EE351: Spectrum Analysis and Discrete Time Systems (Signals, Systems and Transforms)

EE351: Spectrum Analysis and Discrete Time Systems
(Signals, Systems and Transforms)

Dr. Ha H. Nguyen
Associate Professor
Department of Electrical Engineering
University of Saskatchewan
August 2005
EE351Spectrum Analysis and Discrete Time Systems University of Saskatchewan
Introduction
The concepts of signals and systems arise in a wide variety of areas, such as
communications, circuit design, biomedical engineering, power systems, speech
processing, etc.
The ideas and techniques associated with these concepts play an important role
in such diverse areas.
Although the physical nature of the signals and systems that arise in these
various disciplines may be drastically dierent, two basic features in common are:
The signals, which are functions of one or more independent variables,
contain information about the behavior or nature of some phenomenon.
The systems respond to particular signals by producing other signals or some
desired behavior.
Examples of signals and systems:
Voltages and currents as functions of time in an electrical circuit are
examples of signals. A circuit is itself an example of a system, which
responds to applied voltages and currents.
A camera is a system that receives light from dierent sources and produces
a photograph.
Dr. H. Nguyen Page 1
Example problems of signal and system analysis:
Analyzing existing systems: We are presented with a specic system and are
interested in characterizing it in detail to understand how it will respond to
various inputs (analysis of a circuit).
Designing systems to process signals in particular ways. For example, to
design systems to enhance or restore signals that have been degraded in
some way (image restoration, image enhancement).
Designing systems to extract specic pieces of information from signals.
Examples include the estimation of heart rate from an electrocardiogram,
weather forecast.
Designing of signals with particular properties. For example, the design of
communication signals must take into account the need for reliable reception
in the presence of distortion (due to transmission media) and interference
(such as noise).
Chapter I: Signals and Systems
There is an analytical frameworkthat is, a language for describing signals and
systems and an extremely powerful set of tools for analyzing themthat applies
equally well to problems in many elds.
This chapter begins the development of such an analytical framework for signals and
systems by introducing their mathematical description and representations.
Signals are represented mathematically as functions of one or more
independent variables.
Examples:
A speech signal can be represented mathematically by acoustic pressure as a
function of time.
A picture can be represented by brightness as a function of two spatial variables.
This course focuses only on signals involving a single independent variable.
For convenience, the independent variable will generally referred to as time, although
it may not in fact represent time in specic applications.
Examples of Signals
Arterial Blood Pressure
0 0.5 1 1.5 2 2.5
60
70
80
90
100
110
120
Time (sec)
A
B
P

(
m
m
H
g
)
Portland State University ECE 222 Signal Fundamentals Ver. 1.06 7
Microelectrode Recording
2 2.01 2.02 2.03 2.04 2.05 2.06
0.6
0.4
0.2
0
0.2
0.4
0.6
0.8
Time (sec)
Speech
1.2 1.25 1.3 1.35 1.4 1.45 1.5 1.55 1.6 1.65
0.4
0.3
0.2
0.1
0
0.1
0.2
0.3
0.4
Time (sec)
Linus: Philosophy of Wet Suckers
Electrocardiogram
0 0.5 1 1.5 2 2.5
6.5
7
7.5
8
8.5
Time (sec)
3
1
1
.
4
.
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Discrete-time & Continuous-time Signals
The course will cover concepts and techniques associated both with
continuous-time and discrete-time signals (and systems).
Continuous-time signals are dened for a continuum of values of the
independent variable (the independent variable is continuous).
Will always be treated as a function of t.
Parentheses are used to denote continuous-time functions, for example x(t).
The independent variable t is a real-valued and continuous.
Discrete-time signals are only dened at discrete times (the independent variable
takes on only a discrete set of values).
Will always be treated as a function of n.
Square brackets are used to denote discrete-time functions, for example x[n].
The independent variable n is an integer.
Examples:
The speech signal as a function of time and atmospheric pressure as a
function of altitude are examples of continuous time signals.
The daily closing stock market index is an example of discrete-time signal.
It is often useful to represent the signals graphically:
Arterial Blood Pressure
0 0.5 1 1.5 2 2.5
60
70
80
90
100
110
120
Time (sec)
A
B
P

(
m
m
H
g
)
Microelectrode Recording
2 2.01 2.02 2.03 2.04 2.05 2.06
0.6
0.4
0.2
0
0.2
0.4
0.6
0.8
Time (sec)
Speech
1.2 1.25 1.3 1.35 1.4 1.45 1.5 1.55 1.6 1.65
0.4
0.3
0.2
0.1
0
0.1
0.2
0.3
0.4
Time (sec)
Linus: Philosophy of Wet Suckers
Electrocardiogram
0 0.5 1 1.5 2 2.5
6.5
7
7.5
8
8.5
Time (sec)
Figure 1: A continuous-time signal (electrocardiogram).
3
Discretetime signals
Symbol n for independent variable
Use brackets []
Discretetime signal: x[n]
Graphical representation
x[0]
x[1]
x[1]
0 3 2 1
x[2] x[2]
n
3 2 1
5 4
x[n]
Lampe, Schober: Signals and Communications
4
1.1.2 Signal Energy and Power
Often classication of signals according to energy and power
Terminology energy and power used for any signal x(t), x[n]
Need not necessarily have a physical meaning
Signal energy
Energy of a possibly complex continuoustime signal x(t) in
interval t
1
t t
2
E(t
1
, t
2
) =
t
2
t
1
|x(t)|
2
dt
Energy of a possibly complex discretetime signal x[n] in interval
n
1
n n
2
E(n
1
, n
2
) =
n
2
n=n
1
|x[n]|
2
Total energy
E
= E(, ) =
|x(t)|
2
dt
E
= E(, ) =
n=
|x[n]|
2
Figure 2: An example of discrete-time signal.
Continuous-time signals (and systems) have very strong roots in problems
associated with physics, and, more recently, with electrical circuits and
communications.
The techniques of discrete-time signals (and systems) have strong roots in
numerical analysis, statistics and time series analysis (associated with such
applications as the analysis of economic and demographic data).
In the past decades, there has been a growing interrelationship between
continuous-time signals and systems and discrete-time signals and systems. This
has come from the dramatic advances in technology for the implementation of
systems and for the generation of signals. For example, it is increasingly
advantageous to consider processing continuous-time signals by representing
them with time samples.
This course develops the concepts of continuous-time and discrete-time signals
and systems in parallel. Since many of the concepts are similar, by treating them
in parallel, insight and intuition can be shared and both the similarities and
dierences between them become better focused.
Signal Energy and Power
In many (but not all) applications, the signals are directly related to physical
quantities that capturing power and energy in a physical system.
Example: Let v(t) and i(t) be the voltage and current across a resistor with
resistance R. Then the instantaneous power is
p(t) = v(t)i(t) =
1
R
v
2
(t)
The total energy expended over the time interval t
1
t t
2
is
_
t
2
t
1
p(t)dt =
_
t
2
t
1
1
R
v
2
(t)dt
The average power over this time interval is
1
t
2
t
1
_
t
2
t
1
p(t)dt =
1
t
2
t
1
_
t
2
t
1
1
R
v
2
(t)dt
For most of this course we will use a broad denition of power and energy that
applies to any signal x(t) or x[n]. Such denitions need not necessarily have a
physical meaning.
Signal energy
The total energy over the time interval t
1
t t
2
of a possibly complex
continuous-time signal x(t) is:
E(t
1
, t
2
) =
_
t
2
t
1
|x(t)|
2
dt
Similarly, the total energy of a possibly complex discrete-time signal x[n] over
the time interval n
1
n n
2
is:
E(n
1
, n
2
) =
n
2
n=n
1
|x[n]|
2
The total energy is the energy in a signal over an innite time interval:
E
= E(, ) =
_

|x(t)|
2
dt
E
= E(, ) =
n=
|x[n]|
2
Example: Find the total energy of the following discrete-time signal:
x[n] =
_
_
_
a
n
, n 0
0, n < 0
where |a| < 1. The answer is:
E
n=
|x[n]|
2
=
n=0
(|a|
2
)
n
=
1
1 |a|
2
Signal power
Consider the time-averaged signal power.
The average powers of x(t) and x[n] over the intervals t
1
t t
2
and
n
1
n n
2
are:
P(t
1
, t
2
) =
1
t
2
t
1
_
t
2
t
1
|x(t)|
2
dt and P(n
1
, n
2
) =
1
n
2
n
1
+ 1
n
2
n=n
1
|x[n]|
2
respectively.
Analogously, when the limits are taken over an innite time interval, then:
P
= lim
T
1
2T
_
T
T
|x(t)|
2
dt, P
= lim
N
1
2N + 1
N
n=N
|x[n]|
2
Classication of signals based on their energy and power
Signals with nite total energy, E
< , are known as energy signals.

The energy signals have zero average power: P
= 0.
Examples of energy signals: All signals seen previously, any signal with
nite amplitude and nite duration (x(t) = 0 for |t| > and
max(|x(t)|) < ).
Signals with nite average power, P
> 0, are known as power signals.

The power signals have innite total energy: E
= if P
> 0.
Examples of power signals: periodic signals such as x(t) = cos(t) and
x[n] = sin(5n).
Signals with innite power (P
= ) and innite energy (E
= ).
These signals are not desirable in engineering applications.
Examples: x(t) = e
t
and x[n] = n
5
.
Signal Transformations
Transformation of a signal is a central concept in signal and system analysis. For
example, an audio system takes an input signal representing music recorded on a
compact disc, and modies it to enhance desirable characteristics.
This section focuses on a very limited but important class of elementary signal
transformations that involve simple modication of the independent variable
(i.e., the time axis).
Elementary signal transformations:
Time shift: x(t) x(t t
0
) and x[n] x[n n
0
]
If t
0
> 0 or n
0
> 0, signal is shifted to the right (i.e., delayed)
If t
0
< 0 or n
0
< 0, signal is shifted to the left (i.e., advanced)
Time reversal : x(t) x(t) and x[n] x[n]
Time scaling: x(t) x(t) and x[n] x[n]
If > 1, signal appears compressed
If < 1, signal appears stretched
Example: The following gure plots x(t) and its transformations x(t),
x(t 1), x(t + 2) and x(t/2).
5 4 3 2 1 0 1 2 3 4 5
1
0
1
t
x(t)
5 4 3 2 1 0 1 2 3 4 5
1
0
1
t
x(t)
5 4 3 2 1 0 1 2 3 4 5
1
0
1
t
x(t1)
5 4 3 2 1 0 1 2 3 4 5
1
0
1
t
x(t)
5 4 3 2 1 0 1 2 3 4 5
1
0
1
t
x(t+2)
5 4 3 2 1 0 1 2 3 4 5
1
0
1
t
x(t/2)
5 4 3 2 1 0 1 2 3 4 5
1
0
1
t
x(t)
5 4 3 2 1 0 1 2 3 4 5
1
0
1
t
x(2t)
5 4 3 2 1 0 1 2 3 4 5
1
0
1
t
x(2(t1))
More Example:
7
1.1.3 Transformations of the Independent Variable
Time shift
Replace t t t
0
x(t) x(t t
0
)
n n n
0
x[n] x[n n
0
]
Delay: t
0
, n
0
> 0, Advance: t
0
, n
0
< 0
Time reversal
Replace t t x(t) x(t)
n n

x[n] x[n]
Time scaling
Replace t t , IR x(t) x(t)
n n , ZZ

x[n] x[n]
Continuoustime case: || < 1 : signal is linearly stretched
|| > 1 : signal is linearly compressed
Time shift, time reversal, and time scaling operations arise naturally
in the processing of signals
8
Example:
n
n t
n
n
t
t
Time-scaled signals
Time-reversed signals
Time-shifted signals
Signals
t
x[2n]
x[n]
x[n 4]
x(2/3t)
x(t)
x(t t
0
)
x(t) x[n]
t
0
A more general signal transformation is x(t) y(t) = x(t +) . For this
transformation, a systematic approach to obtain the plot of y(t) is as follows:
First shift x(t) in accordance with the value of : The signal x(t) is shifted to
the right if < 0, shifted to the left if > 0.
Then perform time scaling and/or time reversal on the resulting signal in
accordance with the value of : The resulting signal is linearly stretched if
|| < 1, linearly compressed if || > 1 and reversed in time if < 0.
Example: The following gures draw x(t), x(2t) and x(2(t 1)).
5 4 3 2 1 0 1 2 3 4 5
1
0
1
t
x(t)
5 4 3 2 1 0 1 2 3 4 5
1
0
1
t
x(2t)
5 4 3 2 1 0 1 2 3 4 5
1
0
1
t
x(2(t1))
Even & Odd Symmetry
There is a set of useful properties of signals that relate to their symmetry under time
reversal.
Even signal: x(t) = x(t) or x[n] = x[n].
Examples: The signal plotted below, cos[k
0
n].
9
1.1.4 Periodic Signals
Periodic continuoustime signal
x(t) = x(t + T) , t
T > 0: Period
x(t) periodic with T x(t) also periodic with mT, m IN
Smallest period of x(t): Fundamental period T
0
.
Example (T
0
= T):
0
x(t)
t
4T 3T 2T 3T 2T T T
Periodic discretetime signal
x[n] = x[n + N] , n
Integer N > 0: Period
x[n] periodic with N x[n] also periodic with mN, m IN
Smallest period of x[n]: Fundamental period N
0
.
Example (N
0
= 4):
n
3 6
x[n]
0 1
2
5 4
A signal that is not periodic is referred to as aperiodic.
10
1.1.5 Even and Odd Signals
Even signal
x(t) = x(t) or x[n] = x[n]
Example:
x(t)
t
Odd signal
x(t) = x(t) or x[n] = x[n]
Example:
n
x[n]
Necessarily: x(0) = 0 or x[0] = 0
Decomposition of any signal into an even and odd part:
x(t) = Ev{x(t)} + Od{x(t)} or x[n] = Ev{x[n]} + Od{x[n]}
with
Ev{x(t)} =
1
2
(x(t) + x(t)) or Ev{x[n]} =
1
2
(x[n] + x[n])
and
Od{x(t)} =
1
2
(x(t) x(t)) or Od{x[n]} =
1
2
(x[n] x[n])
Odd signal: x(t) = x(t) or x[n] = x[n]. Note that an odd signal must
be zero at t = 0 or n = 0.
Examples: The signal plotted below, sin(k
0
t).
9
x(t) = x(t + T) , t
T > 0: Period
0
.
Example (T
0
= T):
0
x(t)
t
4T 3T 2T 3T 2T T T
x[n] = x[n + N] , n
0
.
Example (N
0
= 4):
n
3 6
x[n]
0 1
2
5 4
10
Even signal
x(t) = x(t) or x[n] = x[n]
Example:
x(t)
t
Odd signal
x(t) = x(t) or x[n] = x[n]
Example:
n
x[n]
with
Ev{x(t)} =
1
2
(x(t) + x(t)) or Ev{x[n]} =
1
2
(x[n] + x[n])
and
Od{x(t)} =
1
2
1
2
(x[n] x[n])
Any signal can be written as a sum of an odd signal and an even signal:
x(t) = x
e
(t) +x
o
(t), where
x
e
(t) =
1
2
[x(t) +x(t)]
x
o
(t) =
1
2
[x(t) x(t)]
5 4 3 2 1 0 1 2 3 4 5
1
0
1
t
x(t)
5 4 3 2 1 0 1 2 3 4 5
1
0
1
t
x(t)
5 4 3 2 1 0 1 2 3 4 5
1
0
1
t
x
e
(t)
5 4 3 2 1 0 1 2 3 4 5
1
0
1
t
x(t)
5 4 3 2 1 0 1 2 3 4 5
1
0
1
t
x(t)
5 4 3 2 1 0 1 2 3 4 5
1
0
1
t
x
o
(t)
Periodic Signals
An important class of signals that are encountered frequently in this course is the
class of periodic signals.
A signal is periodic if there is a positive value of T or N such that
x(t) = x(t +T) or x[n] = x[n +N]
T and N are called the periods of x(t) and x[n], respectively.
For any integer m, mT and mN are also the periods.
For continuous-time signals, the fundamental period T
0
is the smallest positive
value of T such that x(t) = x(t +T). For a special case where x(t) is a
constant, the fundamental period is undened.
For discrete-time signals, the fundamental period N
0
is the smallest positive
integer of N such that x[n] = x[n +N].
Signals that are not periodic are said to be aperiodic.
Examples of periodic signals are shown below:
9
x(t) = x(t + T) , t
T > 0: Period
0
.
Example (T
0
= T):
0
x(t)
t
4T 3T 2T 3T 2T T T
x[n] = x[n + N] , n
0
.
Example (N
0
= 4):
n
3 6
x[n]
0 1
2
5 4
10
Even signal
x(t) = x(t) or x[n] = x[n]
Example:
x(t)
t
Odd signal
x(t) = x(t) or x[n] = x[n]
Example:
n
x[n]
with
Ev{x(t)} =
1
2
(x(t) + x(t)) or Ev{x[n]} =
1
2
(x[n] + x[n])
and
Od{x(t)} =
1
2
1
2
(x[n] x[n])
9
x(t) = x(t + T) , t
T > 0: Period
0
.
Example (T
0
= T):
0
x(t)
t
4T 3T 2T 3T 2T T T
x[n] = x[n + N] , n
0
.
Example (N
0
= 4):
n
3 6
x[n]
0 1
2
5 4
10
Even signal
x(t) = x(t) or x[n] = x[n]
Example:
x(t)
t
Odd signal
x(t) = x(t) or x[n] = x[n]
Example:
n
x[n]
with
Ev{x(t)} =
1
2
(x(t) + x(t)) or Ev{x[n]} =
1
2
(x[n] + x[n])
and
Od{x(t)} =
1
2
1
2
(x[n] x[n])
Elementary Signals: Complex Exponential and Sinusoidal
Several classes of signals play prominent role:
They model many physical signals.
They serve as building blocks for many other signals.
They serve for system analysis.
1. Continuous-Time Complex Exponential Signal: x(t) = Ce
at
where, in
general, both C and a are complex numbers.
If both C and a are real Real exponential signal.
Example: The following gures plot Ce
at
with C = 1 and a =
1
10
.
10 5 0 5 10 15 20 25 30
0
5
10
15
20
25
t
10 5 0 5 10 15 20 25 30
0
0.5
1
1.5
2
2.5
3
t
If a is imaginary (i.e., a = j
0
) Periodic complex exponential.
To see that the signal is indeed periodic, let C = Ae
j
. Then
x(t) = Ae
j(
0
t+)
?
= Ae
j(
0
(t+T)+)
= Ae
j(
0
t+)
e
j
0
T
where T is chosen such that e
j
0
T
= 1. Excluding the trivial solution of
0
= 0, the fundamental period is T
0
=
2
|
0
|
.
Example: The following gures plot Ce
at
with C = 1 and a = j.
10
0
10
20
30
1
0
1
1
0.5
0
0.5
1
Imaginary Part
Time (s)
R
e
a
l

P
a
r
t
A harmonically related set of complex periodic exponentials is a set of
exponentials with fundamental frequencies that are all multiples of a single
positive frequency
0
:
k
(t) = e
jk
0
t
where k = 0, 1, 2, . . .
For k = 0,
0
(t) is a constant
For all other values of k,
k
(t) is periodic with fundamental frequency
|k|
0
and fundamental period
2
|k|
0
=
T
0
|k|
This is consistent with how the term harmonic is used in music
Sets of harmonically related complex exponentials are used to
represent many other periodic signals (Fourier series)
Example: The gure below plots the real parts of several harmonically
related complex exponentials.
20 15 10 5 0 5 10 15 20
1
0
1
0
(
t
)
20 15 10 5 0 5 10 15 20
1
0
1
1
(
t
)
20 15 10 5 0 5 10 15 20
1
0
1
2
(
t
)
20 15 10 5 0 5 10 15 20
1
0
1
3
(
t
)
20 15 10 5 0 5 10 15 20
1
0
1
4
(
t
)
Time (s)
General complex exponential signal: For the most general case, C = Ae
j
and a = r +j
0
, then
x(t) = Ce
at
= Ae
rt
e
j(
0
t+)
= Ae
rt
cos(
0
t +) +jAe
rt
sin(
0
t +)
If r > 0, then x(t) is exponentially growing signal
If r < 0, then x(t) is exponentially decaying signal
Example: The following gures plot the real parts of Ce
at
with C = 1 and
a = 0.1 +j0.5.
20 15 10 5 0 5 10 15 20
8
6
4
2
0
2
4
6
8
t
R
e
a
l

P
a
r
t

o
f

x
(
t
)
20 15 10 5 0 5 10 15 20
8
6
4
2
0
2
4
6
8
t
R
e
a
l

P
a
r
t

o
f

x
(
t
)
2. Continous-Time Sinusoidal Signals
x
c
(t) = Acos(
0
t +) = Re{Ae
j(w
0
t+)
}
x
s
(t) = Asin(
0
t +) = Im{Ae
j(w
0
t+)
}
Of course, both x
c
(t) and x
s
(t) also have fundamental period T
0
=
2
|
0
|
.
Periodic signals have innite total energy, but nite average power. This
can be seen for the exponential x(t) = Ae
j
0
t
(assuming A is real) as follows:
The energy over one period T
0
is
E(0, T
0
) =
_
T
0
0
A
2
|e
j
0
t
|
2
dt = A
2
T
0
Thus, the total energy is E
= .
The average power over one period is
P(0, T
0
) =
E(0, T
0
)
T
0
= A
2
The average power is
P
= lim
T
1
2T
_
T
T
A
2
|e
j
0
t
|
2
dt = A
2
2T
2T
= A
2
.
3. Discrete-Time Complex Exponential and Sinusoidal Signals
Complex exponential signal:
x[n] = C
n
= Ce
n
(where = e
)
Real complex signal if both C and are real.
General complex exponential signal: With C = Ae
j
and = ||e
j
0
, then
x[n] = A||
n
e
j(
0
n+)
= A||
n
cos(
0
n +) +jA||
n
sin(
0
n +)
If || > 1, then x[n] is exponentially growing signal
If || < 1, then x[n] is exponentially decaying signal
If || = 1, then
x[n] = Ae
j(
0
n+)
= Acos(
0
n +) +jAsin(
0
n +)
Sinusoidal signal
x
c
[n] = Acos(
0
n +) = Re{Ae
j(
0
n+)
}
x
s
[n] = Asin(
0
n +) = Im{Ae
j(
0
n+)
}
The functions Ae
j(
0
n+)
, Acos(
0
n +) and Asin(
0
n +) are
discrete-time signals with nite average power but innite total energy.
Example: The gures below plot the real parts of Ce
n
for C = 1 and
= 0.1 +j0.5.
20 15 10 5 0 5 10 15 20
10
5
0
5
10
Time Index (n)
R
e
a
l

P
a
r
t
20 15 10 5 0 5 10 15 20
10
5
0
5
10
Time Index (n)
R
e
a
l

P
a
r
t
DT complex exponential vs. CT complex exponential: There are
important dierences between the properties of continuous-time and
discrete-time exponential signals e
j
0
t
and e
j
0
n
.
The DT exponential signals are not distinct for distinct values of
0
:
x[n] = e
j
0
n
= e
j(
0
+k2)n
, k = 0, 1, 2, . . .
Need only consider a frequency interval of 2 for
0
, typically
0
0
< 2.
As
0
increases from 0, the signals oscillate more and more rapidly until
0
= . As we continue to increase
0
, we decrease the rate of
oscillation until
0
= 2.
Low-frequency exponentials have
0
near 0, 2 and other even multiples
of .
High-frequencies are near and other odd multiples of .
The exponential e
j
0
n
is periodic if
0
/2 is a rational number:
x[n] = e
j
0
n
= e
j
0
(n+N)

0
2
=
m
N
, for some integer m
If
0
= 0 and if N and m have no factors in common, then N is the
fundamental period and the fundamental frequency is
2
N
=

0
m
.
Examples:
(a) Let x(t) = cos(8t/35).
Then
0
=
The fundamental period T
0
= 2/
0
=
(b) Let x[n] = cos(8n/35).
Then
0
=
If the signal is periodic?
The fundamental period N
0
= m(2/
0
) =
for m =
(c) x[n] = cos(n/6).
Then
0
=
If the signal is periodic?
The fundamental period N
0
= m(2/
0
) =
for m =
Discerete-Time Complex Exponential Harmonics: A harmonically related set
of discrete-time complex exponentials is a set of exponentials with a common
period N:
k
[n] = e
jk(2/N)n
where k = 0, 1, 2, . . .
All the harmonics are not all distinct for all the values of k:
k+N
[n] = e
j(k+N)(2/N)n
= e
jk(2/N)n
e
j2n
= e
jk(2/N)n
=
k
[n]
There are only N distinct periodic exponentials:
0
[n] = 1
1
[n] = e
j2n/N
2
[n] = e
j4n/N

N1
[n] = e
j2(N1)n/N
Discrete-Time Unit Impulse and Unit Step
The discrete-time unit impulse is dened as [n] =
_
_
_
0, n = 0
1, n = 0
-
6
r
r r r r r r r r
1
[n]
n
It is also known as the unit sample or Kronecker delta
It is an even function: [n] = [n]
The discrete-time unit step is dened as u[n] =
_
_
_
0, n < 0
1, n 0
-
6
r r r r r
r r r r
1
u[n]
n
There is a close relationship between [n] and u[n]
First order dierence:
[n] = u[n] u[n 1]
Running sum:
u[n] =
n
k=
[k]
The unit impulse can be used to sample the discrete time signal x[n] (sampling
property):
x[n][n n
0
] = x[n
0
][n n
0
]
This ability to use the unit impulse to extract a single value of x[n] through
multiplication will play an important role later.
Continuous-Time Unit Step
-
6
1
u(t)
t
The continuous-time unit step is dened as
u(t) =
_
_
_
0, t < 0
1, t > 0
Discontinuous at t = 0.
u(0) is not dened.
Not of consequence because it is undened for an innitesimal period of time.
Continuous-Time Unit Impulse
Unit Step for Switches
v
s
Linear
Circuit
t=0
v
s
u(t)
Linear
Circuit
Linear
Circuit
t=0
i
s
i
s
u(t)
Linear
Circuit
u(t) useful for representing the opening or closing of switches
We will often solve for or be given initial conditions at t = 0
We can then represent independent sources as though they were
immediately applied at t = 0. More later.
Discrete-Time Basis Functions
There is a close relationship between [n] and u[n]
[n] = u[n] u[n 1]
u[n] =
n
k=
[k]
u[n] =
k=0
[n k]
The unit impulse can be used to sample a discrete-time signal
x[n]:
x[0] =
k=
x[k][k] x[n] =
k=
x[k][n k]
This ability to use the unit impulse to extract a single value of x[n]
through multiplication will play an important role later in the term
Continuous-Time Unit Impulse
t
u
e
(t)
t
-e e
-e e
t
u(t)
t
1 1
e
(t)
(t)

e
(t)
du
e
(t)
dt
As e 0 ,
u
e
(t) u(t)

e
(t) for t = 0 becomes very large

e
(t) for t = 0 becomes zero
(t) lim
e0
e
(t)
Continuous-Time Unit Step
t
u(t)
1
u(t)
0 t < 0
1 t > 0
Sometimes known as the Heaviside function
Discontinuous at t = 0
u(0) is not dened
Not of consequence because it is undened for an innitesimal
period of time
Dene
e
(t)
du
e
(t)
dt
As e 0:
u
e
(t) u(t)
e
(t) for t = 0 becomes very large
e
(t) for t = 0 becomes zero
The continuous-time unit impulse function (also known as Dirac delta impulse)
is dened as (t) = lim
e0
e
(t).
Continuous-Time Unit Impulse (Continued)
-
6
6
1
(t)
t
An equivalent denition of the unit-impulse function:
(t)
_
_
_
0, t = 0
, t = 0
and
_
e
e
(t)dt = 1 for any e > 0
The function is zero everywhere, except zero.
The most important property of an impulse is its area. The impulse area serves
as a measure of the impulse amplitude.
Graphical representations:
19
Remark:
We use the short-hand notation:
dx(t)
dt
= x(t)
Relation between (t) and u(t)
First order derivative
(t) = u(t)
Running integral
u(t) =
t
() d
Formal diculty: u(t) is not dierentiable in the conventional sense
because of its discontinuity at t = 0.
Some more thoughts on (t)
Consider functions u
(t) and
(t) instead of u(t) and (t):

u
(t)
t
(t)
1
1
where
(t) = u
(t)
u
(t) =
t
() d
20
Limit 0
u(t) = lim
0
u
(t)
(t) :
t
1
(t)
3
(t)
2
(t)
3

2

1
1
1
1
3
1
2
Observe: Area under
(t) always 1
(t) is an innitesimally narrow impulse with area 1.
(t) = lim
0
(t)
() d = 1
Representation
a
t
a(t)
t
0
1
t
(t t
0
)
1
t
(t)
Real systems do not respond instantaneously. Most systems will respond nearly
the same to sharp pulses regardless of their shapeif
They have the same amplitude.
Their duration is much briefer than the systems response.
The unit impulse is an idealization of such pulses, which is short enough for any
system.
Properties of Unit Impulse
The relationship between (t) and u(t):
First order derivative: (t) =
du(t)
dt
Running integral: u(t) =
_
t
()d
Sampling properties:
x(t)(t t
0
) = x(t
0
)(t t
0
)
_

x(t)(t t
0
)dt =
_

x(t
0
)(t t
0
)dt = x(t
0
)
_

(t t
0
)dt = x(t
0
)
Time scaling:
(at) =
1
|a|
(t), (a = 0)
This is because
_

(at)dt =
_

1
|a|
()d =
1
|a|
.
Continuous-Time and Discrete-Time Systems
x(t)
-
CT
system
-
y(t) x[n]
-
DT
system
-
y[n]
System: A process in which input signals are transformed by the system or cause the
system to respond in some way, resulting in other signals as outputs.
All of the systems that we will consider have a single input and a single output
Continuous-time system transforms continuous-time signals.
Discrete-time system transforms discrete-time signals.
We will use the notation x(t) y(t) to mean the input signal x(t) causes the
output signal y(t).
Similar meaning is used for the notation x[n] y[n].
Simple examples of systems are given below.
Quadratic system:
x(t) y(t) = (x(t))
2
Delay system:
x[n] y[n] = x[n 1]
System represented by a rst order dierential equation:
dy(t)
dt
+ay(t) = bx(t)
where a and b are constants.
System described by a rst order dierence equation:
y[n] = ay[n 1] +bx[n]
where a and b are constants.
Interconnections of Systems
Many real systems are built as interconnections of several subsystems.
It is useful to use the understanding of the component systems and of how they
are interconnected to analyze the operation and behavior of the overall system.
Basic system interconnections:
23
1.3 ContinuousTime and DiscreteTime Systems
Unied representation of physical processes by systems
System: Entity that transforms input signals into new output signals
One or more input and output signals
Continuoustime system transforms continuoustime signals
Discretetime system transforms discretetime signals
Formal representation of inputoutput relation
Continuoustime system
x(t) y(t)
Discretetime system
x[n] y[n]
Remark: Another popular notation that you may nd in books is
y(t) = S{x(t)}, where S{} represents the system operator.
Pictorial representation of systems
Continuoustime
system
x(t) y(t)
system
Discretetime
x[n] y[n]
24
1.3.1 Simple Examples of Systems
Quadratic system
y(t) = (x(t))
2
System represented by a rst order dierential equation
y(t) + ay(t) = bx(t)
with constants a and b
Delay system
y[n] = x[n 1]
System described by a rst order dierence equation
y[n] = ay[n 1] + bx[n]
with constants a and b
1.3.2 Interconnections of Systems
Often convenient: break down a complex system into smaller subsystems
Series (cascade) interconnection
System 1 System 2 Input Output
Examples: Communication channel and receiver, detector and de-
coder in communications
25
Parallel interconnection
System 1
System 2
Output Input
Example: Diversity transmission: transmission of the same signal
over two antennas and receiving it with one antenna
Feedback interconnection
Output Input
System 2
System 1
Examples: Closed-loop frequency/phase/timing synchronization in
communications, human motion control
1.4 Basic System Properties
Simple mathematical formulation of basic (physical) system proper-
ties
Classication of systems
For conciseness: only denitions for continuous-time systems
Replacing (t) by [n] denitions for discrete-time systems
26
1.4.1 Linearity
Let x
1
(t) y
1
(t) and x
2
(t) y
2
(t)
Linear system if
1. Additivity
x
1
(t) + x
2
(t) y
1
(t) + y
2
(t)
2. Homogeneity
ax
1
(t) ay
1
(t) , a C
Linear systems possess property of superposition
Let x
k
(t) y
k
(t), then
K
k=1
a
k
x
k
(t) = x(t) y(t) =
K
k=1
a
k
y
k
(t)
Not linear systems are referred to as nonlinear.
Example:
1. System y(t) = tx(t) is linear.
To see this let
x
1
(t) y
1
(t) = tx
1
(t)
x
2
(t) y
2
(t) = tx
2
(t)
and
x
3
(t) = ax
1
(t) + bx
2
(t) ,
25
Parallel interconnection
System 1
System 2
Output Input
Example: Diversity transmission: transmission of the same signal
over two antennas and receiving it with one antenna
Feedback interconnection
Output Input
System 2
System 1
Examples: Closed-loop frequency/phase/timing synchronization in
communications, human motion control
1.4 Basic System Properties
Simple mathematical formulation of basic (physical) system proper-
ties
Classication of systems
For conciseness: only denitions for continuous-time systems
Replacing (t) by [n] denitions for discrete-time systems
26
1.4.1 Linearity
Let x
1
(t) y
1
(t) and x
2
(t) y
2
(t)
Linear system if
1. Additivity
x
1
(t) + x
2
(t) y
1
(t) + y
2
(t)
2. Homogeneity
ax
1
(t) ay
1
(t) , a C
Linear systems possess property of superposition
Let x
k
(t) y
k
(t), then
K
k=1
a
k
x
k
(t) = x(t) y(t) =
K
k=1
a
k
y
k
(t)
Example:
1. System y(t) = tx(t) is linear.
To see this let
x
1
(t) y
1
(t) = tx
1
(t)
x
2
(t) y
2
(t) = tx
2
(t)
and
x
3
(t) = ax
1
(t) + bx
2
(t) ,
Basic System Properties
The basic system properties of continuous-time and discrete-time systems have
important physical interpretations.
They also have relatively simple mathematical descriptions using the signals and
systems language.
1. Systems with and without memory: A system is said to be memoryless if and
only if the output y(t) at any time t
0
depends only on the input x(t) at the
same time, i.e., x(t
0
).
Memory indicates the system has the capability to store (remember)
information about input values at times other than the current time.
In many physical systems, memory is directly associated with the storage of
energy.
As examples, capacitors and inductors store energy and therefore create
systems with memory. In contrast, resistors have no such mechanism and
therefore create memoryless systems.
Other examples of memoryless systems:
(a) Limiter: y[n] =
_
_
x[n], A x[n] A
A, x[n] < A
A, x[n] > A
(b) Amplier: y(t) = Ax(t)
Other examples of systems with memory:
(a) Accumulator: y[n] =

n
k=
x[k] = x[n] +y[n 1]
(b) Delay: y(t) = x(t t
0
)
While the concept of memory in a system typically suggest storing past input
and output values, our formal denition of systems with memory also
includes the ones whose current output is dependent on the future values of
the input and output.
Such systems can be found in applications in which the independent variable
is not time, such as in image processing, in processing signals that have been
recorded previously (speech, geophysical, meteorological signals).
2. Invertibility: A system is invertible if distinct inputs lead to distinct outputs.
If the system is invertible, then an inverse system exists.
When the inverse system is cascaded with the original system, the output is the
same as the input:
x[n]
-
System
-
y[n] Inverse
system
-
x[n]
Examples of invertible systems:
(a) Amplier: y(t) = Ax(t), A = 0. The inverse system is w(t) =
1
A
y(t).
(b) Accumulator: y[n] = y[n 1] +x[n]. The inverse system is
w[n] = y[n] y[n 1] (which is the direntiator)
Examples of non-invertible systems:
(a) Limiter: y[n] =
_
_
x[n], A x[n] A
A, x[n] < A
A, x[n] > A
(b) Slicer: y[n] =
_
_
_
1, x[n] 0
1, else
3. Causal: A system is causal if the output y(t) at any time t
0
depends on values
of the input x(t) at only the present and past times, < t t
0
.
If two inputs to a causal system are identical up to some point in time, the
corresponding outputs must also be equal up to this same time:
If x
1
(t) = x
2
(t) for t t
0
then y
1
(t) = y
2
(t) for t t
0
, t
0
All analog circuits are causal.
All memoryless systems are causal, since the systems only respond to the
current value of the input.
Not all causal systems are memoryless (very few are).
Note that causality is not often an essential constraint in applications in which
the independent variable is not time, such as in image processing, in processing
data that have been recorded previously.
Examples:
(a) Accumulator is a causal system: y[n] =

n
k=
x[k] = x[n] +y[n 1]
(b) A smoothing averager is a noncausal system: y[n] =
1
2M + 1
M
k=M
x[n k]
4. Stability: A system is bounded-input bounded-output (BIBO) stable if all
bounded inputs (|x(t)| B
x
< , t) result in bounded outputs
(|y(t)| B
y
< , t).
Informally, stable systems are those in which small inputs do not lead to
outputs that diverge (grow without bound).
All physical circuits are technically stable.
Ideal op amp without negative feedback are usually unstable.
Examples:
(a) The smoothing averager is a stable system: y[n] =
1
2M + 1
M
k=M
x[n k].
For bounded input |x[n]| B
x
, the output |y[n]| B
y
= B
x
, which is also
bounded.
(b) The integrator is unstable system: y(t) =
_
t
x()d.
Let the bounded input be x(t) = u(t), then the output y(t) = t is
unbounded.
System stability is important in engineering applications. Unstable systems need
to be stabilized.
E
E
3
5
1
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31
1.4.6 Stability
Consider boundedinput boundedoutput (BIBO) stability
Stable system if for any bounded input signal
|x(t)| B
x
< , t
the output signal is bounded
|y(t)| B
y
< , t
Example:
Stable system
Averager: y[n] =
1
2N + 1
N
k=N
x[n k]
Bounded input |x[n]| < B
x
bounded output |y[n]| < B
y
=
B
x
Instable system
Integrator: y(t) =
t
x() d
E.g. bounded input x(t) = u(t) unbounded output y(t) = t
System stability is important in engineering applications, unstable
systems need to be stabilized.
32
Example: The rst Tacoma Narrows suspension bridge collapsed due
to wind-induced vibrations, November 1940.
(
P
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t
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p
:
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w
w
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c
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.
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.
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g
u
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e
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P
a
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4
6
5. Time Invariance: A system is time invariant if it produces identical response to
the same input signal no matter when input signal is applied. Mathematically:
x(t) y(t) implies x(t t
0
) y(t t
0
).
x[n] y[n] implies x[n n
0
] y[n n
0
].
In other words, a system is time invariant if a time shift in the input signal
results in a corresponding time shift in the output signal.
Circuits that have non-zero energy stored on capacitors or in inductors at time
t = 0 are generally not time-invariant (i.e., they are time-variant).
Memoryless does not imply time-invariant. For example, y(t) = x(t) f(t).
Examples:
(a) The system y(t) = (x(t))
2
is?
(b) The system y[n] = nx[n] is?
6. Linearity: Consider a system with x
1
(t) y
1
(t) and x
2
(t) y
2
(t). The
system is said to be linear if:
a
1
x
1
(t) +a
2
x
2
(t) a
1
y
1
(t) +a
2
y
2
(t)
for any constant complex coecients a
1
and a
2
.
x(t)
-
CT
system
-
y(t) x[n]
-
DT
system
-
y[n]
a
1
x
1
(t) +a
2
x
2
(t) a
1
y
1
(t) +a
2
y
2
(t)
a
1
x
1
[n] +a
2
x
2
[n] a
1
y
1
[n] +a
2
y
2
[n]
There are two related properties:
Additive: x
1
(t) +x
2
(t) y
1
(t) +y
2
(t)
Scaling: ax
1
(t) ay
1
(t)
Linear systems enable the application of superposition: If the input consists of a
linear combination of dierent inputs, the output is the same linear combination
of the corresponding outputs.
Linear Time-Invariant (LTI) Systems
A system is said to be linear time invariant (LTI) if it is both linear and time
invariant.
The linearity and time-invariance properties play a fundamental role in signal
and system analysis because of the following two main reasons:
Many physical processes posses these properties can be modeled as LTI
systems.
LTI systems can be analyzed in considerable detail, providing both insight
into their properties and a set of powerful tools for signal and system analysis.
Key idea: If one can represent the input to an LTI system in terms of a linear
combination of a set of basic signals, one can apply the superposition principle to
compute the output of the system in terms of its responses to these basic signals.
As will be seen shortly, the basic signals can be chosen to be the delayed
impulses an LTI system is completely characterized by its response to a unit
impulse, i.e., the impulse response.
Impulse Response of an LTI System
Impulse response is the systems response to a unit impulse.
The impulse response is denoted by h(t) or h[n]. Thus:
(t) h(t)
[n] h[n]
For any input x(t) (or x[n]), it is possible to use the impulse response h(t) (or
h[n]) to nd the output y(t) (or y[n]):
x(t)
-
h(t)
-
y(t) x[n]
-
h[n]
-
y[n]
This method is called convolution sum in the discrete-time case and convolution
integral in the continuous-time case.
Impulse response is an important concept (for example, it is used to implement
digital lters).
Discrete-Time Convolution Sum
Any discrete-time input signal x[n] can be expressed as a sum of scaled and
delayed unit impulses (sampling property of the unit-impulse):
x[n] =
k=
x[k][n k]
By linearity and time-invariance properties, the output of an LTI system is the
corresponding scaled sum of the outputs due to the delayed impulses:
x[k][n k] x[k]h[n k]
x[n] =
k=
x[k][n k] y[n] =
k=
x[k]h[n k] = x[n] h[n]
The above operation is called the discrete-time convolution sum.
Observe that the impulse response h[n] completely characterizes a discrete-time
LTI system: If we know h[n] then we can calculate the output y[n] for any
input x[n].
Example: The impulse response and the input of a discrete-time LTI system are
given below. Find the output.
h[n] = a
n
u[n]
_
_
_
a
n
, n 0
0, n < 0
(where a = 0.5)
x[n] =
_
_
1, n = 0
1, n = 2
2, n = 5
0, otherwise
0 5 10
1
0
1
2
x
[
n
]
0 5 10
1
0
1
2
h
[
n
]
In terms of unit-impulses, the input signal can be expressed as
x[n] = 1 [n] 1 [n 2] + 2 [n 5]
Since x
1
[n] = [n] y
1
[n] = h[n], x
2
[n] = [n 2] y
2
[n] = h[n 2] and
x
3
[n] = 2 [n 5] y
3
[n] = 2 h[n 5]. Therefore,
y[n] = y
1
[n] +y
2
[n] +y
3
[n] = h[n] h[n 2] + 2 h[n 5]
0 5 10
1
0
1
2
x
1
[
n
]
Input
0 5 10
1
0
1
2
y
1
[
n
]
Output
0 5 10
1
0
1
2
x
2
[
n
]
0 5 10
1
0
1
2
y
2
[
n
]
0 5 10
1
0
1
2
x
3
[
n
]
0 5 10
1
0
1
2
y
3
[
n
]
0 5 10
1
0
1
2
x
[
n
]
Time (n)
0 5 10
1
0
1
2
y
[
n
]
Time (n)
Technique to Find Discrete-Time Convolution Sum
x[n]
-
h[n]
-
y[n]
y[n] =
k=
x[k]h[n k]
Consider the evaluation of the output value at some specic time n, say n = n
0
:
y[n
0
] =
k=
x[k]h[n
0
k]
Plot the two signals (or two sequences) x[k] and h[n
0
k] as functions of k.
Multiplying these two functions to obtain a sequence g[k] = x[k]h[n
0
k].
Summing all the samples in the sequence g[k] yields the output value at the
selected time n
0
.
To plot h[n
0
k] as a function of k, it is convenient to follow the following two steps:
Plot the signal h[k] rst.
Obtain h[n
0
k] simply by shifting h[k] to the right (by n
0
) if n
0
is positive,
or to the left (by |n
0
|) if n
0
is negative.
Example: Compute y[6] for the previous example.
5 0 5 10
2
0
2
x
[
k
]
5 0 5 10
0
0.5
1
h
[
k
]
5 0 5 10
0
0.5
1
h
[
k
]
5 0 5 10
0
0.5
1
h
[
6
k
]
Time (k)
y[6] = h[6] h[4] + 2h[1] = 0.5
6
0.5
4
+ 2 0.5
1
= 0.9531 1.
Convolution Sum Derivation: Summary
-
h[n]
-
x[n] =
k=
x[k][n k] y[n] =
k=
x[k]h[n k]
Linearity
-
h[n]
-
x[k][n k] x[k]h[n k]
Linearity
-
h[n]
-
[n k] h[n k]
Time Invariance
-
h[n]
-
[n] h[n]
Denition of h[n]
-
h[n]
-
x[n] y[n]
LTI System
Input Output
Continuous-Time Convolution Integral
x(t)
-
h(t)
-
y(t)
Recall that if the input x(t) = (t), the output of the system is called the
impulse response, denoted by h(t).
The goal is to obtain a complete characterization of a continuous-time LTI
system in terms of its impulse response.
This means that, for any input x(t), we must be able to use the impulse
response h(t) to nd the output y(t). This method is called convolution integral.
To derive the convolution integral, we shall decompose and approximate the
input signal by rectangular pulses (or rectangles).
Derivation of the Convolution Integral
x(t)
0
( )
r
x t
( ) x kw
kw
( )
1
2
k w + ( )
1
2
k w
w
( ) ( ) ( ) ( )
1 1
2 2
( ) x kw u t k w u t k w

+

t
Approximate the input signal x(t) by a sum of weighted delayed rectangular pulses:
x(t) x
r
(t) =
k=
w x(kw)
_
u
_
t
_
k
1
2
_
w
_
u
_
t
_
k +
1
2
_
w
_
w
Since (t kw) =
du(t kw)
dt
= lim
w0
_
u
_
t
_
k
1
2
_
w
_
u
_
t
_
k +
1
2
_
w
_
w
So for very small w, x(t) can also be approximated as a sum of impulses:
x(t) x
w
(t) =
k=
w x(kw) (t kw)
By linearity and time-invariance properties, we have
y(t) y
w
(t) =
k=
w x(kw) h(t kw)
Finally, in the limit w 0 the above approximations become exact representations
and the summations become the integrals:
x(t) = lim
w0
x
w
(t) =
_

x()(t )d
y(t) = lim
w0
y
w
(t) =
_

x()h(t )d
The last equation above is the continuous-time convolution integral.
Example: Approximations of a triangle with rectangular pulses and impulses:
0 1 2
0
0.5
1
w=0.50
Approximation with Rectangles
0 1 2
0
0.5
Approximation with Impulses
0 1 2
0
0.5
1
w=0.25
0 1 2
0
0.1
0.2
0 1 2
0
0.5
1
w=0.10
0 1 2
0
0.05
0.1
0 1 2
0
0.5
1
w=0.05
Time (s)
0 1 2
0
0.05
Time (s)
Example:
R
x(t)
m
r
r
+
y(t)
C
The impulse response of the above RC circuit can be shown to be:
h(t) = RC e
t
RC
u(t)
= e
t
u(t) =
_
_
_
e
t
, t 0
0, t < 0
(for RC = 1 s)
Let the input signal to the circuit be the triangle considered in the previous page :
x(t) =
_
_
t, 0 t 1
(t 2), 1 t 2
0, otherwise
Approximations of this signal by impulses were also shown in the previous page.
The following gures show the responses of the circuit to dierent approximations of
the input signal by impulses.
5 0 5 10
0
0.2
0.4
0.6
0.8
Input
5 0 5 10
0
0.2
0.4
0.6
0.8
Output
True
Approximation
5 0 5 10
0
0.1
0.2
0.3
0.4
5 0 5 10
0
0.2
0.4
0.6
0.8
True
Approximation
5 0 5 10
0
0.05
0.1
Time (s)
5 0 5 10
0
0.2
0.4
0.6
0.8
Time (s)
True
Approximation
Convolution Integral: An Alternative Form
-
h(t)
-
x(t) y(t) =
_

x()h(t )d
Let u t , then = t u and du = d.
y(t) =
_

x(t u)h(u)(du)
=
_

x(t u)h(u)du
=
_

x(t )h()d
Both forms are called the convolution integral. It is often written as:
y(t) = x(t) h(t) =
_

x()h(t )d =
_

x(t )h()d
Convolution Integral: Solving Graphically
-
h(t)
-
x(t) y(t) = x(t) h(t) =
_

x()h(t )d
To calculate this integral graphically, perform the following steps for any specic
value of t:
1. First obtain the signal h(t ) (regarded as a function of with t xed)
from h() by reection about the origin and a shift to the right by t if t > 0
or a shift to the left by |t| for t < 0.
2. Next, multiply together the signals x() and h(t ) to obtain the function
g() = x()h(t ).
3. Finally, y(t) is obtained by integrating the function g() from = to
= .
Generally, it is sucient to plot both x() and h(t ) on the same axis.
Example: Let x(t) =
_
_
_
1, 0 t 2T
0, otherwise
and h(t) =
_
_
2, 0 t T
3, T t 2T
0, otherwise
.
Find y(t) = x(t) h(t) =
_

x()h(t )d.
41
Graphical interpretation
Time-reverse h() h()
Shift h() by t h(t )
Integrate products of overlapping components
Example:
Input: x(t) =
1, 0 t 2T
0, otherwise
Impulse response: h(t) =
2, 0 t < T
3, T t 2T
0 otherwise
t < 0
t 2T 2T T t
h(t )
x()
No overlap: y(t) = 0
0 t < T
t T 2T
y(t) =
t
0
1 2 dt = 2t
42
T t < 2T
t T 2T
y(t) =
tT
0
3 dt +
t
tT
2 dt = 3(t T) + 2T
2T t < 3T
t T 2T
y(t) =
2T
tT
2 dt +
tT
t2T
3 dt = 2(3T t) + 3T
3T t < 4T
t T 2T
y(t) =
2T
t2T
3 dt = 3(4T t)
41
Example:
Input: x(t) =
1, 0 t 2T
0, otherwise
2, 0 t < T
3, T t 2T
0 otherwise
t < 0
t 2T 2T T t
h(t )
x()
0 t < T
t T 2T
y(t) =
t
0
1 2 dt = 2t
42
T t < 2T
t T 2T
y(t) =
tT
0
3 dt +
t
tT
2 dt = 3(t T) + 2T
2T t < 3T
t T 2T
y(t) =
2T
tT
2 dt +
tT
t2T
3 dt = 2(3T t) + 3T
3T t < 4T
t T 2T
y(t) =
2T
t2T
3 dt = 3(4T t)
41
Example:
Input: x(t) =
1, 0 t 2T
0, otherwise
2, 0 t < T
3, T t 2T
0 otherwise
t < 0
t 2T 2T T t
h(t )
x()
0 t < T
t T 2T
y(t) =
t
0
1 2 dt = 2t
42
T t < 2T
t T 2T
y(t) =
tT
0
3 dt +
t
tT
2 dt = 3(t T) + 2T
2T t < 3T
t T 2T
y(t) =
2T
tT
2 dt +
tT
t2T
3 dt = 2(3T t) + 3T
3T t < 4T
t T 2T
y(t) =
2T
t2T
3 dt = 3(4T t)
41
Example:
Input: x(t) =
1, 0 t 2T
0, otherwise
2, 0 t < T
3, T t 2T
0 otherwise
t < 0
t 2T 2T T t
h(t )
x()
0 t < T
t T 2T
y(t) =
t
0
1 2 dt = 2t
42
T t < 2T
t T 2T
y(t) =
tT
0
3 dt +
t
tT
2 dt = 3(t T) + 2T
2T t < 3T
t T 2T
y(t) =
2T
tT
2 dt +
tT
t2T
3 dt = 2(3T t) + 3T
3T t < 4T
t T 2T
y(t) =
2T
t2T
3 dt = 3(4T t)
41
Example:
Input: x(t) =
1, 0 t 2T
0, otherwise
2, 0 t < T
3, T t 2T
0 otherwise
t < 0
t 2T 2T T t
h(t )
x()
0 t < T
t T 2T
y(t) =
t
0
1 2 dt = 2t
42
T t < 2T
t T 2T
y(t) =
tT
0
3 dt +
t
tT
2 dt = 3(t T) + 2T
2T t < 3T
t T 2T
y(t) =
2T
tT
2 dt +
tT
t2T
3 dt = 2(3T t) + 3T
3T t < 4T
t T 2T
y(t) =
2T
t2T
3 dt = 3(4T t)
43
t 4T
t T 2T
In summary
y(t) =
0, t < 0
2t, 0 t < T
2T + 3(t T), T t < 2T
2(3T t) + 3T, 2T t < 3T
3(4T t), 3T t < 4T
0, t 4T
5
4
3
2
1
y(t)/T
3 4 2 1
t/T
44
2.3 Properties of Linear TimeInvariant Systems
Linearity and time invariance complete characterization by im-
pulse response
y[k] =
k=
x[k]h[n k] = x[k] h[k]
y(t) =
x()h(t ) d = x(t) h(t)

Further properties based on and in terms of impulse response repre-
sentation
2.3.1 Commutative, Distributive, and Associative Property
of Convolution Interconnections of LTI Systems
Commutative property: order of the signals to be convolved can be
changed
Continuoustime case
x(t) h(t) = h(t) x(t) =
h()x(t ) d
Discretetime case
x[n] h[n] = h[n] x[n] =
k=
h[k]x[n k]
Combine all the cases, the nal expression for y(t) is:
43
t 4T
t T 2T
In summary
y(t) =
0, t < 0
2t, 0 t < T
2T + 3(t T), T t < 2T
2(3T t) + 3T, 2T t < 3T
3(4T t), 3T t < 4T
0, t 4T
5
4
3
2
1
y(t)/T
3 4 2 1
t/T
44
pulse response
y[k] =
k=
y(t) =
x()h(t ) d = x(t) h(t)

sentation
changed
Continuoustime case
x(t) h(t) = h(t) x(t) =
h()x(t ) d
Discretetime case
x[n] h[n] = h[n] x[n] =
k=
h[k]x[n k]
43
t 4T
t T 2T
In summary
y(t) =
0, t < 0
2t, 0 t < T
2T + 3(t T), T t < 2T
2(3T t) + 3T, 2T t < 3T
3(4T t), 3T t < 4T
0, t 4T
5
4
3
2
1
y(t)/T
3 4 2 1
t/T
44
pulse response
y[k] =
k=
y(t) =
x()h(t ) d = x(t) h(t)

sentation
changed
Continuoustime case
x(t) h(t) = h(t) x(t) =
h()x(t ) d
Discretetime case
x[n] h[n] = h[n] x[n] =
k=
h[k]x[n k]
Convolution Integral Derivation: Summary
-
h(t)
-
x(t) =
_

x()(t )d y(t) =
_

x()h(t )d
Linearity
-
h(t)
-
x()(t ) x()h(t )
Linearity
-
h(t)
-
(t ) h(t )
Time Invariance
-
h(t)
-
() h(t )
Denition of h(t)
-
h(t)
-
x(t) y(t)
LTI System
Input Output
Summary of Convolution Integral
-
h(t)
-
x(t) y(t) = x(t) h(t) =
_

x()h(t )d
The convolution integral describes how the output y(t) is related to the input
signal x(t) and the impulse response h(t).
Only two assumptions were made about the system:
Linear
Time Invariant
Key points:
The impulse response h(t) completely denes the behavior of a
continuous-time LTI system.
If h(t) is known, then the output of a continuous-time LTI system can be
found for any input x(t) using the convolution integral.
Properties of LTI Systems
1 Commutative property: The order of the signals to be convolved can be
interchanged:
y[n] = x[n] h[n] = h[n] x[n]
y(t) = x(t) h(t) = h(t) x(t)
Implications:
The output of an LTI system with input x(t) and impulse response h(t) is
identical to the output of an LTI system with input h(t) and impulse
response x(t).
Irrelevant whether h(t) or x(t) is reected and shifted to compute the
convolution integral.
2 Distributive property:
x[n] (h
1
[n] +h
2
[n]) = x[n] h
1
[n] +x[n] h
2
[n]
x(t) (h
1
(t) +h
2
(t)) = x(t) h
1
(t) +x(t) h
2
(t)
Implication: A parallel combination of LTI systems can be replaced by a single
LTI system whose unit impulse response is the sum of the individual unit impulse
responses in the parallel combination.
45
Implications:
output of system with impulse response h(t) to input x(t)
identical to output of system with impulse response x(t) to
input h(t)
Irrelevant whether h(t) or x(t) is reected and shifted for
convolution
Distributive property
Continuoustime case
x(t) (h
1
(t) + h
2
(t)) = x(t) h
1
(t) + x(t) h
2
(t)
Discretetime case
x[n] (h
1
[n] + h
2
[n]) = x[n] h
1
[n] + x[n] h
2
[n]
Implication
Parallel connection of two LTI systems represented by single
equivalent LTI system
h
2
(t)
h
1
(t)
h
1
(t) + h
2
(t)
y(t) x(t)
x(t) y(t)
46
Associative Property
Continuoustime case
x(t) (h
1
(t) h
2
(t)) = (x(t) h
1
(t)) h
2
(t)
Discretetime case
x[n] (h
1
[n] h
2
[n]) = (x[n] h
1
[n]) h
2
[n]
Consequently
y[n] = x[n] h
1
[n] h
2
[n]
Implications
Chose convenient order of convolution
Cascade connection of two LTI system represented by single
LTI system
h
1
(t) h
2
(t)
h
1
(t) h
2
(t) y(t) x(t)
x(t) y(t)
Associative + commutative property order in a cascade of LTI
systems irrelevant
y(t) = (x(t) h
1
(t)) h
2
(t) = (x(t) h
2
(t)) h
1
(t)
h
2
(t)
h
2
(t)
h
1
(t)
h
1
(t)
y(t) x(t)
x(t) y(t)
3 Associative property:
x[n] (h
1
[n] h
2
[n]) = (x[n] h
1
[n]) h
2
[n]
x(t) (h
1
(t) h
2
(t)) = (x(t) h
1
(t)) h
2
(t)
Implications:
The order of convolution is not important.
Cascade connection of two LTI system is represented by a single LTI system.
Using both the associative and commutative properties The order in a
cascade of LTI systems is irrelevant.
45
Implications:
input h(t)
convolution
Continuoustime case
x(t) (h
1
(t) + h
2
(t)) = x(t) h
1
(t) + x(t) h
2
(t)
Discretetime case
x[n] (h
1
[n] + h
2
[n]) = x[n] h
1
[n] + x[n] h
2
[n]
Implication
h
2
(t)
h
1
(t)
h
1
(t) + h
2
(t)
y(t) x(t)
x(t) y(t)
46
Continuoustime case
x(t) (h
1
(t) h
2
(t)) = (x(t) h
1
(t)) h
2
(t)
Discretetime case
x[n] (h
1
[n] h
2
[n]) = (x[n] h
1
[n]) h
2
[n]
Consequently
y[n] = x[n] h
1
[n] h
2
[n]
Implications
LTI system
h
1
(t) h
2
(t)
h
1
(t) h
2
(t) y(t) x(t)
x(t) y(t)
systems irrelevant
y(t) = (x(t) h
1
(t)) h
2
(t) = (x(t) h
2
(t)) h
1
(t)
h
2
(t)
h
2
(t)
h
1
(t)
h
1
(t)
y(t) x(t)
x(t) y(t)
45
Implications:
input h(t)
convolution
Continuoustime case
x(t) (h
1
(t) + h
2
(t)) = x(t) h
1
(t) + x(t) h
2
(t)
Discretetime case
x[n] (h
1
[n] + h
2
[n]) = x[n] h
1
[n] + x[n] h
2
[n]
Implication
h
2
(t)
h
1
(t)
h
1
(t) + h
2
(t)
y(t) x(t)
x(t) y(t)
46
Continuoustime case
x(t) (h
1
(t) h
2
(t)) = (x(t) h
1
(t)) h
2
(t)
Discretetime case
x[n] (h
1
[n] h
2
[n]) = (x[n] h
1
[n]) h
2
[n]
Consequently
y[n] = x[n] h
1
[n] h
2
[n]
Implications
LTI system
h
1
(t) h
2
(t)
h
1
(t) h
2
(t) y(t) x(t)
x(t) y(t)
systems irrelevant
y(t) = (x(t) h
1
(t)) h
2
(t) = (x(t) h
2
(t)) h
1
(t)
h
2
(t)
h
2
(t)
h
1
(t)
h
1
(t)
y(t) x(t)
x(t) y(t)
Proofs (for continuous-time case only):
Commutative property:
x(t) h(t) =
_

x()h(t )d
=t
=
_

x(t )h()d
=
_

x(t )h()d = h(t) x(t)

Distributive property:
x(t) (h
1
(t) +h
2
(t)) =
_

x()(h
1
(t ) +h
2
(t ))d
=
_

x()h
1
(t )d +
_

x()h
2
(t )d
= x(t) h
1
(t) +x(t) h
2
(t)
Associative property:
x(t) (h
1
(t) h
2
(t)) = x(t) (h
2
(t) h
1
(t))
= x(t)
_

h
2
()h
1
(t )d
=
_

x()
_

h
2
()h
1
(t )dd
=
_

x()h
1
(t )dh
2
()d
=
__

x()h
1
(t )d
_
h
2
(t)
= (x(t) h
1
(t)) h
2
(t)
LTI Systems Properties by the Impulse Response
LTI systems with and without memory
Recall that for memoryless systems, the output signal depends only on the
present value of the input signal.
For discrete-time LTI systems:
y[n] =
k=
x[k]h[n k]
?
= Kx[n], (K constant)
h[n] = K[n] = h[0][n]
For continuous-time LTI systems:
y(t) =
_

x()h(t )d
?
= Kx(t)
h(t) = K(t) = h(0)(t)
Identity systems (K = 1): x[n] = x[n] [n] and x(t) = x(t) (t).
Invertibility of LTI systems
An Invertible LTI system has LTI inverse
h[n] h
inv
[n] = [n]
For continuous-time LTI systems:
h(t) h
inv
(t) = (t)
Example: For an accumulator, h[n] = u[n], the output is:
y[n] =
k=
x[k]u[n k] =
n
k=
x[k]
Note that u[n] u[n 1] = h[n] h[n 1] = [n]
h
inv
[n] = [n] [n 1]
Causality of LTI systems
Recall that for causal systems, the output at anytime depends on only past and
present values of the input.
y[n] =
k=
x[k]h[n k] =
k=
h[k]x[n k]
?
=
k=0
h[k]x[n k]
h[n] = 0, for n < 0
Similarly, for continuous-time LTI system:
h(t) = 0, for t < 0
Stability of LTI systems
Recall that a system is stable if it produces bounded output for any bounded
input.
Bounded input |x[n]| B
x
, for all n
Output:
|y[n]| =
k=
h[k]x[n k]
k=
|h[k]||x[n k]| B
x
k=
|h[k]|
?
B
y
Sucient condition for BIBO stability is that h[n] is absolutely summable:
n=
|h[n]| <
It can be shown that the above condition is also necessary.
For continuous-time LTI systems, the sucient and necessary condition for
BIBO stability is that h(t) is absolutely integrable:
_

|h()|d
Example: For an integrator, h(t) = u(t), one has
_

|u()|d =
_

0
1d =
This system is therefore not stable.
LTI Systems Properties by the Impulse Response: Summary
Property Discrete-time LTI systems Continuous-time LTI systems
Memoryless h[n] = [n] h(t) = (t)
Invertibility h[n] h
inv
[n] = [n] h(t) h
inv
(t) = (t)
Causal h[n] = 0, for n < 0 h(t) = 0, for t < 0
Stability
n=
|h[n]| <
|h()|d <
Fourier Representation of Signals and Systems
The representation and analysis of LTI systems through the convolution
operation developed before are based on representing signals as linear
combinations of shifted impulses:
x[n] =
k=
x[k][n k]
x(t) =
_

x()(t )d
An alternative representation for signals and LTI systems is considered by using
complex exponentials as the basic signals. Such representations are known as
the continuous-time and discrete-time Fourier series and transform.
Fourier series and transform can be used to construct broad and useful
classes of signals.
They provide another convenient expression for the input-output relationship
of LTI systems.
They allow for insightful characterization and analysis of signals and LTI
systems.
A Brief History
1748 Euler observed that if a vibrating string could be written as a sum of normal
modes, this expression is also true for future times.
1753 Bernoulli argued that all physical positions of a string could be written as this
type of sum.
1759 Lagrange criticized this representation based on the belief that it could not
represent signals with corners, so it was of limited use.
1807 Fourier claimed any periodic signal could be represented as a sum of sinusoids.
Many of his ideas were developed by others. Lacroix, Monge and Laplace were
in favor, but Lagrange fervently opposed.
1822 Fourier nally published a book.
1829 Dirichlet provided precise conditions under which periodic signals could be
represented.
1965 Fast Forier Transform (FFT) published independently by Cooley & Tukey.
1984 Grossman & Morlet introduced wavelets as a specialized eld.
Jean Baptiste Joseph Fourier (1768-1830)
Engineers need historical perspective. For background material on J.B.J. Fourier see,
for example,
www-history.mcs.st-andrews.ac.uk/history/Mathematicians/Fourier.html
Fourier Series Representation of Periodic Signals
As mentioned before, it is advantageous in the study of LTI systems to represent
signals as linear combinations of basic signals that possess the following two
properties:
1. The set of basic signals can be used to construct a broad and useful class of
signals.
2. The response of an LTI system to each basic signal is simple enough in structure
so that the representation for the response of the system to any signal
constructed as a linear combination of the basic signals can be easily obtained.
Fourier representation and analysis are developed mainly from the fact that both of
the above properties are provided by the set of complex exponential signals (in
continuous and discrete time).
Recall that complex exponential signals are the signals of the form e
st
in continuous
time and z
n
in discrete time, where s and z are complex numbers.
Response of an LTI System to a Complex Exponential
The importance of complex exponential signals in the study of LTI systems
stems from the fact that the response of an LTI system to a complex exponential
input is the same complex exponential with only a change in amplitude:
Continuous-time: e
st
H(s)e
st
Discrete-time: z
n
H(z)z
n
where the complex amplitude factor H(s) or H(z) is, in general, a function of
the complex variable s or z. As will be seen shortly, the amplitude factor H(s)
and H(z) are directly related to the impulse responses h(t) and h[n],
respectively.
A signal for which the system output is a (possibly complex) constant times the
input is referred to as an eigenfunction of the system, and the amplitude factor
is referred to as the systems eigenvalue.
Thus, e
st
and H(s) are the eigenfunction and eigenvalue of a continuous-time
LTI system. Similarly, z
n
and H(z) are the eigenfunction and eigenvalue of a
discrete-time LTI system.
Derivation of the Continuous-Time Eigenvalue
For the input x(t) = e
st
, the output can be found through the convolution integral:
y(t) = x(t) h(t) =
_
+
x(t )h()d
=
_
+
e
s(t)
h()d = e
st
_
+
h()e
s
d
. .
H(s)
= H(s)e
st
where H(s)

=
_
+
h()e
s
d =
_
+
h(t)e
st
dt
is a complex constant whose value depends on the complex variable s and the
systems impulse response h(t).
Fourier analysis only involves the variable s that is purely imaginary, i.e., s = j.
This means that we consider only complex exponentials of the form e
jt
. With this
restriction, one has:
e
jt
__
+
h(t)e
jt
dt
_
e
jt
= H(j)e
jt
Derivation of the Discrete-Time Eigenvalue
Similarly, the response of a discrete-time LTI system to a complex exponential input
can be computed through the convolution sum:
y[n] = x[n] h[n] =
+
k=
x[n k]h[k]
=
+
k=
z
nk
h[k] = z
n
+
k=
h[k]z
k
. .
H(z)
= H(z)z
n
where H(z)

=
+
k=
h[k]z
k
=
+
n=
h[n]z
n
is a complex constant whose value
depends on the complex variable z and the systems impulse response h[n].
For the discrete-time case, Fourier analysis only involves variable z that has unit
magnitude, i.e., z = e
j
. Thus we focus on complex exponentials of the form e
jn
:
e
jn
_
+
n=
h[n]e
jn
_
e
jn
= H
_
e
j
_
e
jn
Response to A Linear Combination of Complex Exponentials
By linearity and time-invariance properties of LTI systems, one easily obtains:
x(t) =
k
a
k
e
s
k
t
y(t) =
k
a
k
_
H(s
k
)e
s
k
t
x[n] =
k
a
k
(z
k
)
n
y[n] =
k
a
k
[H(z
k
)(z
k
)
n
]
If the input to an LTI system can be expressed as a linear combination of
complex exponentials, then the output can also be represented as a linear
combination of the same complex exponential signals.
Note that each coecient in the representation of the output is obtained as the
product of the corresponding coecient a
k
of the input and the systems
eigenvalue H(s
k
) or H(z
k
) associated with the eigenfunction e
s
k
t
or z
n
k
,
respectively.
But what types of signals can be represented in this form?
Virtually all of the (periodic) signals that we are interested in!
This is important and interesting idea.
Fourier Series Representation of CT Periodic Signals
Recall that a continuous-time signal x(t) is periodic if there exists a positive
constant T (T > 0) such that:
x(t +T) = x(t) for all t
The fundamental period of x(t) is the minimum value of T for which the
above is satised. The fundamental period is often denoted as T
0
, while the
value
0
=
2
T
0
(rad/s) is referred to as the fundamental frequency.
Note that when the fundamental frequency and/or the fundamental period is
clear from the context, the subscript 0 in
0
and/or T
0
might be dropped to
simplify notation.
Introduced earlier are two basic periodic signals, namely the sinusoidal signal
x(t) = cos(
0
t) and the periodic complex exponential x(t) = e
j
0
t
. Both of
these signals are periodic with fundamental frequency
0
and fundamental
period T = 2/
0
.
The set of harmonically related complex exponentials is:
k
(t) = e
jk
0
t
= e
jk(2/T)t
, k = 0, 1, 2, ...
where each of these signals has a fundamental frequency that is a multiple of
0
.
The kth harmonic components
k
(t) and
k
(t) have a fundamental frequency
of |k|
0
and a fundamental period of T/|k|.
Observe that a linear combination of harmonically related exponentials is also
periodic with fundamental period T (i.e., fundamental frequency
0
= 2/T):
x(t) =
k=
a
k
e
jk
0
t
(1)
This is simply because each term in the above sum has one or more complete
cycles every T =
2
0
seconds.
The representation of a periodic signal as a linear combination of harmonically
related complex exponentials in the form of (1) is referred to as Fourier series
representation.
The question is how to nd the coecients a
k
.
Finding the CT Fourier Series Coecients
Assume that the periodic signal x(t) with fundamental period T
0
can be written as
x(t) =
+
k=
a
k
e
jkt
, where = 2/T
0
.
Then: x(t)e
jnt
=
+
k=
a
k
e
jkt
e
jnt
_
T
0
x(t)e
jnt
dt =
_
T
0
+
k=
a
k
e
j(kn)t
dt =
+
k=
a
k
_
T
0
e
j(kn)t
dt
=
+
k=
a
k
_
T
0
[cos((k n)t) +j sin((k n)t)]dt = T
0
a
n
Thus a
n
=
1
T
0
_
T
0
x(t)e
jnt
dt, or a
k
=
1
T
0
_
T
0
x(t)e
jkt
dt .
Remark: The notation
_
T
0
implies that the integration is performed over any interval
of length T
0
.
To summarize, if a periodic signal x(t) has a Fourier series representation:
x(t) = x(t +T
0
) =
+
k=
a
k
e
jkt
(synthesis equation)
then the coecients are given by
a
k
=
1
T
0
_
T
0
x(t)e
jkt
dt (analysis equation)
where = 2/T
0
.
The coecients a
k
are called the spectral coecients or Fourier series
coecients of x(t).
These complex coecients measure the portions of the signal x(t) at each
harmonic of the fundamental component.
The coecient a
0
is the dc (or constant) component of x(t), which is simply the
average value of x(t) over one period:
a
0
=
1
T
0
_
T
0
x(t)dt
Example: Consider the continuous-time periodic signal:
x(t) = 2 sin(2t 3) + sin(6t) (2)
It is clear that the fundamental frequency is
0
= 2. Using Eulers relation one has:
2 sin(2t 3) = 2 sin(
0
t 3) =
2
2j
_
e
j(
0
t3)
e
j(
0
t3)
_
=
e
3j
j
e
j
0
t
e
3j
j
e
j
0
t
sin(6t) = sin(3
0
t) =
1
2j
_
e
j3
0
t
e
j3
0
t
_
=
1
2j
e
j3
0
t
1
2j
e
j3
0
t
Hence,
x(t) =
1
2j
e
j3
0
t
e
3j
j
e
j
0
t
+
e
3j
j
e
j
0
t
+
1
2j
e
j3
0
t
(3)
From Equation (3), the nonzero Fourier series coecients a
k
are identied to be:
a
3
=
1
2j
=
j
2
=
1
2
e
j(
2
)
a
1
=
e
3j
j
= je
3j
= e
j(
2
+3)
= e
j(3
3
2
)
= e
1.7124j
a
1
=
e
3j
j
= je
3j
= e
j[(
2
+3)]
= e
j(
3
2
3)
= e
1.7124j
a
3
=
1
2j
=
j
2
=
1
2
e
j(
2
)
With a
k
= |a
k
|e
j
k
, then |a
k
| and
k
are the magnitude and phase of a
k
,
respectively. Also, as a convention, the phase angles are always converted to the
range [, ] by adding (or subtracting) with multiples of 2.
The magnitude and phase spectra of x(t) are sketched in Figure 3. Note that the
magnitude spectrum is even and the phase spectrum is odd. These are the common
properties for real-valued periodic signals (which will be discussed shortly).
5 0 5
0
0.5
1
Magnitude Spectrum
5 0 5
2
1
0
1
2
Phase Spectrum
Normalized Frequency (/
0
)
Figure 3: Magnitude spectrum of x(t).
Example: Consider the periodic square wave:
t
( ) x t
0 1
1
t
( ) x t
0
1

2
1 2 1 2
t
( ) x t
0
K

1
T
T
2T
1
T
T
2T
a
0
=
1
T
_
T/2
T/2
x(t)dt = K
2T
1
T
a
k
=
1
T
_
T/2
T/2
x(t)e
jk
0
t
dt =
1
T
_
T
1
T
1
Ke
jk
0
t
dt
=
K
jk
0
T
e
jk
0
t
T
1
T
1
=
2K
k
0
T
_
e
jk
0
T
1
e
jk
0
T
1
2j
_
= K
sin(k
0
T
1
)
k
= K
2T
1
T
sinc
_
k
2T
1
T
_
where
0
=
2
T
and sinc(x) =
sin(x)
x
.
20 15 10 5 0 5 10 15 20
0.5
0
0.5
T=4T
1
a
k
20 15 10 5 0 5 10 15 20
0.5
0
0.5
T=8T
1
a
k
20 15 10 5 0 5 10 15 20
0.2
0
0.2
T=16T
1
a
k
Normalized frequency (k=/
0
)
Figure 4: FS coecients of the square wave for dierent ratios T/T
1
: Combined
magnitude and phase spectrum.
20 15 10 5 0 5 10 15 20
0.2
0
0.2
T=16T
1
a
k
20 15 10 5 0 5 10 15 20
0
0.1
0.2
|
a
k
|
20 15 10 5 0 5 10 15 20
5
0
5
a
k
Normalized frequency (k=/
0
)
Figure 5: FS coecients of the square wave for T/T
1
= 16: Combined magni-
tude/phase spectrum and separate magnitude and phase spectra.
Properties of FS Coecients for Real-Valued Signals
If x(t) is a real-valued periodic signal and can be represented as a Fourier series:
x(t) =
+
k=
a
k
e
jkt
x
(t) =
+
k=
a
k
e
jkt
Let l = k, then one has
x
(t) =
+
l=
a
l
e
jlt
=
+
k=
a
k
e
jkt
= x(t) =
+
k=
a
k
e
jkt
Thus, by comparison of coecients, it follows that
a
k
= a
k
, a
k
= a
k
The above is known as the complex-conjugate symmetry of the Fourier series
coecients of real-valued periodic signals. The complex-conjugate symmetry implies
that:
The magnitude spectrum of the FS coecients is even: |a
k
| = |a
k
|.
The phase spectrum of the FS coecients is odd: a
k
= a
k
.
Alternative Forms for the FS of Real Periodic Signals
Since a
k
= a
k
for real periodic signals, one can write:
x(t) =
k=
a
k
e
jkt
= a
0
+
k=1
a
k
e
jkt
+
k=1
a
k
e
jkt
= a
0
+
k=1
_
(a
k
e
jkt
) + (a
k
e
jkt
)
= a
0
+ 2
k=1
Re{a
k
e
jkt
} = a
0
+ 2
k=1
Re{|a
k
|e
j
k
e
jkt
}
= a
0
+ 2
k=1
|a
k
|Re{e
j(kt+
k
)
}
= a
0
+ 2
k=1
A
k
cos(kt +
k
) (amplitude-phase form)
where A
k
= |a
k
| is the magnitude of a
k
and
k
= a
k
is the phase angle of a
k
.
The above representation is known as the amplitude-phase form of the Fourier series.
Alternatively, one can also write:
x(t) =
k=
a
k
e
jkt
= a
0
+ 2
k=1
Re{a
k
e
jkt
}
= a
0
+ 2
k=1
Re{a
k
cos(kt) +ja
k
sin(kt)}
= a
0
+ 2
k=1
(Re{a
k
} cos(kt) Im{a
k
} sin(kt))
= a
0
+ 2
k=1
[B
k
cos(kt) C
k
sin(kt)] (trigonometric form)
where B
k
= Re{a
k
} and C
k
= Im{a
k
}.
The above is referred to as the trigonometric form of Fourier series.
Alternative Forms of FS for Real Periodic Signals: Summary
Exponential: x(t) =
k=
a
k
e
jkt
Amplitude-Phase: x(t) = a
0
+ 2
k=1
A
k
cos(kt +
k
)
Trigonometric: x(t) = a
0
+ 2
k=1
[B
k
cos(kt) C
k
sin(kt)]
a
k
= A
k
e
j
k
= B
k
+jC
k
A
k
= |a
k
| =
_
B
2
k
+C
2
k

k
= a
k
= tan
1
_
C
k
B
k
_
B
k
= Re{a
k
} = A
k
cos(
k
) C
k
= Im{a
k
} = A
k
sin(
k
)
a
k
=
1
T
0
_
T
0
x(t)e
jkt
dt
B
k
=
1
T
0
_
T
0
x(t) cos(kt)dt, C
k
=
1
T
0
_
T
0
x(t) sin(kt)dt
Convergence & Discontinuities
It has been shown that if a periodic signal can be represented by a Fourier series
then one can compute the Fourier coecients.
It is not clear, however, if any periodic signal could be represented by a Fourier
series.
In general, the Fourier series representation x(t) of a periodic signal x(t) might
not always equal to x(t) for all t:
x(t) =
k=
a
k
e
jkt
, with a
k
=
1
T
0
_
T
0
x(t)e
jkt
dt
In fact, if x(t) is discontinuous, x(t) and x(t) are not equal for all t. Intuitively,
this should make sense because how can a linear combination of continuous
signals (sinusoids) represent a discontinuous signal?
The question is when a periodic signal x(t) does in fact have a Fourier series
representation, i.e., when the innite FS converges to the original signal x(t)?
Consider the dierence between the true signal and its FS representation:
e(t)

= x(t) x(t)
The Fourier series is said to converge if this error signal has zero energy over
one period:
E =
_
T
0
|e(t)|
2
dt = 0
Note that, just because the zero error energy (E = 0), it does not imply
x(t) = x(t) for all t. It, however, does imply that any dierences occur only at a
nite number of discrete (zero duration) points in time.
In general, if t
0
is a point of discontinuity, then:
x(t
0
) =
1
2
lim
0
[x(t
0
+ ) +x(t
0
)]
At all other points the two signals x(t) and x(t) are equal.
A sucient condition for convergence is that the signal has a nite energy over a
single period:
_
T
0
|x(t)|
2
dt <
The above is true of all signals you could generate in the lab. Thus all of the
periodic signals generated by a function generator have equivalent FS
representations.
If x(t) is a continuous signal, then it is safe to assume that x(t) = x(t). Note
that this is a stronger statement than merely stating that the FS converges.
Dirichlet Conditions for Convergence: An alternative sets of conditions developed by
Dirichlet guarantees that x(t) equals its FS representation, except at isolated values
of t for which x(t) is discontinuous. The Dirichlet conditions are as follows:
1. The signal x(t) must be absolutely integrable over any period:
_
T
0
|x(t)|dt < .
2. There must be a nite number of distinct maxima and minima during any single
period T
0
of the signal.
3. In any nite interval of time, there are only nite number of discontinuities.
Furthermore, each of these discontinuities is nite.
Figure 3.8 in textbook gives examples of signals that do not satisfy Dirichlet
condition(s).
Note that the above are sucient, but not necessary, conditions.
Gibbs Phenomenon
To gain some understanding of how the Fourier series converges for a periodic signal
with discontinuities, consider approximating a given periodic signal x(t) by the nite
Fourier series (with 2N + 1 terms):
x
N
(t) =
N
k=N
a
k
e
jkt
For discontinuous signals x(t), it is observed that:
There are ripples at the vicinity of the discontinuity.
Gibbs showed that the peak amplitude of these ripples does not decrease with
increasing N.
Specically, there is an overshoot of 9% of the height of the discontinuity, no
matter how large N becomes.
In fact, as N increases, the ripples in the nite Fourier series approximation
become compressed toward the discontinuity, but for any nite value of N, the
peak amplitude of the ripples remain constant.
The above behavior is known as the Gibbs phenomenon.
Example: Fourier series approximations of the periodic square wave.
2 1 0 1 2
0.5
0
0.5
1
1.5
Fourier Series Approximation (N=5)
t (sec)
2 1 0 1 2
0.5
0
0.5
1
1.5
t (sec)
2 1 0 1 2
0.5
0
0.5
1
1.5
t (sec)
2 1 0 1 2
0.5
0
0.5
1
1.5
t (sec)
Example: Consider the continuous-time periodic waveform shown below:
t
( ) x t
0 1
1
t
( ) x t
0
1
2
1 2 1 2
(a) Obviously, the fundamental period of x(t) is T
0
= 2 and the fundamental
frequency is
0
=
2
T
0
= .
(b) Next, we compute the trigonometric Fourier series coecients B
k
and C
k
. To
this end, consider x(t) in one period, from 0 t 2. Then
x(t) =
t
2
, 0 t 2
Hence,
B
k
=
1
T
0
_
T
0
x(t) cos(k
0
t)dt =
1
T
0
_
T
0
t cos(k
0
t)dt = 0
since t cos(k
0
t) is an odd function. Similarly
C
k
=
1
T
0
_
T
0
x(t) sin(k
0
t)dt =
1
4
_
2
0
t sin(
0
kt)dt
=
1
4
_
sin(
0
kt)
(
0
k)
2

t
0
k
cos(
0
kt)
_
2
0
=
1
4

2
k
=
1
2k
The DC component of x(t) is simply
a
0
=
1
T
0
_
T
0
x(t)dt =
1
2
_
2
0
t
2
dt =
t
2
8
2
0
=
1
2
(c) The magnitude and phase spectrum for the FS coecients of x(t) are plotted in
Figure 6. Observe that, since x(t) is a real function, it has an even amplitude
spectrum and an odd phase spectrum.
5 0 5
0
0.2
0.4
0.6
Magnitude Spectrum
5 0 5
2
1
0
1
2
Phase Spectrum
Normalized Frequency (/
0
)
Figure 6: Magnitude and phase spectra of x(t).
(d) The following partial Fourier series approximation
x(t) = a
0
+ 2
N
k=1
[B
k
cos(k
0
t) C
k
sin(k
0
t)] of x(t) is also plotted below for
N = 5, 10, 50. Note the Gibbs phenomenon at the points of discontinuity.
3 2 1 0 1 2 3
0
0.5
1
N=5
3 2 1 0 1 2 3
0
0.5
1
N=10
3 2 1 0 1 2 3
0
0.5
1
N=50
t (sec)
Properties of CT Fourier Series
The Fourier series representation possesses a number of useful properties. Many
properties are listed in Table 3.1 of textbook (p. 206). Though we only discuss a few
of these properties in class, you should be familiar with all of them.
1. Linearity: If x
1
(t) and x
2
(t) are periodic signals with period T and they have
Fourier series representations:
x
1
(t)
FS
a
k
, x
2
(t)
FS
b
k
then
y(t) =
1
x
1
(t) +
2
x
2
(t)
FS
1
a
k
+
2
b
k
2. Time Shifting: If x(t) is a periodic signal with fundamental period T that has
Fourier series coecients a
k
, then
y(t) = x(t t
0
)
FS
b
k
= e
jkt
0
a
k
= e
jk(2/T)t
0
a
k
Observe that time-shifting only changes the phase spectrum, not the magnitude
spectrum: |b
k
| = |a
k
|.
3. Time Reversal : If x(t) is a periodic signal with fundamental period T that has
Fourier series coecients a
k
, then
y(t) = x(t)
FS
b
k
= a
k
It follows from the above relationship that:
If x(t) is even, then b
k
= a
k
= a
k
, i.e., the FS coecients are even.
If x(t) is odd, then b
k
= a
k
= a
k
, i.e., the FS coecients are odd.
Furthermore, if x(t) is a real-valued function, then combining the above
time-reversal property and the complex-conjugate symmetry (a
k
= a
k
)
discussed before leads to the following conclusions:
If x(t) is real and even, then a
k
= a
k
= a
k
. This means the FS coecients
are real and even. Furthermore, the phases of a
k
can only be 0, or .
If x(t) is real and odd, then b
k
= a
k
= a
k
= a
k
. This implies that the FS
coecients are purely imaginary and odd. Furthermore, the phase of a
k
can
only be 0, /2 or /2.
4. Multiplication: If x
1
(t) and x
2
(t) are periodic signals with period T and they
have Fourier series representations:
x
1
(t)
FS
a
k
, x
2
(t)
FS
b
k
then
y(t) = x
1
(t)x
2
(t)
FS
c
k
=
l=
a
l
b
kl
= a
k
b
k
Observe that
l=
a
l
b
kl
is precisely the discrete-time convolution sum of two
sequences a
k
and b
k
.
5. Conjugation and Conjugate Symmetry: If x(t) is a periodic signal with
fundamental period T that has Fourier series coecients a
k
, then
y(t) = x
(t)
FS
b
k
= a
k
It follows from the above relationship that:
If x(t) is real, i.e., x(t) = x
(t), then b
k
= a
k
= a
k
the FS coecients
are conjugate symmetric.
If x(t) is purely imaginary, i.e., x(t) = x
(t) then b
k
= a
k
= a
k
.
6. Parsevals Relation: If the signal x(t) is a periodic signal with fundamental
period T that has Fourier coecients a
k
, then its average power can be
computed as follows:
1
T
_
T
|x(t)|
2
dt =
k=
|a
k
|
2
Observe that the average power of the kth harmonic component is
1
T
_
T
a
k
e
jkt
2
dt =
1
T
_
T
|a
k
|
2
dt = |a
k
|
2
Thus Parsevals relation simply states that the average total power of the signal
is the sum of the average powers in all of its harmonic components.
7. FS coecients of symmetric signals: Recall that any signal x(t) can be written
as a sum of even and odd signals: x(t) = x
e
(t) +x
o
(t), where
x
e
(t) =
x(t)+x(t)
2
and x
o
(t) =
x(t)x(t)
2
. Consider the trigonometric form of
the Fourier series of a real periodic signal x(t):
x(t) = a
0
+ 2
k=1
[B
k
cos(kt) C
k
sin(kt)]
The coecients B
k
and C
k
are determined by:
B
k
=
1
T
0
_
T
0
x(t) cos(kt)dt
=
1
T
0
_
+T
0
/2
T
0
/2
x
e
(t) cos(kt)dt +
1
T
0
_
+T
0
/2
T
0
/2
x
o
(t) cos(kt)dt
=
1
T
0
_
+T
0
/2
T
0
/2
x
e
(t) cos(kt)dt =
2
T
0
_
+T
0
/2
0
x
e
(t) cos(kt)dt
and
C
k
=
1
T
0
_
T
0
x(t) sin(kt)dt
=
1
T
0
_
+T
0
/2
T
0
/2
x
e
(t) sin(kt)dt
1
T
0
_
+T
0
/2
T
0
/2
x
o
(t) sin(kt)dt
=
1
T
0
_
+T
0
/2
T
0
/2
x
o
(t) sin(kt)dt =
2
T
0
_
+T
0
/2
0
x
o
(t) sin(kt)dt
It follows that:
If x(t) is even, then x
o
(t) = 0 and therefore C
k
= 0 and a
k
= B
k
(real).
If x(t) is odd, then x
e
(t) = 0 and hence B
k
= 0 and a
k
= jC
k
(imaginary).
Fourier Series Representation of Discrete-Time Periodic Signals
Recall that the set of all discrete-time complex exponential signals that are
periodic with period N is given by
k
[n] = e
jkn
= e
jk
2
N
n
, k = 0, 1, 2, ...
where e
jn
is called the fundamental component, e
j2n
is called the 2nd
harmonic component, and, in general, e
jkn
is called the kth harmonic
component.
Unlike continuous-time exponentials, there are only N distinct harmonics. This
is because:
k+lN
[n] = e
j(k+lN)
2
N
n
= e
jk
2
N
n+jl2n
= e
jk
2
N
n
+ e
jl2n
= e
jk
2
N
n
=
k
[n]
Wish to represent general DT periodic signals in terms of linear combinations of
harmonically related exponentials
k
[n]:
x[n] =
k
a
k
k
[n] =
k
a
k
e
jkn
=
k
a
k
e
jk
2
N
n
(4)
The above representation yields a signal x[n] that is periodic with fundamental
period N.
Since the signals (sequences)
k
[n] are distinct only over a range of N
successive values of k, the summation in (4) needs only include terms in this
range. We use the notation < N > to indicate any range of k that contains N
successive integers. Thus the representation in (4) is rewritten as:
x[n] =
k=<N>
a
k
k
[n] =
k=<N>
a
k
e
jkn
=
k=<N>
a
k
e
jk
2
N
n
(5)
Equation (5) is referred to as the discrete-time Fourier series and the
coecients a
k
as the Fourier series coecients.
Any nite-valued discrete-time periodic signal can be written exactly in this
form.
The task now is to nd the coecients a
k
.
In our derivation of the CT Fourier series coecients we used the following
relation:
_
T
e
jkt
dt =
_
T
e
jk(2/T)t
dt =
_
_
_
T k = 0,
0 k = 0
To solve for the DTFS coecients, we need a similar relation:
n=<N>
e
jk(2/N)n
=
_
_
_
N k = lN,
0 k = lN
for any integer l.
The above essentially states that the sum over one period of the values of a
periodic complex exponential is zero, unless that complex exponential is a
constant.
Now, since x[n] =
k=<N>
a
k
e
jkn
, one has
n=<N>
x[n]e
jln
=
n=<N>
_

k=<N>
a
k
e
jkn
_
e
jln
=
k=<N>
a
k
n=<N>
e
j(kl)n
= Na
l
Thus a
l
=
1
N
n=<N>
x[n]e
jln
, or a
k
=
1
N
n=<N>
x[n]e
jkn
.
To summarize, the Fourier series representation of a discrete-time periodic signal
x[n] with fundamental period N is given by:
x[n] =
k=<N>
a
k
e
jkn
=
k=<N>
a
k
e
jk(2/N)n
where a
k
=
1
N
n=<N>
x[n]e
jkn
=
1
N
n=<N>
x[n]e
jk(2/N)n
The above two equations are known as the discrete-time Fourier series pair.
The rst equation is called the synthesis equation, whereas the second
equation is called the analysis equation.
The coecients a
k
are called the spectral coecients or the Fourier series
coecients of x[n].
Since
k
[n] =
k+Nl
[n] (there are only N distinct discrete-time exponential
harmonics) and the DTFS sum is over any N consecutive terms, one has:
x[n] = a
0
0
[n] +a
1
1
[n] +. . . +a
N1
N1
[n]
= a
Nl
0
[n] +a
1+Nl
1
[n] +. . . +a
(N1)+Nl
N1
[n]
By comparing terms by terms, it can be concluded that a
k
= a
k+Nl
, i.e., the
values of a
k
repeat periodically with period N.
Example: Consider the following discrete-time periodic square wave with period N:
83
Short-hand notation
x[n]
FS
a
k
Denomination:
a
k
: Fourier series coecients or spectral coecients
Dierences to continuous-time case
Discrete-time Fourier series is nite
(Recall from Section 1.2.2: There are only N distinct discrete
time complex exponential signals
k
[n] = e
jk(2/N)n
that are
periodic with period N (harmonically related signals).)
No mathematical issues with convergence discretetime Fourier
series representation always exists
a
k
= a
k+N
since
k
[n] =
k+N
[n]
Remark:
The set of coecients
a
k
=
1
N
N1
n=0
x[n]e
jk(2/N)n
is commonly referred to as the N-point discrete Fourier transform
(DFT) of a nite duration signal x[n] with x[n] = 0 outside the
interval 0 n N. Due to the existence of an extremely fast
algorithm for the calculation of the DFT, called the fast Fourier
transform (FFT), the DFT (FFT) is of utmost importance in digital
signal processing.
84
Example:
Discretetime periodic square wave with period N
N
1
1
N N
1
n
x[n]
Fourier series coecients
a
k
=
1
N
n=N
x[n]e
jk(2/N)n
=
1
N
N
1
n=N
1
e
jk(2/N)n
=
1
N
sin(2k(N
1
+ 1/2)/N)
sin(k/N)
, k = 0, N, 2N, . . .
2N
1
+ 1
N
, k = 0, N, 2N, . . .
a
k
k
Remark: x[n] corresponds to a nite number of Fourier coe-
cients no Gibbs phenomenon, no convergence issues
Example: Partial Fourier series for a discrete-time periodic square wave:
15 10 5 0 5 10 15
1
0.5
0
0.5
1
1.5
t (sec)
15 10 5 0 5 10 15
1
0.5
0
0.5
1
1.5
t (sec)
15 10 5 0 5 10 15
1
0.5
0
0.5
1
1.5
t (sec)
15 10 5 0 5 10 15
1
0.5
0
0.5
1
1.5
t (sec)
Example: A discrete-time periodic signal x[n] is real-valued and has a fundamental
period N = 5 (i.e., the fundamental frequency is
0
=
2
N
). The nonzero Fourier
series coecients for x[n] are:
a
0
= 2, a
2
= a
2
= 2e
j/6
, a
4
= a
4
= 2e
j/3
Express x[n] in the form:
x[n] =
0
+
k=1
k
cos(
k
n +
k
)
Solution: We know that x[n] can be represented by a linear combination of the set
of harmonically-related complex exponentials:
x[n] =
<N>
a
k
e
jk
0
n
Consider one period of a
k
with 0 k 4, then
a
0
= 2
a
1
= a
1N
= a
4
= 2e
j/3
a
2
= 2e
j/6
a
3
= a
3N
= a
2
= 2e
j/6
a
4
= 2e
j/3
Hence,
x[n] =
<N>
a
k
e
jk
0
n
=
4
k=0
a
k
e
jk(2/5)n
= 2 +
_
a
1
e
j(2/5)n
+a
4
e
j(8/5)n
_
+
_
a
2
e
j(4/5)n
+a
3
e
j(6/5)n
_
= 2 +
_
a
1
e
j(2/5)n
+a
4
e
j(2/5)n
_
+
_
a
2
e
j(4/5)n
+a
3
e
j(4/5)n
_
= 2 + 2
_
e
j[(2/5)n/3]
+ e
j[(2/5)n/3]
_
+2
_
e
j[(4/5)n+/6]
+ e
j[(4/5)n+/6]
_
= 2 + 4 cos
_
2
5
n /3
_
+ 4 cos
_
4
5
n +/6
_
Compare and Contrast DTFS with CTFS
x(t) =
+
k=
a
k
e
jkt
, a
k
=
1
T
_
T
x(t)e
jkt
dt, (where =
2
T
)
x[n] =
k=<N>
a
k
e
jkn
, a
k
=
1
N
n=<N>
x[n]e
jkn
, (where =
2
N
)
Unlike the continuous-time Fourier series (CTFS):
There are only N terms in the sum of the discrete-time Fourier series (DTFS)
(since there are only N distinct DT exponential harmonics)
The nite DTFS sum can be obtained using any N consecutive Fourier series
coecients a
k
.
The DTFS coecients a
k
form a discrete-time periodic signal with the same
fundamental period N as that of x[n].
The DTFS always converges for all periodic signals such that |x[n]| <
There is no Gibbs phenomenon with DTFS.
DTFS Coecients and The Fast Fourier Transform (FFT)
a
k
=
1
N
n=<N>
x[n]e
jk
2
N
n
. .
DTFS coecients
, X(k) =
N1
n=0
x[n]e
jk
2
N
n
. .
discrete Fourier transform (DFT)
X(k) dened as above is commonly referred to as the N-point discrete Fourier
transform (DFT) of a nite duration signal x[n], where x[n] = 0 outside the
interval 0 n N 1.
Observe from the above two equations that the DFT of one period of x[n] is
proportional to the Fourier series coecients: X(k) = Na
k
The Fast Fourier Transform (FFT) is just a fast algorithm for the calculation of
DFT:
The direct approach to nd each of the N coecients a
k
would require N
2
computations.
The FFT enables this to be solved using N log
2
N computations.
FFT was developed by Tukey and Cooley in 1965.
Properties of Discrete-Time Fourier Series (DTFS)
Many properties of DTFS are summarized in Table 3.2 in the textbook (page
221). You should be familiar with all of them.
There are strong similarities between the properties of the discrete-time and
continuous-time Fourier series .
As examples, several properties are listed below.
Conjugate Symmetry for Real Signals: If x[n] is real, then a
k
= a
k
.
Multiplication: If x
1
[n] and x
2
[n] are periodic signals with period N and they
have Fourier series representations:
x
1
[n]
FS
a
k
, x
2
[n]
FS
b
k
then y[n] = x
1
[n]x
2
[n] is periodic with period N and
y[n] = x
1
[n]x
2
[n]
FS
c
k
=
l=<N>
a
l
b
kl
The sum
l=<N>
a
l
b
kl
is known as the periodic convolution sum between
the two periodic sequences a
k
and b
k
.
Parsevals Relation: If the signal x[n] is a periodic signal with fundamental
period N and it has FS representation x[n]
FS
a
k
, then its average power
can be computed as follows:
1
N
n=<N>
|x[n]|
2
=
k=<N>
|a
k
|
2
Observe that the average power of the kth harmonic component is
1
N
n=<N>
a
k
e
jk(2/N)n
2
= |a
k
|
2
Thus Parsevals relation simply states that the average total power of the
signal is the sum of the average powers in all of its harmonic components.
Fourier Series and LTI Systems
x(t)
-
h(t)
-
y(t) x[n]
-
h[n]
-
y[n]
Let x(t) and x[n] be continuous-time and discrete-time periodic signals at the
inputs of LTI systems with impulse responses h(t) and h[n], respectively. Then
the outputs of the LTI systems can be easily found as follows:
x(t) =
+
k=
a
k
e
jkt
y(t) =
+
k=
a
k
H(jk)e
jkt
x[n] =
k=<N>
a
k
e
jkn
y[n] =
k=<N>
a
k
H(e
jk
)e
jkn
where H(j) and H(e
j
) are called the frequency responses of continuous-time
and discrete-time LTI systems, respectively. They are dened as:
H(j) =
_

h(t)e
jt
dt, and H(e
j
) =
n=
h[n]e
jn
Observations:
x(t) =
+
k=
a
k
e
jkt
y(t) =
+
k=
a
k
H(jk)e
jkt
x[n] =
k=<N>
a
k
e
jkn
y[n] =
k=<N>
a
k
H(e
jk
)e
jkn
The outputs y(t) and y[n] are also periodic with the same fundamental
frequencies as x(t) and x[n], respectively.
If the set {a
k
} is the set of Fourier series coecients for the input x(t), then
{a
k
H(jk)} is the set of the Fourier series coecients for the output y(t):
x(t)
FS
a
k
y(t)
FS
a
k
H(jk)
Thus, the eect of the LTI system is to modify individually each of the
Fourier series coecients of the input through multiplication by the value of
the frequency response at the corresponding frequency.
For discrete-time case, the relationship between the Fourier series coecients
of the input and output of an LTI system is exactly the same:
x[n]
FS
a
k
y[n]
FS
a
k
H
_
e
jk(2/N)
_
Continuous-Time Fourier Transform: Overview
Fourier series enables us to represent periodic signals as linear combinations of
harmonically related complex exponentials. Such representation can be used to
clearly describe the eect of LTI systems on periodic signals.
This chapter extends such representation to continuous-time signals that are
not periodic (i.e., aperiodic signals).
We will see that, a rather large class of signals, including all energy signals, can
also be represented through a linear combination of complex exponentials.
The main dierence compared to FS representations of periodic signals is that
the complex exponentials in the representation of aperiodic signals are
innitesimally close in frequency The representation in terms of linear
combination takes the form of an integral rather than the sum.
In the FS representation of a periodic signal, as the period increases, the
fundamental frequency decreases and the harmonically related components
become closer in frequency.
Fourier reasoned that any aperiodic signal can be viewed as a periodic signal with
an innite period. As the period becomes innite, the frequency components
form a continuum and the FS sum becomes an integral (i.e., Fourier transform).
Derivation of Fourier Transform
Consider an aperiodic signal x(t) that is of nite duration: x(t) = 0 if |t| > T
1
.
Form a periodic signal x(t), with a period T, for which x(t) is one period.
As T (or
0
=
2
T
0), x(t) is equal to x(t) for any nite value of t.
x(t)
t
t
1
T
1
T
0
1
T
1
T T
T
T 2 T 2
( )
t j
j X

e
( )
t jk
jk X
0
e
0
( )
0 0
0
e Area
t jk
jk X =
0
k
0
) 1 ( + k
0
0
) (
~
t x
We know that x(t) can be represented with a Fourier series:

x(t) =
+
k=
a
k
e
jk
0
t
, a
k
=
1
T
_
T/2
T/2
x(t)e
jk
0
t
dt
The FS coecients of x(t) is:
a
k
=
1
T
_
T/2
T/2
x(t)e
jk
0
t
dt =
1
T
_
+
x(t)e
jk
0
t
dt =
1
T
X(jk
0
)
where X(j) =
_

x(t)e
jt
dt is the envelope of Ta
k
. Alternatively, Ta
k
can be
obtained as the sample X(j)|
=k
0
of the envelope function X(j). Now:
x(t) =
+
k=
1
T
X(jk
0
)e
jk
0
t
=
1
2
+
k=
X(jk
0
)e
jk
0
t
0
x(t) = lim
T
x(t) = lim
0
0
_
1
2
+
k=
X(jk
0
)e
jk
0
t
0
_
=
1
2
_
+
X(j)e
jt
d
x(t)
t
t
1
T
1
T
0
1
T
1
T T
T
T 2 T 2
( )
t j
j X

e
( )
t jk
jk X
0
e
0
( )
0 0
0
e Area
t jk
jk X =
0
k
0
) 1 ( + k
0
0
) (
~
t x

Fourier Transform: Summary
Fourier transform pair :
F{x(t)} = X(j)

=
_
+
x(t)e
jt
dt : Fourier transform
(Fourier integral, spectrum)
F
1
{X(j)} = x(t)

=
1
2
_
+
X(jw)e
jt
d : inverse Fourier transform
The synthesis equation represents an aperiodic signal as a linear combination of
complex exponentials.
For a periodic signal, the complex exponentials in its FS respresentation have
amplitudes a
k
occurring at a discrete set of harmonically related frequencies
k
0
, k = 0, 1, 2, . . .
For aperiodic signals, the complex exponentials occur at a continuum of
frequencies and have amplitude X(j)(d/2). Thus X(j) should be
interpreted as the amplitude density
X(j) provides information for describing x(t) as a linear combination of
sinusoidal signals at dierent frequencies.
Conditions for Convergence of Fourier Transforms
The Fourier transform of a signal x(t) exists if
_
+
|x(t)|
2
dt <
and any discontinuities are nite
The above is true for all signals of nite amplitude and duration.
Convergence does not imply that the inverse Fourier transform x(t) will be equal
to the original signal x(t) for all values of t.
However, convergence implies that, although x(t) and x(t) may dier
signicantly at individual values of t, there is no energy in their dierence (i.e.,
the energy of the dierence signal e(t) = x(t) x(t) is zero.
An alternative set of sucient conditions (Dirichlet conditions) for the
convergence of Fourier transforms is stated in the textbook (page 290).
Does a periodic signal have a Fourier transform? The answer is No, but we can
nd one if we allow X(j) to be expressed in terms of impulse functions The
Fourier series and Fourier transform can be incorporated into a common
framework (convenient).
Example: Transform of a Rectangular Pulse
x(t) =
_
_
_
1 |t| < T
1
0 otherwise
FT
X(j) =
_
T
1
T
1
e
jt
dt =
2 sin(T
1
)
= 2T
1
sinc
_
T
1
_
5 0 5
0
0.5
1
x
(
t
)
t (sec)
20 10 0 10 20
0.5
0
0.5
1
T
1
=0.5
X
(
j
)
(rad/sec)
5 0 5
0
0.5
1
x
(
t
)
t (sec)
20 10 0 10 20
1
0
1
2
T
1
=1
X
(
j
)
(rad/sec)
5 0 5
0
0.5
1
x
(
t
)
t (sec)
20 10 0 10 20
5
0
5
10
T
1
=5
X
(
j
)
(rad/sec)
If the signal stretches in time, its spectrum compresses in frequency.
Sinc Function
The sinc function arises frequently in Fourier analysis: sinc()

=
sin()
sinc(0) = 1, sinc(n) = 0 if n is a nonzero integer.
sinc(t)dt = 1
8 6 4 2 0 2 4 6 8
0.4
0.2
0
0.2
0.4
0.6
0.8
1
s
i
n
c
(

More Examples
Fourier transform of the unit impulse (t):
X(j) =
_

(t)e
jt
dt =
_

(t)e
j0
dt =
_

(t)dt = 1
The unit impulse has a FT consisting of equal contributions at all
frequencies.
Remark: In class, the above result was obtained by considering the FT of a
rectangular pulse over [T
1
, T
1
] and with amplitude
1
2T
1
, and then let T
1
0.
Fourier transform of a constant: Let x(t) = 1. The Fourier transform of x(t)
can be found as a limit of the FT of the rectangular pulse when T
1
.
Observe that the function 2T
1
sinc
_
T
1
_
approaches an impulse at = 0 as
T
1
. Specically, since
_

2T
1
sinc
_
T
1
_
d = 2, then
x(t) = 1
FT
X(j) = 2()
The above result is intuitively satisfying since x(t) only contains a DC
component at = 0.
Transform of a decaying exponential : Let x(t) = e
at
u(t) where a > 0. Then
the Fourier transform of x(t) is
X(j) =
_

0
e
at
e
jt
dt =
1
a +j
e
(a+j)t
0
=
1
a +j
, a > 0
The magnitude and phase spectra of x(t) are:
|X(j)| =
1
a
2
+
2
, X(j) = tan
1
_
a
_
25 20 15 10 5 0 5 10 15 20 25
0
0.2
0.4
0.6
0.8
1
(a=1)
M
a
g
n
i
t
u
d
e

o
f

X
(
j
)
(rad/sec)
25 20 15 10 5 0 5 10 15 20 25
2
1
0
1
2
(a=1)
P
h
a
s
e

o
f

X
(
j
)
(rad/sec)
Inverse Fourier Transform of a Rectangular Pulse
X(j) =
_
_
_
1 |w| < W
0 otherwise
FT
x(t) =
1
2
_
W
W
e
jt
d =
sin(Wt)
t
10 5 0 5 10
0
0.2
0.4
0.6
0.8
1
X
(
j
10 5 0 5 10
0.1
0
0.1
0.2
0.3
0.4
x
(
t
)
t (sec)
10 5 0 5 10
0
0.2
0.4
0.6
0.8
1
X
(
j
10 5 0 5 10
0.2
0
0.2
0.4
0.6
0.8
x
(
t
)
t (sec)
If the signals spectrum compresses in frequency, the signal stretches in time.
Fourier Transforms of Periodic Signals
Consider a signal x(t) with Fourier transform X(j) = 2(
0
). Then x(t)
can be found by performing the inverse Fourier transform:
x(t) =
1
2
_

2(
0
)e
jt
d = e
j
0
t
The above implies the following Fourier transform pair:
e
j
0
t
FT
2(
0
)
Now, if x(t) is a periodic signal, it has a Fourier series representation:
x(t) =
k=
a
k
e
jk
0
t
By the linearity of integral, it is straightforward to verify that
x(t) =
k=
a
k
e
jk
0
t
FT
X(j) =
k=
2a
k
( k
0
)
Properies of Fourier Transform
Linearity: If x
1
(t)
FT
X
1
(j) and x
2
(t)
FT
X
2
(j), then:
a
1
x
1
(t) +a
2
x
2
(t)
FT
a
1
X
1
(j) +a
2
X
2
(j)
This property follows directly from the linearity of integrals.
Time-Shifting:
x(t t
0
)
FT
e
jt
0
X(j)
Note that:
|X(j)e
jt
0
| = |X(j)|
X(j)e
jt
0
= X(j) t
0
Thus, a shift in time does not aect the magnitude of the Fourier transform.
The eect of a time shift is to introduce a phase shift t
0
, which is a linear
function of the frequency .
Conjugation and Conjugate Symmetry:
x
(t)
FT
X
(j)
If x(t) is real, then
x(t) = x
(t) X(j) = X
(j)
The magnitude spectrum is even: |X(j)| = |X(j)|
The phase spectrum is odd: X(j) = X(j)
Because of this symmetry
X(j) for a real signal is often only plotted for positive frequencies.
X(j) for w < 0 can be inferred from these plots.
If x(t) is both real and even, then X(j) is a real and even function of . This
implies that the value of phase spectrum X(j) can only be 0, or .
If x(t) is both real and odd, then X(j) is a purely imaginary and odd function
of . This implies that the value of phase spectrum X(j) can only be 0, /2
or /2.
Dierentiation:
dx(t)
dt
FT
jX(j)
d
n
x(t)
dt
n
FT
(j)
n
X(j)
This property can be easily derived by taking the derivative of both sides of the
synthesis equation. The key advantage is that ordinary dierential equations
become algebraic in the frequency domain.
Integration:
_
t
x()d
FT
1
j
X(j) +X(0)()
Dierentiation in time = Multiplication by j in frequency Integration in
time = Division by j in frequency. Also there is an impulse at = 0 to reect
the possible existence of a DC (average) value.
Example: To see the useful application of the dierentiation property of the Fourier
transform, consider nding the Fourier transform of the signal x(t) shown below.
t
1
1
0
x(t)
-1
-1
t
1
0
-1
-1
dx(t)/dt
) 1 ( t ) 1 ( + t
The rst approach: Applying FT directly
X(j) =
_

x(t)e
jt
dt =
_
1
1
(t)[cos(jt) j sin(jt)]dt
=
_
1
1
t cos(jt)dt
. .
=0 since t cos(jt) is an odd function
+j
_
1
1
t sin(jt)dt
= 2j
_
sin(t)
2

t cos(t)
1
0
= 2j
_
sin()
2

cos()
_
The second approach: Applying dierentiation property
dx(t)
dt
FT
2 sin()
+ e
j
+ e
j
=
2 sin()
+ 2 cos()
x(t)
FT
X(j) =
1
j
_
2 sin()
+ 2 cos()
_
= 2j
_
sin()
2

cos()
_
Remark: In general, impulses arise when a function to be dierentiated is
discontinuous. The impulse occurs at the point of the discontinuity with a strength
equal to the size of the jump.
t
( ) x t
0 T
A
t
( ) u t
0
1
T
t
( ) x t
0 T
A
T
A
t
0
1 2
t
0
1 2
1 2
) ( sign
2
1
t
) ( exp
2
1
at
1
exp( )
2
at
t
( ) x t
0
1
t
2
t
Upward-going impulse in the
derivative at t
1
of strength
equal to the size of the jump
Downward-going (i.e., negative
strength) impulse in the
derivative at t
2
of strength equal
to the negative size of the jump
Time & Frequency Scaling: x(at)
FT
1
|a|
X
_
j
a
_
This applies for both positive and negative values of a.
If the signal is stretched out in time (|a| < 1), the Fourier transform is
compressed (high frequencies are moved to lower frequencies).
If the signal is compressed in time (|a| > 1), the Fourier transform is expanded
(low frequencies are moved to higher frequencies).
This time-frequency relationship can be seen in many examples of Fourier
transform pairs considered before.
10 5 0 5 10
0
0.2
0.4
0.6
0.8
1
X
(
j
10 5 0 5 10
0.1
0
0.1
0.2
0.3
0.4
x
(
t
)
t (sec)
10 5 0 5 10
0
0.2
0.4
0.6
0.8
1
X
(
j
10 5 0 5 10
0.2
0
0.2
0.4
0.6
0.8
x
(
t
)
t (sec)
Parsevals Relation: The energy of a signal x(t) is dened as:
W
x
=
_
+
|x(t)|
2
dt
If x(t) represents the voltage across a 1 resistor, then W
x
is the energy dissipated
by the resistor.
Parsevals relation states that:
_
+
|x(t)|
2
dt =
1
2
_
+
|X(j)|
2
d
The important of this theorem is that it tells us to think of |X(j)|
2
as the energy
spectral density (i.e., how the signals energy is distributed over frequency).
A more general version of Parsevals relation is
_
+
x(t)y
(t)dt =
1
2
_
+
X(j)Y
(j)d
where x(t) and y(t) are two arbitrary continuous-time signals.
Example: To illustrate the application of Parsevals relation, consider the following
problem: An engineer wishes to send a signal sinc
_
Wt
_
across a communications
channel, where W is a positive constant. The channel can be modeled as an ideal
low-pass lter with a transfer function
H(j) =
_
_
_
1, 2 25 10
3
rad/sec
0, otherwise
What range of values of W can the engineer guarantee that at least 95% of the
signal energy will reach the receiver?
Solution:
x(t) = sinc
_
Wt
_
=
sin(Wt)
Wt
=

W
sin(Wt)
t
Since
a(t) =
sin(Wt)
t
F
A(j) =
_
_
_
1, W
0, otherwise
Then
x(t) =

W
sin(Wt)
t
F
X(j) =
_
_
_
W
, W
0, otherwise
The energy of x(t) is
E
x
=
1
2
_
W
W
|X(j)|
2
d =
1
2
_
W
W
_

W
_
2
d =

W
The Fourier transform of the output signal y(t) is Y (j) = X(j)H(j). The
channel can be modeled as an ideal low-pass lter with a transfer function
H(j) =
_
_
_
1,
c
0, otherwise
where
c
= 2 25 10
3
rad/sec is the cuto frequency of the channel. Obviously
the eect of the channel on the input signal depends on the value of W:
If W
c
then Y (j) = X(j) and therefore y(t) = x(t): The channel passes
the signal x(t) without any distortion and of course E
y
= E
x
.
The more interesting situation is when W >
c
. In this case the transmitted
signal x(t) will be ltered by the channel and not all the energy of the
transmitted signal can be received at the receiver. Since
Y (j) = X(j)H(j) =
_
_
_
W
,
c
0, otherwise
the energy of the output signal y(t) is
E
y
=
1
2
_

c
c
|Y (j)|
2
d =
1
2
_

c
c
_

W
_
2
d =

c
W
2
Thus if one requires E
y
= 0.95E
x
, then

c
W
2
= 0.95

W
. It then follows that
W =

c
0.95
= 52.6 10
3
rad/sec
To conclude: 0 < W 52.6 10
3
rad/sec.
The Convolution Property
-
h(t)
-
x(t) y(t) = x(t) h(t)
Recall the response of an LTI system to a periodic input signal is:
+
k=
a
k
e
jk
0
t
k=
a
k
H(jk
0
)e
jk
0
t
where H(j) is the frequency response of the LTI system. It was dened as
H(j) =
_

h(t)e
jt
dt
Thus the systems frequency response is precisely the Fourier transform of the
systems impulse response!
The above says that the FS coecients of the output are those of the input
multiplied by the frequency response of the system evaluated at the
corresponding harmonic frequencies.
For aperiodic input signals, what is the eect of the LTI system in
frequency-domain (i.e., how does it change the spectrum of the input)?
By the linearity property of LTI systems:
x(t) =
1
2
_
+
X(j)e
jt
d y(t) =
1
2
_
+
X(j)[H(j)e
jt
]d
since y(t) =
1
2
_
+
Y (j)e
jt
d
then Y (j) = X(j)H(j)
To summarize: x(t) h(t)
FT
X(j) H(j)
The Fourier transform maps the convolution of two signals into the product of
their Fourier transforms (a complicated convolution in the time-domain is
equivalent to a simple multiplication in the frequency-domain).
For an LTI system, the frequency response captures the change in complex
amplitude of the Fourier transform of the input at each frequency .
Since h(t) completely characterizes an LTI system, then so must H(j).
H(j) is also known as the transfer function.
|H(j)| is the magnitude-frequency response: |Y (j)| = |X(j)||H(j)|
H(j) is the phase-frequency response: Y (j) = X(j) +H(j)
A formal proof of the convolution property:
Suppose y(t) = x(t) h(t). One can solve for Y (j) in terms of X(j) and H(j)
as follows:
Y (j) =
_
+
__
+
x()h(t )d
_
e
jt
dt
=
_
+
x()
__
+
h(t )e
jt
dt
_
d
=
_
+
x()
__
+
h(u)e
j(u+)
du
_
d
=
_
+
x()
__
+
h(u)e
ju
du
_
e
j
d
= H(j)
_
+
x()e
j
d
= H(j) X(j)
= X(j) H(j)
Example: This example illustrates the use of convolution property. It also shows the
importance of the phase-frequency response of an LTI system.
Consider the continuous-time LTI system with frequency response
H(j) =
a j
a +j
, a > 0
(a) The magnitude and phase responses of the system are:
|H(j)| =
a
2
+
2
a
2
+
2
= 1
H(j) = (tan
1

a
) (tan
1

a
) = 2 tan
1

a
10 5 0 5 10
0
0.5
1
|H(j)|
(rad/sec)
10 5 0 5 10
4
2
0
2
4
H(j)
(rad/sec)
(b) Using the table of Fourier transform on page 329 of textbook, the impulse
response of the system is found to be
H(j) = 1 +
2a
a +j
FT
h(t) = (t) + 2ae
at
u(t)
(c) Let a = 1. We wish to nd the output of the system when the input is
x(t) = cos(t/
3) + cos(t) + cos(
3t)
Since a = 1, then |H(j)| = 1 and H(j) = 2 tan
1
. First, it is
convenient to nd the output of an LTI system to the sinusoidal input
cos(
0
t +). To this end, recall that e
j(
0
t+)
is an eigenfunction of the LTI
systems:
e
j(
0
t+)
H(j
0
)e
j(
0
t+)
= |H(j
0
)|e
j[(
0
t+)+H(j
0
)]
e
j(
0
t+)
H(j
0
)e
j(
0
t+)
= |H(j
0
)|e
j[(
0
t+)H(j
0
)]
where we have used the fact that
H(j
0
) = |H(j
0
)|e
jH(j
0
)
= |H(j
0
)|e
jH(j
0
)
. Thus
cos(
0
t +) |H(j
0
)| cos(
0
t + +H(jw
0
))
Therefore,
y(t) = cos
_
t
3

3
_
+ cos
_
t

2
_
+ cos
_
3t
2
3
_
Plots of the input and output are shown below. Observe that since the systems
phase-frequency response is not a linear function of , the output is not simply a
shifted version of the input. This example clearly shows that one cannot ignore
the phase-frequency response of an LTI system.
20 15 10 5 0 5 10 15 20
4
2
0
2
4
Input
t (sec)
20 15 10 5 0 5 10 15 20
4
2
0
2
4
Output
t (sec)
Filtering
In may applications, it is of interest to change the relative amplitudes of the
frequency components in a signal, or to eliminate some frequency components
entirely. Such processes are referred to as ltering.
LTI systems that change the shape of the input spectrum are often referred to as
frequency-shaping lters.
LTI systems that are designed to pass some frequencies essentially undistorted
and signicantly attenuate or eliminate other frequencies are referred to as
frequency-selective lters.
From the convolution property of the Fourier transform, it is evident that
ltering can be conveniently accomplished through the use of LTI systems with
an appropriate chosen frequency response H(j).
Frequency-domain method thus provides ideal tools to examine this very
important class of applications.
Ideal Frequency-Selective Filters
0
1
) ( j H
c
Lowpass
0
1
) ( j H
c
Highpass
0
1
) ( j H
c
Notch
0
1
) ( j H
Bandpass
0
1
) ( j H
Bandstop
1 c
2 c
1 c
2 c
The frequency components within the passband are passed without modication,
whereas the freq. components that fall into the stopband are completely eliminated.
Lowpass lters pass low frequencies ( <
c
).
Highpass lters pass high frequencies ( >
c
).
Bandpass lters pass a range of frequencies (
c1
< <
c2
).
Bandstop lters pass two ranges of frequencies ( <
c1
and >
c2
).
Notch lters pass all frequencies except
c
.
Example: Impulse Response of the Ideal Lowpass Filter
H(j) =
_
_
_
1, |w| < W
0, otherwise
FT
h(t) =
1
2
_
W
W
e
jt
d =
sin(Wt)
t
10 8 6 4 2 0 2 4 6 8 10
0
0.2
0.4
0.6
0.8
1
H
(
j
10 8 6 4 2 0 2 4 6 8 10
0.2
0
0.2
0.4
0.6
0.8
h
(
t
)
t (sec)
The above ideal LPF cannot be implemented in practice since it is
non-causal. Moreover, the oscillatory behavior in the lters impulse response
is highly undesirable.
An Example of Frequency-Shaping Filter: A Practical LPF
( )
i
v t
R
( ) i t
C ( )
o
v t
+ +

v
i
(t) = Ri(t) +
1
C
_
t
i()d, v
o
(t) =
1
C
_
t
i()d
Taking the Fourier transform of both sides of the above equations (and ignoring any DC
signals in the circuit) gives
V
i
(j) = RI(j) +
1
C
I(j)
j
, V
o
(j) =
1
C
I(j)
j
H(j) =
Y (j)
X(j)
=
1
1 +jRC
=
1
1 +j/
c
where
c
=
1
RC
is called the cuto frequency.
The impulse response of the lter is
h(t) =
c
e
c
t
u(t) =
1
RC
e
t
RC
u(t) (Note that the lter is causal)
The magnitude-frequency and phase-frequency spectra are
|H(j)| =
1
_
1 + (/
c
)
2
, H(j) = tan
1
(/
c
)
10 5 0 5 10
0
0.5
1
Magnitude Frequencyresponse
/
c
10 5 0 5 10
2
1
0
1
2
Phase frequencyresponse
/
c
The RC lowpass lter is a relatively crude approximation of the ideal lowpass lter. Better
approximations can be obtained with more-complicated circuits.
More Properties of Fourier Transform
Multiplication:
x(t) w(t)
FT
1
2
X(j) W(j)
Frequency-Shifting:
e
j
0
t
x(t)
FT
X(j(
0
))
Multiplication by a complex exponential shifts the Fourier transform to the
specied frequency.
This is the basis of amplitude modulation (AM) technique used in
communications.
This is also a convenient method for multiplexing multiple bandlimited signals
into a single channel.
The signal can then be recovered by bandpass ltering and AM demodulation
Example: To illustrate the application of the frequency-shifting property, consider
nding the Fourier transform of a windowed cosine. Specically, consider n cycles of
a cosine of frequency
0
(rad/sec) that lasts only over the time interval T/2 to
T/2 seconds, i.e., T =
2
0
n. A typical plot of it is shown below.
15 10 5 0 5 10 15
1
0.8
0.6
0.4
0.2
0
0.2
0.4
0.6
0.8
1
t (sec)
Such a signal can be written as x(t) = Acos(
0
t)w(t), where the window function
w(t) = u(t +T/2) u(t T/2).
The Fourier transform of Aw(t) is AW(j) = AT
sin(T/2)
(T/2)
. Using the
frequency-shifting property of the Fourier transform one has
x(t) cos(
0
t) =
1
2
_
x(t)e
j
0
t
+x(t)e
j
0
t
FT
1
2
X(j( +
0
)) +
1
2
X(j(
0
))
The above states that multiplying a function x(t) by a cos(
0
t) shifts its spectrum
up and down by the frequency
0
and scales the amplitude by 2. Thus the result is
G(j
n
) = n
A
0
_
sin(n(
n
1))
n(
n
1)
+
sin(n(
n
+ 1))
n(
n
+ 1)
_
where
n
= /
0
is the normalized frequency. A further normalization by (A/
0
)
yields:
G
norm
(j
n
) =
G(j)
(A/
0
)
= n
_
sin(n(
n
1))
n(
n
1)
+
sin(n(
n
+ 1))
n(
n
+ 1)
_
Plots of G
norm
(j
n
) for dierent values of n are shown below. Note that as more
and more cycles are taken the Fourier transform (i.e., the spectrum density) becomes
more and more concentrated around
0
(rad/sec). In the limit as n or
equivalently T the density becomes 2 impulses located at
0
(rad/sec).
40 20 0 20 40
0.5
0
0.5
1
1.5
n
A
m
p
l
i
t
u
d
e

D
e
n
s
i
t
y

S
p
e
c
t
r
u
m
n=1
40 20 0 20 40
2
0
2
4
6
n
A
m
p
l
i
t
u
d
e

D
e
n
s
i
t
y

S
p
e
c
t
r
u
m
n=5
40 20 0 20 40
5
0
5
10
n
A
m
p
l
i
t
u
d
e

D
e
n
s
i
t
y

S
p
e
c
t
r
u
m
n=10
40 20 0 20 40
20
10
0
10
20
30
40
n
A
m
p
l
i
t
u
d
e

D
e
n
s
i
t
y

S
p
e
c
t
r
u
m
n=50
Example: Figure (a) below shows a system with input signal x(t) and output signal
y(t). The input signal has Fourier transform X(j) shown in Figure (b). Determine
and sketch the spectrum Y (j) of y(t).
-3W 3W 5W -5W
1
x(t)
cos(5Wt) cos(3Wt)
3W -3W
1
y(t)
(a)
-2W
-2W
1
(b)
) ( j X
After modulated with carrier frequency 5W and passing through band-pass lter, the
output signal is shown in Figure (c) (shaded area). That signal is modulated one
more time with carrier frequency 3W. The modulated signal is plotted in Figure (d).
Therefore, Y (j), the output of low-pass lter with cut-o frequency 3W can be
seen in Figure (e).
5W
1/2
-3W 3W 5W -5W
1
x(t)
cos(5Wt) cos(3Wt)
3W -3W
1
y(t)
(a)
7W 3W
-5W -3W -7W
6W -6W 8W -8W 2W -2W

0
0
2W -2W 0
) ( jw Y
1/4
1/4
(c)
(d)
(e)
-2W 2W
1
(b)
) ( j X
0
After the first modulation and
passing via band-pass filter
The second modulation
Output of the low-pass filter

EE351: Spectrum Analysis and Discrete Time Systems (Signals, Systems and Transforms)

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EE351: Spectrum Analysis and Discrete Time Systems

(Signals, Systems and Transforms)

< , are known as energy signals.

> 0, are known as power signals.

= ) and innite energy (E

(t) instead of u(t) and (t):

x()h(t ) d = x(t) h(t)

x()h(t ) d = x(t) h(t)

x()h(t ) d = x(t) h(t)

x(t )h()d = h(t) x(t)

We know that x(t) can be represented with a Fourier series:

Dr. H. Nguyen Page 131

sinc(0) = 1, sinc(n) = 0 if n is a nonzero integer.

Dr. H. Nguyen Page 135

6W -6W 8W -8W 2W -2W

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