrete-time unit step signal is defined as

Note that by the time-shifting property, we have

δ[n] = u[n] − u[n − 1]

f [n]δ[n − k] = f [k]δ[n − k].

More generally, for any signal f [n], we have

Note that u(t) is discontinuous at t = 0 (we will take it to be continuous from

δ(t) = lim δ (t).

In other words, if we take an infinite sequence of shifted impulse functions, scale

Just as the discrete-time impulse function could be viewed as a difference of

2.6 Properties of Systems

2.6.1 Interconnections of Systems

We will often be interested in connecting different systems together in order to

An example of a series interconnection of systems occurs in communication

δ(t) = lim δ (t).