Bouquin - Signals-And-Systems-10.10

Signals and Systems
Collection Editor:
Marco F. Duarte
Signals and Systems
Collection Editor:
Marco F. Duarte
Authors:
Thanos Antoulas
Richard Baraniuk
Dan Calderon
Marco F. Duarte
Catherine Elder
Natesh Ganesh
Michael Haag
Don Johnson
Stephen Kruzick
Matthew Moravec
Justin Romberg
Louis Scharf
Melissa Selik
JP Slavinsky
Dante Soares
Online:
< http://legacy.cnx.org/content/col11557/1.10/ >
OpenStax-CNX
This selection and arrangement of content as a collection is copyrighted by Marco F. Duarte. It is licensed under the
Creative Commons Attribution License 4.0 (http://creativecommons.org/licenses/by/4.0/).
Collection structure revised: September 13, 2014
PDF generated: December 6, 2014
For copyright and attribution information for the modules contained in this collection, see p. 198.
Table of Contents
1 Review of Prerequisites: Complex Numbers
1.1 Geometry of Complex Numbers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1
1.2 Complex Numbers: Algebra of Complex Numbers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5
1.3 Representing Complex Numbers in a Vector Space . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 11
2 Continuous-Time Signals
2.1 Signal Classications and Properties . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 17
2.2 Common Continuous Time Signals . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 22
2.3 Signal Operations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 25
2.4 Energy and Power of Continuous-Time Signals . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 28
2.5 Continuous Time Impulse Function . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 31
2.6 Continuous-Time Complex Exponential . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33
3 Introduction to Systems
3.1 Introduction to Systems . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 37
3.2 System Classications and Properties . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 39
3.3 Linear Time Invariant Systems . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 43
4 Time Domain Analysis of Continuous Time Systems
4.1 Continuous Time Systems . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 51
4.2 Continuous Time Impulse Response . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 52
4.3 Continuous-Time Convolution . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 55
4.4 Properties of Continuous Time Convolution . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 58
4.5 Causality and Stability of Continuous-Time Linear Time-Invariant Systems . . . . . . . . . . . . . . . . . 62
5 Introduction to Fourier Analysis
5.1 Introduction to Fourier Analysis . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 65
5.2 Continuous Time Periodic Signals . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 66
5.3 Eigenfunctions of Continuous-Time LTI Systems . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 68
5.4 Continuous Time Fourier Series (CTFS) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 69
6 Continuous Time Fourier Transform (CTFT)
6.1 Continuous Time Aperiodic Signals . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 75
6.2 Continuous Time Fourier Transform (CTFT) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 76
6.3 Properties of the CTFT . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 79
6.4 Common Fourier Transforms . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 83
6.5 Continuous Time Convolution and the CTFT . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 85
6.6 Frequency-Domain Analysis of Linear Time-Invariant Systems . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 87
Solutions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 89
7 Discrete-Time Signals
7.1 Common Discrete Time Signals . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 91
7.2 Energy and Power of Discrete-Time Signals . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 93
7.3 Discrete-Time Signal Operations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 95
7.4 Discrete Time Impulse Function . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 98
7.5 Discrete Time Complex Exponential . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 100
8 Time Domain Analysis of Discrete Time Systems
8.1 Discrete Time Systems . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 105
8.2 Discrete Time Impulse Response . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .. . . . . . . . . . . . 107
8.3 Discrete-Time Convolution . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 110
iv
8.4 Properties of Discrete Time Convolution . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 115

8.5 Causality and Stability of Discrete-Time Linear Time-Invariant Systems . . . . . . . . . . . . . . . . . . . 118
9 Discrete Time Fourier Transform (DTFT)
9.1 Discrete Time Aperiodic Signals . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 121
9.2 Eigenfunctions of Discrete Time LTI Systems . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 124
9.3 Discrete Time Fourier Transform (DTFT) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 125
9.4 Properties of the DTFT . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 128
9.5 Common Discrete Time Fourier Transforms . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .. . . . . . . . . . . . 132
9.6 Discrete Time Convolution and the DTFT . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 133
10 Computing Fourier Transforms
10.1 Discrete Fourier Transform (DFT) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 137
10.2 DFT: Fast Fourier Transform . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 138
10.3 The Fast Fourier Transform (FFT) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 139
Solutions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 145
11 Sampling and Reconstruction

11.1 Signal Sampling . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 147
11.2 Sampling Theorem . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 149
11.3 Signal Reconstruction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 152
11.4 Perfect Reconstruction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 157
11.5 Aliasing Phenomena . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 160
11.6 Anti-Aliasing Filters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 163
11.7 Changing Sampling Rates in Discrete Time . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .. . . . . . . . . . . . 165
11.8 Discrete Time Processing of Continuous Time Signals . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 171
12 Appendix: Mathematical Pot-Pourri
12.1 Basic Linear Algebra . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 177
12.2 Linear Constant Coecient Dierence Equations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 182
12.3 Solving Linear Constant Coecient Dierence Equations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 183
Solutions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 187
13 Appendix: Viewing Interactive Content

13.1 Viewing Embedded LabVIEW Content in Connexions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 190
13.2 Getting Started With Mathematica . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 190
Glossary . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 193
Index . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 195
Attributions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 198
Available for free at Connexions <http://legacy.cnx.org/content/col11557/1.10>
Chapter 1
Review of Prerequisites: Complex
Numbers
1.1 Geometry of Complex Numbers
note: This module is part of the collection,
ing.
A First Course in Electrical and Computer Engineer-
The LaTeX source les for this collection were created using an optical character recognition
technology, and because of this process there may be more errors than usual. Please contact us if
you discover any errors.
The most fundamental new idea in the study of complex numbers is the imaginary number
imaginary number is dened to be the square root of
(1.1)
j 2 = 1.
(1.2)
is used to build complex numbers
and
in the following way:
z = x + jy.
We say that the complex number
In MATLAB, the variable

nication theory,
call
z =x + jy
has real part
(1.3)
and imaginary part
y:
z = Re [z] + jIm [z]
(1.4)
Re [z] = x; Im [z] = y.
(1.5)
is denoted by
real(z),
and the variable
is called the in-phase component of
the
This
j=
The imaginary number
j.
1:
Cartesian
z,
and
is denoted by
imag(z).
In commu-
is called the quadrature component. We
representation of z , with real component x and imaginary component y . We

(x, y)codes the complex number z .
complex number z on the plane as in Figure 1.1. We call the horizontal axis the real
say that the Cartesian pair

We may plot the
axis and the vertical axis the imaginary axis.

angle of the line to the point
z = x + jy
The plane is called the complex plane.
r=
1 This
The radius and
are
p
x2 + y 2
content is available online at <http://legacy.cnx.org/content/m21411/1.6/>.

1
(1.6)
CHAPTER 1. REVIEW OF PREREQUISITES: COMPLEX NUMBERS
= tan1
See Figure 1.1. In MATLAB,
Figure 1.1:
is denoted by
abs(z),
y
x
and
(1.7)
is denoted by
angle(z).
Cartesian and Polar Representations of the Complex Number z
The original Cartesian representation is obtained from the radius
The complex number
and angle
as follows:
x = rcos
(1.8)
y = r sin .
(1.9)
may therefore be written as
x + jy
rcos + jrsin
(1.10)
= r ( cos + j sin ) .
The complex number cos
+ j sin is, itself, a number that may be represented on the complex plane and
(cos, sin). This is illustrated in Figure 1.2. The radius and angle to the
1 and . Can you see why?
coded with the Cartesian pair

point
z = cos + j sin
Figure 1.2:
are
The Complex Number cos + jsin
The complex number cos
+ j sin is
ej :
of such fundamental importance to our study of complex numbers
that we give it the special symbol
ej = cos + j sin.
As illustrated in Figure 1.2, the complex number
may write the complex number
ej
(1.11)
has radius 1 and angle
With the symbol
complex number
z.
We call
z = re
to be the angle, or
polar
(1.12)
z . We say that the polar pair rcodes the

|z| = r to be the magnitude of z and arg (z) =
representation for the complex number
In this polar representation, we dene
phase, of z :
|z| = r
(1.13)
arg (z) = .
(1.14)
With these denitions of magnitude and phase, we can write the complex number
as
z = |z|ejarg(z) .
Let's summarize our ways of writing the complex number
= x + jy
In "Roots of Quadratic Equations"

We show, in fact, that
call
ej
rej
(x, y)
and record the corresponding geometric codes:
= |z|ej
arg(z)
.
(1.16)
we show that the denition
is just the familiar function
(1.15)
ej = cos + j sin
is more than symbolic.
evaluated at the imaginary argument
x = j.
a complex exponential, meaning that it is an exponential with an imaginary argument.
Exercise 1.1.1
Prove
2n
(j)
= (1)
Exercise 1.1.2
and
2n+1
(j)
= (1) j .
Evaluate
j3, j4, j5.
ej[(/2)+m2] = j, ej[(3/2)+m2] = j, ej(0+m2) = 1,

identities on the complex plane. (Assume m is an integer.)
Prove
Exercise 1.1.3
Find the polar representation

a.
b.
c.
d.
z
z
z
z
z = rej
and
ej(+m2) = 1.
Plot these
for each of the following complex numbers:
= 1 + j0;
= 0 + j1;
= 1 + j1;
= 1 j1.
Plot the points on the complex plane.
Exercise 1.1.4
Find the Cartesian representation

a.
b.
c.
z = x + jy
for each of the following complex numbers:
z = 2ej/2 ;
z = 2ej/4 ;
z = ej3/4 ;
2 "Complex
we
as
z = rej .
j
ej ,
Numbers: Roots of Quadratic Equations" <http://legacy.cnx.org/content/m21415/latest/>

We
d.
z=
2ej3/2 .
Plot the points on the complex plane.
Exercise 1.1.5
The following geometric codes represent complex numbers.

corresponding complex number
a.
b.
c.
d.
(0.7, 0.1) z =
(1.0, 0.5) z =
1.6/8 z =?
0.47/8 z =?
z:
Decode each by writing down the
?
?
Exercise 1.1.6
Show that
Im [jz] = Re [z]
and
Re [jz] = Im [z].
Demo 1.1 (MATLAB). Run the following
MATLAB program in order to compute and plot the complex number
ej
for
= i2/360, i =
1, 2, ..., 360:
j=sqrt(-1)
n=360
for i=1:n,circle(i)=exp(j*2*pi*i/n);end;
axis('square')
plot(circle)
Replace the explicit for loop of line 3 by the implicit loop
circle=exp(j*2*pi*[1:n]/n);
to speed up the calculation. You can see from Figure 1.3 that the complex number
= 2/360, 2 (2/360) , ..., turns

j
that e
is a complex number that
ej ,
evaluated
at angles
out complex numbers that lie at angle
We say
lies on the unit circle. We will have much more to say
about the unit circle in Chapter 2.
and radius 1.
Figure 1.3:
The Complex Numbers ej for 0 2 (Demo 1.1)
1.2 Complex Numbers: Algebra of Complex Numbers

ing.
The complex numbers form a mathematical eld on which the usual operations of addition and multiplication are dened. Each of these operations has a simple geometric interpretation.
1.2.1 Addition and Multiplication.

The complex numbers
z1
and
z2
are
added according to the rule
z1 + z2
3 This
(x1 + jy1 ) + (x2 + jy2 )
(x1 + x2 ) + j (y1 + y2 ) .
(1.17)
We say that the real parts add and the imaginary parts add.
number
formed
z1 + z2
from z1
is computed from a parallelogram rule, wherein

and
only
z1 = r1 ej1
z1
z2 = r2 ej2 . Find a polar formula z3 =r3 ej3 for z3 = z1 + z2

r1 , r2 , 1 , and 2 . The formula for r3 is the law of cosines.
and
the variables
The product of
and
z2
that involves
is
z1 z2
Figure 1.4:
lies on the node of a parallelogram
z2 .
Exercise 1.2.1
Let
As illustrated in Figure 1.4, the complex
z1 + z2
(x1 + jy1 ) (x2 + jy2 )
(x1 x2 y1 y2 ) + j (y1 x2 + x1 y2 ) .
(1.18)
Adding Complex Numbers
If the polar representations for
z1 z2
z1
and
z2
are used, then the product may be written as
r1 ej1 r2 ej2
(r1 cos1 + jr1 sin1 ) (r2 cos2 + jr2 sin2 )
( r1 cos 1 r2 cos 2 r1 sin 1 r2 sin 2 )
+ j ( r1 sin 1 r2 cos 2 + r1 cos 1 r2 sin 2 )

=
r1 r2 cos (1 + 2 ) + jr1 r2 sin (1 + 2 )
r1 r2 ej(1 +2 ) .
We say that the magnitudes multiply and the angles add. As illustrated in Figure 1.5, the product
at the angle
(1.19)
z1 z2
lies
(1 + 2 ).
4 We have used the trigonometric identities cos ( + ) = cos cos sin sin and sin ( + ) = sin cos +cos
1
2
1
2
1
2
1
2
1
2
1
sin 2 to derive this result.
Figure 1.5:
Rotation.
Multiplying Complex Numbers
There is a special case of complex multiplication that will become very important in our study
5
of phasors in the chapter on Phasors . When

number
z2 = e
j2
, then the product of
z1
and
z1 is
z2 is
the complex number
z1 = r1 ej1
and
z2
is the complex
z1 z2 = z1 ej2 = r1 ej(1 +2 ) .
As illustrated in Figure 1.6,
Figure 1.6:
z1 z2
is just a rotation of
z1
through the angle
(1.20)
2 .
Rotation of Complex Numbers
Exercise 1.2.2
z1 = x + jy = rej . Compute the complex number z2 = jz1 in

Cartesian and polar forms. The complex number z2 is sometimes called perp(z1 ). Explain why
j
writing perp(z1 ) as z1 e 2 . What is 2 ? Repeat this problem for z3 = jz1 .
Begin with the complex number
5 "Phasors:
Introduction" <http://legacy.cnx.org/content/m21469/latest/>
its
by
Powers.
If the complex number
z1
multiplies itself
(z1 )
times, then the result is
= r1N ejN 1 .
(1.21)
z1k = r1k ejk1 . (The assumption is

k+1
k+1 j(k+1)1
k
true for k = 1.) Then use the recursion z1
= z1 z1 = r1 e
. Iterate this recursion (or induction)
until k + 1 = N . Can you see that, as n ranges from n = 1, ..., N , the angle of zfrom 1 to 21 , ..., to N 1
2
N
and the radius ranges from r1 to r1 , ..., to r1 ? This result is explored more fully in Problem 1.19.
j
Corresponding to every complex number z = x + jy = re
is the complex
This result may be proved with a simple induction argument. Assume
Complex Conjugate.
conjugate
z = x jy = rej .
z
The complex number
and its complex conjugate are illustrated in Figure 1.7.
complex conjugates is to change
jto j .
(1.22)
The recipe for nding
This changes the sign of the imaginary part of the complex
number.
Figure 1.7:
A Complex Variable and Its Complex Conjugate
Magnitude Squared.
and is denoted by
|z|
The product of
and its complex conjugate is called the magnitude squared of
|z| = z z = (x jy) (x + jy) = x2 + y 2 = rej rej = r2 .

Note that
|z| = r
(1.23)
is the radius, or magnitude, that we dened in "Geometry of Complex Numbers"
(Section 1.1).
Exercise 1.2.3
Write
as
z = zw.
Exercise 1.2.4
Prove that angle
Exercise 1.2.5
Find
in its Cartesian and polar forms.
(z2 z1 ) = 2 1 .
Show that the real and imaginary parts of
z = x + jy
may be written as
1
(z + z )
2
(1.24)
Im [z] = 2j (z z ) .
(1.25)
Re [z] =
Commutativity, Associativity, and Distributivity.
The complex numbers commute, associate, and
distribute under addition and multiplication as follows:
z1 + z2
= z2 + z1
z1 z2
(z1 + z2 ) + z3
Identities and Inverses.
(1.26)
z2 z1
= z1 + (z2 + z3 )
(1.27)
z1 (z2 z3 )
(z1 z2 ) z3
z1 (z2 + z3 )
z1 z2 + z1 z3 .
In the eld of complex numbers, the complex number
plays the role of an additive identity, and the complex number
1 + j0
0 + j0
(denoted by 0)
(denoted by 1) plays the role of a
multiplicative identity:
In this eld, the complex number
x
x2 +y 2
+j
y
x2 +y 2
z+0
0+z
z1
1z.
z = x + j (y)
is the additive inverse of
zz
Exercise 1.2.6
Show that the additive inverse of
z = rej
Exercise 1.2.7
Show that the multiplicative inverse of
z 1 =
zz
z,
and the complex number
is the multiplicative inverse:
z + (z)
Show that
(1.28)
is real. Show that
z 1
1.
(1.29)
may be written as
rej(+) .
may be written as
1
1
z = 2
(x jy) .
zz
x + y2
(1.30)
may also be written as
z 1 = r1 ej .
Plot
and
z 1
Exercise 1.2.8
Prove
(j)
for a representative
(1.31)
z.
= j .
Exercise 1.2.9
Find
z 1
when
Exercise 1.2.10

Prove
z 1
z = 1 + j1.
1
= (z )
= r1 ej =
Exercise 1.2.11
Find all of the complex numbers
1
z z z . Plot
and
z 1
for a representative
with the property that
jz = z .
z.
Illustrate these complex
numbers on the complex plane.
Demo 1.2 (MATLAB). Create and run the following script le (name it Complex Numbers)
6 If
you are using PC-MATLAB, you will need to name your le cmplxnos.m.
10
clear, clg
j=sqrt(-1)
z1=1+j*.5,z2=2+j*1.5
z3=z1+z2,z4=z1*z2
z5=conj(z1),z6=j*z2
axis([-4 4 -4 4]),axis('square'),plot([0 z1],'-o')
hold on
plot([0 z2],'-o'),plot([0 z3],'-+'),plot([0 z4],'-*'),
plot([0 z5],'x'),plot([0 z6],'-x')
Figure 1.8:
Complex Numbers (Demo 1.2)
With the help of Appendix 1, you should be able to annotate each line of this program. View your graphics
display to verify the rules for add, multiply, conjugate, and perp. See Figure 1.8.
Exercise 1.2.12
Prove that
z 0 = 1.
Exercise 1.2.13
z1 = 1.05ej2/16 and z2 = 0.95ej2/16 . Write a MATLAB program to compute

n
z2 for n = 1, 2, ..., 32. You should observe a gure like Figure 1.9.
(MATLAB) Choose
n
and plot z1 and
11
Figure 1.9:
Powers of z
1.3 Representing Complex Numbers in a Vector Space

ing.
So far we have coded the complex number
(r).
z = x + jy with the Cartesian pair (x, y) and with the polar pair
z may be coded with a two-dimensional vector z and show
We now show how the complex number
how this new code may be used to gain insight about complex numbers.
Coding a Complex Number
as a Vector.
two-dimensional vector
z=
x
y
We code the complex number
z = x + jy
with the
x + jy = z z =
x
y
We plot this vector as in Figure 1.10. We say that the vector
.
z
belongs to a vector space.
(1.32)
This means
that vectors may be added and scaled according to the rules
z1 + z2
x1 + x2
y1 + y2
ax
.
az =
ay
7 This
(1.33)
(1.34)
12
Figure 1.10:
The Vector z Coding the Complex Number z
Furthermore, it means that an additive inverse
z,
an additive identity
0,
and a multiplicative identity
1 all exist:
The vector
is
0=
0
0
z + (z) = 0
(1.35)
lz = z.
(1.36)
Prove that vector addition and scalar multiplication satisfy these properties of commutation, association,
and distribution:
z1 + z2 = z2 + z1
(1.37)
(z1 + z2 ) + z3 = z1 + (z2 + z3 )
(1.38)
a (bz) = (ab) z
(1.39)
a (z1 + z2 ) = az1 + az2 .
Inner Product and Norm.
The
inner product
between two vectors
(1.40)
z1
and
z2
is dened to be the real
number
(z1 , z2 ) = x1 x2 + y1 y2 .
(1.41)
8
We sometimes write this inner product as the vector product (more on this in Linear Algebra )
(z1 , z2 )
[x1 y1 ]
x2
y2
zT z
1 2
= (x1 x2 + y1 y2 ) .
Exercise 1.3.1
Prove
8 "Linear
(z1 , z2 ) = (z2 , z1 ) .
Algebra: Introduction" <http://legacy.cnx.org/content/m21454/latest/>

(1.42)
13
When
z1 = z2 = z,
then the inner product between
and itself is the
norm squared
of
z:
||z|| = (z, z) = x2 + y 2 .
(1.43)
These properties of vectors seem abstract. However, as we now show, they may be used to develop a vector
calculus for doing complex arithmetic.
A Vector Calculus for Complex Arithmetic.
corresponds to the addition of the vectors
z1
z2
and
The addition of two complex numbers
z1 + z2 z1 + z2
The scalar multiplication of the complex number
cation of the vector
z2
by
x1
z1
and
z2
z2
x1 + x2
y1 + y2
(1.44)
by the real number
x1
corresponds to scalar multipli-
x2
x1 z2 x1
y2
x1 x2
x1 y2
(1.45)
z2 by the real number y1

x2
y1 x2
=
.
y1 z2 y1
y2
y1 y2
Similarly, the multiplication of the complex number
is
The complex product
z1 z2 = (x1 + jy1 ) z2
(1.46)
is therefore represented as
z1 z2
x1 x2 y1 y2
x1 y2 + y1 x2
(1.47)
This representation may be written as the inner product
z1 z2 = z2 z1
where
and
are the vectors
(v, z1 )
(w, z1 )
x2
y2
and w =
.
v=
y2
x2
x y2
2
,
y2 x2
we can represent the complex product
z1 z2
(1.48)
By dening the matrix
(1.49)
as a matrix-vector multiply (more on this in Linear Algebra ):
z1 z2 = z2 z1
x2
y2
y2
x2
x1
y1
(1.50)
With this representation, we can represent rotation as
zej = ej z
9 "Linear
cos
sin
sin
cos
x1
x2
Algebra: Introduction" <http://legacy.cnx.org/content/m21454/latest/>

(1.51)
14
We call the matrix
cos
sin
sin
cos
a rotation matrix.
Exercise 1.3.2
Call
R ()
the rotation matrix:
R () =
Show that
R ()
Exercise 1.3.3
rotates by
().
z
and
in the polar representation
If we dene the coordinate vectors
cos
(1.52)
R () w
when
w = R () z?
x
y
(1.53)
of the matrix.
Inner Product and Polar Representation.

magnitude of
sin
as
a, b, c,
sin
What can you say about
Represent the complex conjugate of
and nd the elements
cos
From the norm of a vector, we derive a formula for the
z = rej
1/2
1/2
r = x2 + y 2
= ||z|| = (z, z) .

0
1
and e2 = , then we
e1 =
0
1
(1.54)
can represent the vector
z = (z, e1 ) e1 + (z, e2 ) e2 .
See Figure 1.11. From the gure it is clear that the cosine and sine of the angle
cos =
Figure 1.11:
as
(1.55)
are
(z, e2 )
(z, e1 )
; sin =
||z||
||z||
Representation of z in its Natural Basis
(1.56)
15
z:
This gives us another representation for any vector
z = ||z||cose1 + ||z||sine2 .
The inner product between two vectors
(z1 , z2 )
z1
and
z2
(1.57)
is now

(z
,
e
)
e
2 1
1
(z1 , e1 ) eT1 (z1 , e2 ) eT2

(z2 , e2 ) e2
(z1 , e1 ) (z2 , e1 ) + (z1 , e2 ) (z2 , e2 )
(1.58)
= ||z1 ||cos1 ||z2 ||cos2 + ||z1 || sin 1 ||z2 ||sin2 .

It follows that
cos (2 1 ) = cos2
cos
1 + sin1 sin2
cos (2 1 ) =
may be written as
(z1 , z2 )
||z1 || ||z2 ||
(1.59)
This formula shows that the cosine of the angle between two vectors
cosine of the angle of
Exercise 1.3.4
z2 z1 ,
z1
and
z2 ,
which is, of course, the
is the ratio of the inner product to the norms.
Prove the Schwarz and triangle inequalities and interpret them:
(Schwarz) (z1 , z2 ) ||z1 || ||z2 ||
(1.60)
(triangle) I z1 z2 || ||z1 z3 || + ||z2 z3 ||.
(1.61)
16
Chapter 2
Continuous-Time Signals
2.1 Signal Classications and Properties
2.1.1 Introduction
This module will begin our study of signals and systems by laying out some of the fundamentals of signal classication. It is essentially an introduction to the important denitions and properties that are fundamental
to the discussion of signals and systems, with a brief discussion of each.
2.1.2 Classications of Signals

2.1.2.1 Continuous-Time vs. Discrete-Time
As the names suggest, this classication is determined by whether or not the time axis is
or
continuous (Figure 2.1).
discrete (countable)
A continuous-time signal will contain a value for all real numbers along the
2
time axis. In contrast to this, a discrete-time signal , often created by sampling a continuous signal, will
only have values at equally spaced intervals along the time axis.
Figure 2.1
1 This content is available

2 "Discrete-Time Signals"
online at <http://legacy.cnx.org/content/m47271/1.3/>.
<http://legacy.cnx.org/content/m0009/latest/>

17
CHAPTER 2. CONTINUOUS-TIME SIGNALS
18
2.1.2.2 Analog vs. Digital

The dierence between analog and digital is similar to the dierence between continuous-time and discretetime.
However, in this case the dierence involves the values of the function.
Analog corresponds to a
continuous set of possible function values, while digital corresponds to a discrete set of possible function
values. An common example of a digital signal is a binary sequence, where the values of the function can
only be one or zero.
Figure 2.2
2.1.2.3 Periodic vs. Aperiodic

3
Periodic signals
repeat with some
period T , while aperiodic, or nonperiodic, signals do not (Figure 2.3).
We can dene a periodic function through the following mathematical expression, where
and
t can be any number
is a positive constant:
f (t) = f (T + t)
The
fundamental period of our function, f (t), is the smallest value of T
(2.1)
that the still allows (2.1) to be
true.
3 "Continuous
Time Periodic Signals" <http://legacy.cnx.org/content/m10744/latest/>
19
(a)
(b)
Figure 2.3:
(a) A periodic signal with period T0 (b) An aperiodic signal
2.1.2.4 Finite vs. Innite Length

As the name implies, signals can be characterized as to whether they have a nite or innite length set of
values. Most nite length signals are used when dealing with discrete-time signals or a given sequence of
values. Mathematically speaking,
f (t)
is a
nite-length signal if it is nonzero over a nite interval

t1 < f (t) < t2
where
f (t),
t1 >
and
t2 < .
An example can be seen in Figure 2.4. Similarly, an
innite-length signal,
is dened as nonzero over all real numbers:
f (t)
Figure 2.4:
Finite-Length Signal. Note that it only has nonzero values on a set, nite interval.
20
2.1.2.5 Even vs. Odd

An even signal is any
are symmetric around
f (t) = f (t)
signal
such that
the vertical axis.
f (t) = f (t).
An
odd signal,
Even signals can be easily spotted as they

on the other hand, is a signal
such that
(Figure 2.5).
(a)
(b)
Figure 2.5:
(a) An even signal (b) An odd signal
Using the denitions of even and odd signals, we can show that any signal can be written as a combination
of an even and odd signal. That is, every signal has an odd-even decomposition. To demonstrate this, we
have to look no further than a single equation.
f (t) =
1
1
(f (t) + f (t)) + (f (t) f (t))
2
2
By multiplying and adding this expression out, it can be shown to be true.
f (t) + f (t)
fullls the requirement of an even function, while
f (t) f (t)
(2.2)
Also, it can be shown that
fullls the requirement of an
odd function (Figure 2.6).
Example 2.1
21
(a)
(b)
(c)

(d)
(a) The signal we will decompose using odd-even decomposition (b) Even part: e (t) =
(c) Odd part: o (t) = 21 (f (t) f (t)) (d) Check: e (t) + o (t) = f (t)
Figure 2.6:
1
(f (t) + f (t))
2
22
Example 2.2
Consider the signal dened for all real
described by
f (t) = {
sin (2t) /t
t1
t<1
(2.3)
This signal is continuous time, analog, aperiodic, innite length, causal, and neither even nor odd.
2.1.3 Signal Classications Summary

This module describes just some of the many ways in which signals can be classied. They can be continuous
time or discrete time, analog or digital, periodic or aperiodic, nite or innite, and deterministic or random.
We can also divide them based on their causality and symmetry properties. There are other ways to classify
signals, such as boundedness, handedness, and continuity, that are not discussed here but will be described
in subsequent modules.
2.2 Common Continuous Time Signals
2.2.1 Introduction
Before looking at this module, hopefully you have an idea of what a signal is and what basic classications
and properties a signal can have. In review, a signal is a function dened with respect to an independent
variable. This variable is often time but could represent any number of things. Mathematically, continuous
time analog signals have continuous independent and dependent variables. This module will describe some
useful continuous time analog signals.
2.2.2 Important Continuous Time Signals

2.2.2.1 Sinusoids
One of the most important elemental signal that you will deal with is the real-valued sinusoid.
In its
continuous-time form, we write the general expression as
Acos (t + )
where
is the amplitude,
is the frequency, and
is the phase. Thus, the period of the sinusoid is
T =
4 This
(2.4)
(2.5)
23
Figure 2.7:
Sinusoid with A = 2, w = 2, and = 0.
2.2.2.2 Unit Step

Another very basic signal is the
unit-step function that is dened as
0
u (t) =
1
if
t<0
if
t0
(2.6)
1
t
Figure 2.8:
Continuous-Time Unit-Step Function
The step function is a useful tool for testing and for dening other signals. For example, when dierent
shifted versions of the step function are multiplied by other signals, one can select a certain portion of the
signal and zero out the rest.
24
2.2.2.3 Unit Pulse

Many engineers interpret the unit step function as the representation of turning on a switch and leaving it
on. The
unit-pulse function can be thought of as turning a switch on and o after a unit of time.
dened as
1
p (t) =
0
0.5 t 0.5
if
if
t < 0.5 or t > 0.5
It is
(2.7)
so that it is an even function.
Figure 2.9:
Continuous-Time Unit-Step Function
Note that the pulse can be easily written in terms of unit step functions as
p (t) = u (t + 0.5) u (t 0.5)
2.2.2.4 Triangle Function

The last function we will introduce is the
triangle function, which represents and input that increases and
then decreases linearly with time. It is dened as
(t) =
t+1
1t
if
if
if
1t0
0t1
(2.8)
t < 1 or t > 1
25
Figure 2.10:
Continuous-Time Triangle Function
2.2.3 Common Continuous Time Signals Summary

Some of the most important and most frequently encountered signals have been discussed in this module.
There are, of course, many other signals of signicant consequence not discussed here. As you will see later,
many of the other more complicated signals will be studied in terms of those listed here. Especially take
note of the complex exponentials and unit impulse functions, which will be the key focus of several topics
included in this course.
2.3 Signal Operations
2.3.1 Introduction
This module will look at two signal operations aecting the time parameter of the signal, time shifting and
time scaling. These operations are very common components to real-world systems and, as such, should be
understood thoroughly when learning about signals and systems.
2.3.2 Manipulating the Time Parameter

2.3.2.1 Time Shifting
Time shifting is, as the name suggests, the shifting of a signal in time. This is done by adding or subtracting
a quantity of the shift to the time variable in the function. Subtracting a xed positive quantity from the
time variable will shift the signal to the right (delay) by the subtracted quantity, while adding a xed positive
amount to the time variable will shift the signal to the left (advance) by the added quantity.
5 This
26
Figure 2.11:
f (t T ) moves (delays) f to the right by T .
2.3.2.2 Time Scaling

Time scaling compresses or dilates a signal by multiplying the time variable by some quantity. If that quantity's absolute value is greater than one, the signal becomes narrower and the operation is called compression,
while if the quantity's absolute value is less than one, the signal becomes wider and is called dilation. Note
that if the quantity is negative, then one must also account for time reversal (described below).
Figure 2.12:
f (at) compresses f by a.
Example 2.3
Given
f (t)
we woul like to plot
f (at b),
with both
a>0
and
b > 0.
The gure below describes
a method to accomplish this.
27
(a)
(b)
(c)
Figure 2.13:
t ab
f
to get
` (a)
` Begin
with f (t) (b) Then replace t with at to get f (at) (c) Finally, replace t with
a t ab = f (at b)
2.3.2.3 Time Reversal

A natural question to consider when learning about time scaling is: What happens when the time variable
is multiplied by a negative number? The answer to this is time reversal, also known as time inversion. This
operation is the reversal of the time axis, or ipping the signal over the y-axis.
Figure 2.14:
Reverse the time axis
28
2.3.3 Time Scaling and Shifting Demonstration
Download6 or Interact (when online) with a Mathematica CDF demonstrating Discrete

Harmonic Sinusoids.
Figure 2.15:
2.3.4 Signal Operations Summary

Some common operations on signals aect the time parameter of the signal. One of these is time shifting in
which a quantity is added to the time parameter in order to advance or delay the signal. Another is the time
scaling in which the time parameter is multiplied by a quantity in order to dilate or compress the signal in
time. In the event that the quantity involved in the latter operation is negative, time reversal occurs.
2.4 Energy and Power of Continuous-Time Signals
From physics we've learned that energy is work and power is work per time unit. Energy was measured in
Joule (J) and work in Watts(W). In signal processing energy and power are dened more loosely without
any necessary physical units, because the signals may represent very dierent physical entities. We can say
that energy and power are a measure of the signal's "size".
6 See the le at <http://legacy.cnx.org/content/m50957/latest/TimeshifterDrill_display.cdf>

7 This content is available online at <http://legacy.cnx.org/content/m47273/1.4/>.
29
2.4.1 Signal Energy

2.4.1.1 Analog signals
Since we often think of a signal as a function of varying amplitude through time, it seems to reason that a
good measurement of the strength of a signal would be the area under the curve. However, this area may
have a negative part. This negative part does not have less strength than a positive signal of the same size.
This suggests either squaring the signal or taking its absolute value, then nding the area under that curve.
It turns out that what we call the energy of a signal is the area under the squared signal, see Figure 2.16
Energy - Analog signal:
Ea =
(|x (t) |) dt
Note that we have used squared magnitude(absolute value) if the signal should be complex valued. If the
signal is real, we can leave out the magnitude operation.
(a)
(b)
Figure 2.16:
Sketch of energy calculation (a) Signal x(t) (b) The energy of x(t) is the shaded region
2.4.2 Signal Power

Our denition of energy seems reasonable, and it is. However, what if the signal does not decay fast enough?
In this case we have innite energy for any such signal. Does this mean that a fty hertz sine wave feeding
into your headphones is as strong as the fty hertz sine wave coming out of your outlet? Obviously not.
This is what leads us to the idea of
signal power, which in such cases is a more adequate description.
30
Figure 2.17:
Signal with ininite energy
2.4.2.1 Analog signals

For analog signals we dene power as
Power - analog signal:
energy per time interval.

1
T T
Pa = lim
T
2
T2
(|x (t) |) dt
For periodic analog signals, the power needs to only be measured across a single period.
Power - periodic analog signal with period
T0 : Pa =
1
T0
Example 2.4
Given the signal
x (t) = sin (2t), shown in Figure 2.18, calculate

R1
Pa = 11 0 sin2 (2t) dt = 12 .
T0
2
T
20
(|x (t) |) dt
the power for one period.
For the analog sine we have
Figure 2.18:
Analog sine.
31
2.5 Continuous Time Impulse Function
2.5.1 Introduction
In engineering, we often deal with the idea of an action occurring at a point.
Whether it be a force at
a point in space or some other signal at a point in time, it becomes worth while to develop some way of
quantitatively dening this. This leads us to the idea of a unit impulse, probably the second most important
function, next to the complex exponential, in this systems and signals course.
2.5.2 Dirac Delta Function

The
Dirac delta function,
often referred to as the unit impulse or delta function, is the function that
denes the idea of a unit impulse in continuous-time. Informally, this function is one that is innitesimally
narrow, innitely tall, yet integrates to one. Perhaps the simplest way to visualize this is as a rectangular

1

2 to a + 2 with a height of . As we take the limit of this setup as approaches 0, we see
that the width tends to zero and the height tends to innity as the total area remains constant at one. The
pulse from
impulse function is often written as
(t).
Z
(t) dt = 1
Figure 2.19:
8 This
This is one way to visualize the Dirac Delta Function.
(2.9)
32
Since it is quite dicult to draw something that is innitely tall, we represent the Dirac
with an arrow centered at the point it is applied. If we wish to scale it, we may write the value it is
scaled by next to the point of the arrow. This is a unit impulse (no scaling).
Figure 2.20:
Below is a brief list a few important properties of the unit impulse without going into detail of their
proofs.
Unit Impulse Properties
1
(t) = ||
(t)
(t) = (t)
d
(t) = dt
u (t), where u (t)
f (t) (t) = f (0) (t)
is the unit step.
The last of these is especially important as it gives rise to the sifting property of the dirac delta function, which
selects the value of a function at a specic time and is especially important in studying the relationship of an
operation called convolution to time domain analysis of linear time invariant systems. The sifting property
is shown and derived below.
f (t) (t) dt =
f (0) (t) dt = f (0)
(t) dt = f (0)
(2.10)
33
2.5.3 Unit Impulse Limiting Demonstration
Figure 2.21: Click on the above thumbnail image (when online) to download an interactive Mathematica
Player demonstrating the Continuous Time Impulse Function.
2.5.4 Continuous Time Unit Impulse Summary

The continuous time unit impulse function, also known as the Dirac delta function, is of great importance
to the study of signals and systems. Informally, it is a function with innite height ant innitesimal width
that integrates to one, which can be viewed as the limiting behavior of a unit area rectangle as it narrows
while preserving area. It has several important properties that will appear again when studying systems.
2.6 Continuous-Time Complex Exponential
2.6.1 Introduction
Complex exponentials are some of the most important functions in our study of signals and systems. Their
importance stems from their status as eigenfunctions of linear time invariant systems. Before proceeding,
you should be familiar with complex numbers.
9 This
34
2.6.2 The Continuous Time Complex Exponential

2.6.2.1 Complex Exponentials
The complex exponential function will become a critical part of your study of signals and systems. Its general
continuous form is written as
Aest
where
s = + j
is a complex number in terms of
(2.11)
, the attenuation constant, and
the angular frequency.
2.6.2.2 Euler's Formula

The mathematician Euler proved an important identity relating complex exponentials to trigonometric functions. Specically, he discovered the eponymously named identity, Euler's formula, which states that
ejx = cos (x) + jsin (x)
(2.12)
which can be proven as follows.

In order to prove Euler's formula, we start by evaluating the Taylor series for
converges for all complex
z,
at
z = jx.
about
z = 0,
which
The result is
ejx
=
=
ez
k=0
k=0
k x2k
(2k)!
(1)
+j
(jx)k
k!
P
k=0
k x2k+1
(2k+1)!
(2.13)
(1)
= cos (x) + jsin (x)

because the second expression contains the Taylor series for
for all real
x.
Choosing
cos (x) and sin (x) about t = 0,
which converge
Thus, the desired result is proven.
x = t
this gives the result
ejt = cos (t) + jsin (t)
(2.14)
which breaks a continuous time complex exponential into its real part and imaginary part.
Using this
formula, we can also derive the following relationships.
1 jt 1 jt
e + e
2
2
(2.15)
1
1 jt
e ejt
2j
2j
(2.16)
cos (t) =
sin (t) =
2.6.2.3 Continuous Time Phasors

It has been shown how the complex exponential with purely imaginary frequency can be broken up into
its real part and its imaginary part. Now consider a general complex frequency
attenuation factor and
is the frequency. Also consider a phase dierence
s = + j
e(+j)t+j = et (cos (t + ) + jsin (t + )) .

Thus, the real and imaginary parts of
st
where
is the
It follows that
(2.17)
appear below.
Re{e(+j)t+j } = et cos (t + )
(2.18)
Im{e(+j)t+j } = et sin (t + )
(2.19)
35
Using the real or imaginary parts of complex exponential to represent sinusoids with a phase delay multiplied
by real exponential is often useful and is called attenuated phasor notation.
We can see that both the real part and the imaginary part have a sinusoid times a real exponential. We
also know that sinusoids oscillate between one and negative one. From this it becomes apparent that the
real and imaginary parts of the complex exponential will each oscillate within an envelope dened by the
real exponential part.
(a)
(b)
(c)
The shapes possible for the real part of a complex exponential. Notice that the oscillations
are the result of a cosine, as there is a local maximum at t = 0. (a) If is negative, we have the case of
a decaying exponential window. (b) If is positive, we have the case of a growing exponential window.
(c) If is zero, we have the case of a constant window.
Figure 2.22:
36
2.6.3 Complex Exponential Demonstration
Interact (when online) with a Mathematica CDF demonstrating the Continuous Time
Complex Exponential. To Download, right-click and save target as .cdf.
Figure 2.23:
2.6.4 Continuous Time Complex Exponential Summary

Continuous time complex exponentials are signals of great importance to the study of signals and systems.
They can be related to sinusoids through Euler's formula, which identies the real and imaginary parts of
purely imaginary complex exponentials. Eulers formula reveals that, in general, the real and imaginary parts
of complex exponentials are sinusoids multiplied by real exponentials. Thus, attenuated phasor notation is
often useful in studying these signals.
Chapter 3
Introduction to Systems
3.1 Introduction to Systems
Signals are manipulated by systems.

y (t) = S (x (t)),
with
Mathematically, we represent what a system does by the notation
representing the input signal and
the output signal.
Denition of a system
x(t)
System
y(t)
The system depicted has input x (t) and output y (t). Mathematically, systems operate
on function(s) to produce other function(s). In many ways, systems are like functions, rules that yield a
value for the dependent variable (our output signal) for each value of its independent variable (its input
signal). The notation y (t) = S (x (t)) corresponds to this block diagram. We term S () the input-output
relation for the system.
Figure 3.1:
This notation mimics the mathematical symbology of a function: A system's input is analogous to an
independent variable and its output the dependent variable. For the mathematically inclined, a system is a
functional:
a function of a function (signals are functions).
Simple systems can be connected togetherone system's output becomes another's inputto accomplish
some overall design. Interconnection topologies can be quite complicated, but usually consist of weaves of
three basic interconnection forms.
1 This

37
CHAPTER 3. INTRODUCTION TO SYSTEMS
38
3.1.1 Cascade Interconnection

cascade
x(t)
w(t)
S1[]
y(t)
S2[]
The most rudimentary ways of interconnecting systems are shown in the gures in this
section. This is the cascade conguration.
Figure 3.2:
The simplest form is when one system's output is connected only to another's input.
w (t) = S1 (x (t)),
and
y (t) = S2 (w (t)),
with the information contained in
x (t)
Mathematically,
processed by the rst, then
the second system. In some cases, the ordering of the systems matter, in others it does not. For example, in
the fundamental model of communication
the ordering most certainly matters.
3.1.2 Parallel Interconnection

parallel
x(t)
S1[]
x(t)
+
S2[]
x(t)
Figure 3.3:
A signal
x (t)
y(t)
The parallel conguration.
is routed to two (or more) systems, with this signal appearing as the input to all systems
simultaneously and with equal strength.
Block diagrams have the convention that signals going to more
x (t) and their

y (t) = S1 (x (t))+S2 (x (t)), and the information
than one system are not split into pieces along the way. Two or more systems operate on
outputs are added together to create the output
in
x (t)
y (t).
Thus,
is processed separately by both systems.
2 "Structure of Communication Systems", Figure 1: Fundamental model of communication

<http://legacy.cnx.org/content/m0002/latest/#commsys>
39
3.1.3 Feedback Interconnection

feedback
x(t)
e(t)
y(t)
S1[]
S2[]
Figure 3.4:
The feedback conguration.
The subtlest interconnection conguration has a system's output also contributing to its input. Engineers
would say the output is "fed back" to the input through system 2, hence the terminology. The mathematical
statement of the feedback interconnection (Figure 3.4: feedback) is that the feed-forward system produces
y (t) = S1 (e (t)). The input e (t) equals the input signal minus the output of some other system's
y (t): e (t) = x (t) S2 (y (t)). Feedback systems are omnipresent in control problems, with the
the output:
output to
error signal used to adjust the output to achieve some condition dened by the input (controlling) signal.
For example, in a car's cruise control system,
x (t)
is a constant representing what speed you want, and
y (t)
is the car's speed as measured by a speedometer. In this application, system 2 is the identity system (output
equals input).
3.2 System Classications and Properties
3.2.1 Introduction
In this module some of the basic classications of systems will be briey introduced and the most important
properties of these systems are explained. As can be seen, the properties of a system provide an easy way
to distinguish one system from another. Understanding these basic dierences between systems, and their
properties, will be a fundamental concept used in all signal and system courses. Once a set of systems can be
identied as sharing particular properties, one no longer has to reprove a certain characteristic of a system
each time, but it can simply be known due to the the system classication.
3.2.2 Classication of Systems

3.2.2.1 Continuous vs. Discrete
One of the most important distinctions to understand is the dierence between discrete time and continuous
time systems. A system in which the input signal and output signal both have continuous domains is said
to be a continuous system. One in which the input signal and output signal both have discrete domains is
said to be a discrete system. Of course, it is possible to conceive of signals that belong to neither category,
such as systems in which sampling of a continuous time signal or reconstruction from a discrete time signal
take place.
3 This

40
3.2.2.2 Linear vs. Nonlinear

A linear system is any system that obeys the properties of scaling (rst order homogeneity) and superposition
(additivity) further described below. A nonlinear system is any system that does not have at least one of
these properties.
To show that a system
obeys the scaling property is to show that
H (kf (t)) = kH (f (t))
Figure 3.5:
A block diagram demonstrating the scaling property of linearity
To demonstrate that a system
obeys the superposition property of linearity is to show that
H (f1 (t) + f2 (t)) = H (f1 (t)) + H (f2 (t))
Figure 3.6:
(3.1)
(3.2)
A block diagram demonstrating the superposition property of linearity
It is possible to check a system for linearity in a single (though larger) step. To do this, simply combine
the rst two steps to get
H (k1 f1 (t) + k2 f2 (t)) = k1 H (f1 (t)) + k2 H (f2 (t))
(3.3)
41
3.2.2.3 Time Invariant vs. Time Varying

A system is said to be time invariant if it commutes with the parameter shift operator dened by
f (t T )
for all
T,
ST (f (t)) =
which is to say
HST = ST H
for all real
T.
(3.4)
Intuitively, that means that for any input function that produces some output function, any
time shift of that input function will produce an output function identical in every way except that it is
shifted by the same amount. Any system that does not have this property is said to be time varying.
This block diagram shows what the condition for time invariance. The output is the same
whether the delay is put on the input or the output.
Figure 3.7:
3.2.2.4 Causal vs. Noncausal

A causal system is one in which the output depends only on current or past inputs, but not future inputs.
Similarly, an anticausal system is one in which the output depends only on current or future inputs, but not
past inputs. Finally, a noncausal system is one in which the output depends on both past and future inputs.
All "realtime" systems must be causal, since they can not have future inputs available to them.
One may think the idea of future inputs does not seem to make much physical sense; however, we have
only been dealing with time as our dependent variable so far, which is not always the case. Imagine rather
that we wanted to do image processing. Then the dependent variable might represent pixel positions to the
left and right (the "future") of the current position on the image, and we would not necessarily have a causal
system.
42
(a)
(b)
(a) For a typical system to be causal... (b) ...the output at time t0 , y (t0 ), can only depend
on the portion of the input signal before t0 .
Figure 3.8:
3.2.2.5 Stable vs. Unstable

There are several denitions of stability, but the one that will be used most frequently in this course will
be bounded input, bounded output (BIBO) stability. In this context, a stable system is one in which the
output is bounded if the input is also bounded. Similarly, an unstable system is one in which at least one
bounded input produces an unbounded output.
In order to understand this concept, we must rst look more closely into exactly what we mean by
bounded. A bounded signal is any signal such that there exists a value such that the absolute value of the
signal is never greater than some value. Since this value is arbitrary, what we mean is that at no point can
the signal tend to innity, including the end behavior.
43
Figure 3.9:
A bounded signal is a signal for which there exists a constant A such that |f (t) | < A
Representing this mathematically, a stable system must have the following property, where
input and
y (t)
x (t)
is the
is the output. The output must satisfy the condition
|y (t) | My <
(3.5)
whenever we have an input to the system that satises
|x (t) | Mx <
Mx and My
(3.6)
both represent a set of nite positive numbers and these relationships hold for all of t. Otherwise,
the system is unstable.
3.2.3 System Classications Summary

This module describes just some of the many ways in which systems can be classied.
Systems can be
continuous time, discrete time, or neither. They can be linear or nonlinear, time invariant or time varying,
and stable or unstable. We can also divide them based on their causality properties. There are other ways
to classify systems, such as use of memory, that are not discussed here but will be described in subsequent
modules.
3.3 Linear Time Invariant Systems
3.3.1 Introduction
Linearity and time invariance are two system properties that greatly simplify the study of systems that exhibit
them. In our study of signals and systems, we will be especially interested in systems that demonstrate both
of these properties, which together allow the use of some of the most powerful tools of signal processing.
4 This

44
3.3.2 Linear Time Invariant Systems

3.3.2.1 Linear Systems
If a system is linear, this means that when an input to a given system is scaled by a value, the output of the
system is scaled by the same amount.
Linear Scaling
(a)
(b)
Figure 3.10
In Figure 3.10(a) above, an input
to the linear system
gives the output
y.
If
is scaled by a value
and passed through this same system, as in Figure 3.10(b), the output will also be scaled by
A linear system also obeys the principle of superposition.
This means that if two inputs are added
together and passed through a linear system, the output will be the sum of the individual inputs' outputs.
(a)
(b)
Figure 3.11
45
Superposition Principle
If Figure 3.11 is true, then the principle of superposition says that Figure 3.12 (Superposition Principle) is true as well. This holds for linear systems.
Figure 3.12:
That is, if Figure 3.11 is true, then Figure 3.12 (Superposition Principle) is also true for a linear system.
The scaling property mentioned above still holds in conjunction with the superposition principle. Therefore,
if the inputs x and y are scaled by factors
and
respectively, then the sum of these scaled inputs will
give the sum of the individual scaled outputs:
(a)
(b)
Figure 3.13
Superposition Principle with Linear Scaling
Given Figure 3.13 for a linear system, Figure 3.14 (Superposition Principle with Linear
Scaling) holds as well.
Figure 3.14:
46
Example 3.1
Consider the system
H1
in which
H1 (f (t)) = tf (t)
for all signals
f.
Given any two signals
f, g
and scalars
(3.7)
a, b
H1 (af (t) + bg (t)) = t (af (t) + bg (t)) = atf (t) + btg (t) = aH1 (f (t)) + bH1 (g (t))
Thus,
H1
is a linear system.
Consider the system
H2
in which
for all real
t.
(3.8)
Example 3.2
H2 (f (t)) = (f (t))
for all signals
f.
(3.9)
Because
H2 (2t) = 4t2 6= 2t2 = 2H2 (t)

for nonzero
t, H2
(3.10)
is not a linear system.
3.3.2.2 Time Invariant Systems

A time-invariant system has the property that a certain input will always give the same output (up to
timing), without regard to when the input was applied to the system.
Time-Invariant Systems
(a)
(b)
Figure 3.15(a) shows an input at time t while Figure 3.15(b) shows the same input
t0 seconds later. In a time-invariant system both outputs would be identical except that the one in
Figure 3.15(b) would be delayed by t0 .
Figure 3.15:
x (t) and x (t t0 ) are passed

x (t) and x (t t0 ) produce
x (t t0 ) is shifted by a time t0 .
In this gure,
through the system TI. Because the system TI is time-
invariant, the inputs
the same output. The only dierence is that the output
due to
Whether a system is time-invariant or time-varying can be seen in the dierential equation (or dierence
equation) describing it.
Time-invariant systems are modeled with constant coecient equations.

not
A constant coecient dierential (or dierence) equation means that the parameters of the system are
changing over time and an input now will give the same result as the same input later.
47
Example 3.3
Consider the system
H1
in which
H1 (f (t)) = tf (t)
for all signals
f.
(3.11)
Because
ST (H1 (f (t))) = ST (tf (t)) = (t T ) f (t T ) 6= tf (t T ) = H1 (f (t T )) = H1 (ST (f (t)))

for nonzero
T , H1
(3.12)
is not a time invariant system.
Example 3.4
Consider the system
H2
in which
H2 (f (t)) = (f (t))
for all signals
f.
For all real
and signals
ST (H2 (f (t))) = ST f (t)

for all real
t.
Thus,
H2
(3.13)
f,
2
= (f (t T )) = H2 (f (t T )) = H2 (ST (f (t)))
(3.14)
is a time invariant system.
3.3.2.3 Linear Time Invariant Systems

Certain systems are both linear and time-invariant, and are thus referred to as LTI systems.
Linear Time-Invariant Systems
(a)
(b)
This is a combination of the two cases above. Since the input to Figure 3.16(b) is a scaled,
time-shifted version of the input in Figure 3.16(a), so is the output.
Figure 3.16:
As LTI systems are a subset of linear systems, they obey the principle of superposition. In the gure
below, we see the eect of applying time-invariance to the superposition denition in the linear systems
section above.
48
(a)
(b)
Figure 3.17
Superposition in Linear Time-Invariant Systems
Figure 3.18:
The principle of superposition applied to LTI systems
3.3.2.3.1 LTI Systems in Series

If two or more LTI systems are in series with each other, their order can be interchanged without aecting
the overall output of the system. Systems in series are also called cascaded systems.
49
Cascaded LTI Systems
(a)
(b)
Figure 3.19:
eect.
The order of cascaded LTI systems can be interchanged without changing the overall
3.3.2.3.2 LTI Systems in Parallel

If two or more LTI systems are in parallel with one another, an equivalent system is one that is dened as
the sum of these individual systems.
Parallel LTI Systems
(a)
Figure 3.20:
(b)
Parallel systems can be condensed into the sum of systems.
50
Example 3.5
Consider the system
H3
in which
H3 (f (t)) = 2f (t)
for all signals
f.
f, g
and scalars
(3.15)
a, b
H3 (af (t) + bg (t)) = 2 (af (t) + bg (t)) = a2f (t) + b2g (t) = aH3 (f (t)) + bH3 (g (t))
for all real
t.
Thus,
H3
is a linear system. For all real
and signals
f,
ST (H3 (f (t))) = ST (2f (t)) = 2f (t T ) = H3 (f (t T )) = H3 (ST (f (t)))

for all real t. Thus,
H3
is a time invariant system. Therefore,
H3
(3.16)
(3.17)
is a linear time invariant system.
Example 3.6
As has been previously shown, each of the following systems are not linear or not time invariant.
H1 (f (t)) = tf (t)
H2 (f (t)) = (f (t))
(3.18)
(3.19)
Thus, they are not linear time invariant systems.
3.3.3 Linear Time Invariant Demonstration
Figure 3.21: Interact(when online) with the Mathematica CDF above demonstrating Linear Time
Invariant systems. To download, right click and save le as .cdf.
3.3.4 LTI Systems Summary

Two very important and useful properties of systems have just been described in detail. The rst of these,
linearity, allows us the knowledge that a sum of input signals produces an output signal that is the summed
original output signals and that a scaled input signal produces an output signal scaled from the original
output signal. The second of these, time invariance, ensures that time shifts commute with application of
the system. In other words, the output signal for a time shifted input is the same as the output signal for the
original input signal, except for an identical shift in time. Systems that demonstrate both linearity and time
invariance, which are given the acronym LTI systems, are particularly simple to study as these properties
allow us to leverage some of the most powerful tools in signal processing.
Chapter 4
Time Domain Analysis of Continuous
Time Systems
4.1 Continuous Time Systems
4.1.1 Introduction
As you already now know, a continuous time system operates on a continuous time signal input and produces
a continuous time signal output. There are numerous examples of useful continuous time systems in signal
processing as they essentially describe the world around us.
The class of continuous time systems that
are both linear and time invariant, known as continuous time LTI systems, is of particular interest as the
properties of linearity and time invariance together allow the use of some of the most important and powerful
tools in signal processing.
4.1.2 Continuous Time Systems

4.1.2.1 Linearity and Time Invariance
A system
is said to be linear if it satises two important conditions. The rst, additivity, states for every
pair of signals
signal
x, y
that
and scalar
H (x + y) = H (x) + H (y).
H (ax) = aH (x). It
we have
The second, homogeneity of degree one, states for every

is clear that these conditions can be combined together
into a single condition for linearity. Thus, a system is said to be linear if for every signals
a, b
x, y
and scalars
we have that
H (ax + by) = aH (x) + bH (y) .
(4.1)
Linearity is a particularly important property of systems as it allows us to leverage the powerful tools of
linear algebra, such as bases, eigenvectors, and eigenvalues, in their study.
A system
is said to be time invariant if a time shift of an input produces the corresponding shifted
output. In other, more precise words, the system
T R.
commutes with the time shift operator
ST
for every
That is,
ST H = HST .
(4.2)
Time invariance is desirable because it eases computation while mirroring our intuition that, all else equal,
physical systems should react the same to identical inputs at dierent times.
When a system exhibits both of these important properties it allows for a more straigtforward analysis
than would otherwise be possible.
1 This
As will be explained and proven in subsequent modules, computation

51
CHAPTER 4. TIME DOMAIN ANALYSIS OF CONTINUOUS TIME

SYSTEMS
52
of the system output for a given input becomes a simple matter of convolving the input with the system's
impulse response signal. Also proven later, the fact that complex exponential are eigenvectors of linear time
invariant systems will enable the use of frequency domain tools such as the various Fouier transforms and
associated transfer functions, to describe the behavior of linear time invariant systems.
Example 4.1
Consider the system
in which
H (f (t)) = 2f (t)
for all signals
f.
f, g
and scalars
(4.3)
a, b
H (af (t) + bg (t)) = 2 (af (t) + bg (t)) = a2f (t) + b2g (t) = aH (f (t)) + bH (g (t))
for all real
t.
Thus,
is a linear system. For all real
and signals
f,
ST (H (f (t))) = ST (2f (t)) = 2f (t T ) = H (f (t T )) = H (ST (f (t)))

for all real
t.
Thus,
is a time invariant system. Therefore,
(4.4)
(4.5)
is a linear time invariant system.
4.1.3 Continuous Time Systems Summary

Many useful continuous time systems will be encountered in a study of signals and systems. This course
is most interested in those that demonstrate both the linearity property and the time invariance property,
which together enable the use of some of the most powerful tools of signal processing. It is often useful to
describe them in terms of rates of change through linear constant coecient ordinary dierential equations.
4.2 Continuous Time Impulse Response
4.2.1 Introduction
The output of an LTI system is completely determined by the input and the system's response to a unit
impulse.
System Output
We can determine the system's output, y(t), if we know the system's impulse response,
h(t), and the input, f(t).
Figure 4.1:
The output for a unit impulse input is called the impulse response.
2 This
53
Figure 4.2
4.2.1.1 Example Approximate Impulses

1. Hammer blow to a structure
2. Hand clap or gun blast in a room
3. Air gun blast underwater
4.2.2 LTI Systems and Impulse Responses

4.2.2.1 Finding System Outputs
By the sifting property of impulses, any signal can be decomposed in terms of an integral of shifted, scaled
impulses.
f ( ) (t ) d
f (t) =
(t )
peaks up where
t = .
(4.6)

SYSTEMS
54
Figure 4.3
Since we know the response of the system to an impulse and any signal can be decomposed into impulses,
all we need to do to nd the response of the system to any signal is to decompose the signal into impulses,
calculate the system's output for every impulse and add the outputs back together.
known as
Integral.
Convolution.
Since we are in
This is the process
Continuous Time, this is the Continuous Time Convolution
4.2.2.2 Finding Impulse Responses

Theory:
a. Solve the system's dierential equation for y(t) with f (t) = (t)
b. Use the Laplace transform
Practice:
a. Apply an impulse-like input signal to the system and measure the output
b. Use Fourier methods
We will assume that h(t) is given for now.
The goal now is to compute the output y(t) given the impulse response h(t) and the input f(t).
55
Figure 4.4
4.2.3 Impulse Response Summary

When a system is "shocked" by a delta function, it produces an output known as its impulse response. For
an LTI system, the impulse response completely determines the output of the system given any arbitrary
input. The output can be found using continuous time convolution.
4.3 Continuous-Time Convolution
4.3.1 Introduction
Convolution, one of the most important concepts in electrical engineering, can be used to determine the
output a system produces for a given input signal.
It can be shown that a linear time invariant system
is completely characterized by its impulse response.
The sifting property of the continuous time impulse
function tells us that the input signal to a system can be represented as an integral of scaled and shifted
impulses and, therefore, as the limit of a sum of scaled and shifted approximate unit impulses. Thus, by
linearity, it would seem reasonable to compute of the output signal as the limit of a sum of scaled and
shifted unit impulse responses and, therefore, as the integral of a scaled and shifted impulse response. That
is exactly what the operation of convolution accomplishes. Hence, convolution can be used to determine a
linear time invariant system's output from knowledge of the input and the impulse response.
4.3.2 Convolution and Circular Convolution

4.3.2.1 Convolution
4.3.2.1.1 Operation Denition
Continuous time convolution is an operation on two continuous time signals dened by the integral
(f g) (t) =
f ( ) g (t ) d
(4.7)
for all signals
f, g
dened on
R.
It is important to note that the operation of convolution is commutative,
meaning that
f g =gf
3 This
(4.8)

SYSTEMS
56
for all signals
f, g
dened on
R.
Thus, the convolution operation could have been just as easily stated using
the equivalent denition
Z
(f g) (t) =
f (t ) g ( ) d
(4.9)
for all signals
f, g
dened on
R.
Convolution has several other important properties not listed here but
explained and derived in a later module.
4.3.2.1.2 Denition Motivation

The above operation denition has been chosen to be particularly useful in the study of linear time invariant
systems. In order to see this, consider a linear time invariant system
a system input signal
with unit impulse response
we would like to compute the system output signal
H (x).
h.
Given
First, we note that the
input can be expressed as the convolution
x ( ) (t ) d
x (t) =
(4.10)
by the sifting property of the unit impulse function. Writing this integral as the limit of a summation,
x (t) = lim
x (n) (t n)
(4.11)
where
(t) = {
approximates the properties of
(t).
1/
0t<
otherwise
(4.12)
By linearity
Hx (t) = lim
x (n) H (t n)
(4.13)
which evaluated as an integral gives
x ( ) H (t ) d.
Hx (t) =
(4.14)
Since
H (t )
is the shifted unit impulse response
h (t ),
x ( ) h (t ) d = (x h) (t) .
Hx (t) =
(4.15)
Hence, convolution has been dened such that the output of a linear time invariant system is given by the
convolution of the system input with the system unit impulse response.
4.3.2.1.3 Graphical Intuition

It is often helpful to be able to visualize the computation of a convolution in terms of graphical processes.
Consider the convolution of two functions
f, g
given by
(f g) (t) =
f ( ) g (t ) d =
f (t ) g ( ) d.
(4.16)
The rst step in graphically understanding the operation of convolution is to plot each of the functions.
Next, one of the functions must be selected, and its plot reected across the
=0
axis. For each real t, that
57
same function must be shifted left by
t.
The product of the two resulting plots is then constructed. Finally,
the area under the resulting curve is computed.
Example 4.2
Recall that the impulse response for the capacitor voltage in a series RC circuit is given by
h (t) =
1 t/RC
e
u (t) ,
RC
(4.17)
and consider the response to the input voltage
x (t) = u (t) .
(4.18)
We know that the output for this input voltage is given by the convolution of the impulse response
with the input signal
y (t) = x (t) h (t) .
(4.19)
We would like to compute this operation by beginning in a way that minimizes the algebraic
complexity of the expression.
Thus, since
x (t) = u (t)
is the simpler of the two signals, it is
desirable to select it for time reversal and shifting. Thus, we would like to compute
y (t) =
1 /RC
e
u ( ) u (t ) d.
RC
(4.20)
The step functions can be used to further simplify this integral by narrowing the region of integration to the nonzero region of the integrand. Therefore,
max{0,t}
Z
y (t) =
0
1 /RC
e
d.
RC
(4.21)
Hence, the output is
y (t) = {
t0
t/RC
t>0
1e
(4.22)
which can also be written as

y (t) = 1 et/RC u (t) .
(4.23)
4.3.3 Online Resources

The following pages have interactive Java applets that demonstrate several aspects of continuous-time convolution.
4
Joy of Convolution (Johns Hopkins University)

5
Step-by-Step Convolution (Rice University)
4 http://www.jhu.edu/signals/convolve/index.html
5 http://www.ece.rice.edu/dsp/courses/elec301/demos/applets/Convo1/

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58
4.3.4 Convolution Demonstration
Interact (when online) with a Mathematica CDF demonstrating Convolution. To Download,

right-click and save target as .cdf.
Figure 4.5:
4.3.5 Convolution Summary

output signal of a linear time invariant system for a given input signal with knowledge of the system's unit
impulse response. The operation of continuous time convolution is dened such that it performs this function
for innite length continuous time signals and systems. The operation of continuous time circular convolution
is dened such that it performs this function for nite length and periodic continuous time signals. In each
case, the output of the system is the convolution or circular convolution of the input signal with the unit
impulse response.
4.4 Properties of Continuous Time Convolution
4.4.1 Introduction
We have already shown the important role that continuous time convolution plays in signal processing. This
section provides discussion and proof of some of the important properties of continuous time convolution.
Analogous properties can be shown for continuous time circular convolution with trivial modication of the
proofs provided except where explicitly noted otherwise.
6 This

59
4.4.2 Continuous Time Convolution Properties

4.4.2.1 Associativity
The operation of convolution is associative. That is, for all continuous time signals
f1 , f2 , f3
the following
relationship holds.
f1 (f2 f3 ) = (f1 f2 ) f3
(4.24)
In order to show this, note that
(f1 (f2 f3 )) (t)
R R
= f1 (1 ) f2 (2 ) f3 ((t 1 ) 2 ) d2 d1
R R
= f1 (1 ) f2 ((1 + 2 ) 1 ) f3 (t (1 + 2 )) d2 d1
R R
= f1 (1 ) f2 (3 1 ) f3 (t 3 ) d1 d3
(4.25)
= ((f1 f2 ) f3 ) (t)
proving the relationship as desired through the substitution
3 = 1 + 2 .
4.4.2.2 Commutativity
The operation of convolution is commutative. That is, for all continuous time signals
f1 , f2
the following
relationship holds.
f1 f2 = f2 f1
(4.26)
(f1 f2 ) (t)
f1 (1 ) f2 (t 1 ) d1
f1 (t 2 ) f2 (2 ) d2
(4.27)
= (f2 f1 ) (t)
2 = t 1 .
4.4.2.3 Distributivity
The operation of convolution is distributive over the operation of addition. That is, for all continuous time
signals
f1 , f2 , f3
the following relationship holds.
f1 (f2 + f3 ) = f1 f2 + f1 f3
(4.28)
(f1 (f2 + f3 )) (t)
f1 ( ) (f2 (t ) + f3 (t )) d
R
= f1 ( ) f2 (t ) d + f1 ( ) f3 (t ) d
=
= (f1 f2 + f1 f3 ) (t)
proving the relationship as desired.
(4.29)

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60
4.4.2.4 Multilinearity
The operation of convolution is linear in each of the two function variables.
Additivity in each variable
results from distributivity of convolution over addition. Homogenity of order one in each variable results
from the fact that for all continuous time signals
f1 , f2
and scalars
a (f1 f2 ) = (af1 ) f2 = f1 (af2 )
(4.30)
(a (f1 f2 )) (t)
R
= a f1 ( ) f2 (t ) d
R
= (af1 ( )) f2 (t ) d
= ((af1 ) f2 ) (t)
=
(4.31)
f1 ( ) (af2 (t )) d
= (f1 (af2 )) (t)

4.4.2.5 Conjugation
The operation of convolution has the following property for all continuous time signals
f1 , f2 .
f1 f2 = f1 f2
(4.32)

f1 f2 (t)
f1 ( ) f2 (t ) d
f1 ( ) f2 (t )d
(4.33)
f1 ( ) f2 (t ) d

= f1 f2 (t)
4.4.2.6 Time Shift

f1 , f2
where
ST
is
the time shift operator.
ST (f1 f2 ) = (ST f1 ) f2 = f1 (ST f2 )
(4.34)
ST (f1 f2 ) (t)
f ( ) f1 ((t T ) ) d
2
R
= f2
( ) ST f1 (t ) d
= ((ST f1 ) f2 ) (t)
=
f ( ) f2 ((t T ) ) d
1
R
= f1
( ) ST f2 (t ) d
= f1 (ST f2 ) (t)
(4.35)
61
4.4.2.7 Dierentiation
d
(f1 f2 ) (t) =
dt
df1
f2 (t) =
dt

f1
df2
dt
f1 , f2 .

(t)
(4.36)
d
dt
(f1 f2 ) (t)
d
f2 ( ) dt
f1 (t ) d

df1
= dt f2 (t)
(4.37)
d
f ( ) dt
f2 (t ) d
1

df2
= f1 dt (t)
4.4.2.8 Impulse Convolution

where
is the Dirac
delta funciton.
f =f
(4.38)
(f ) (t)
f ( ) (t ) d
R
= f (t) (t ) d
(4.39)
= f (t)
4.4.2.9 Width
Duration (f )
gives the duration of a signal
f1 , f2
Duration (f1 f2 ) = Duration (f1 ) + Duration (f2 )

. In order to show this informally, note that
f1 ( ) f2 (t )
see that such a
where
f.
(f1 f2 ) (t)
is nonzero for all
(4.40)
for which there is a
such that
is nonzero. When viewing one function as reversed and sliding past the other, it is easy to
exists for all
t on an interval of length Duration (f1 ) + Duration (f2 ).
Note that this is not
always true of circular convolution of nite length and periodic signals as there is then a maximum possible
duration within a period.
4.4.3 Convolution Properties Summary

As can be seen the operation of continuous time convolution has several important properties that have
been listed and proven in this module.
With slight modications to proofs, most of these also extend to
continuous time circular convolution as well and the cases in which exceptions occur have been noted above.
These identities will be useful to keep in mind as the reader continues to study signals and systems.

SYSTEMS
62
4.5 Causality and Stability of Continuous-Time Linear TimeInvariant Systems

7
4.5.1 Introduction
We have previously dened the system properties of causality and bounded-input bounded-output (BIBO)
stability.
We have also determined that a linear time-invariant (LTI) system is completely determined
by its impulse response
y (t) = x (t) h (t).
h (t):
the output
y (t)
for an arbitrary input
x (t)
is obtained via convolution as
It should not be surprising then that one can determine whether an LTI system is causal
or BIBO stable simply by inspecting its impulse response
h (t).
4.5.2 Causality
Recall that a system is causal if its output
t t0 .
y (t0 )
t0
at time
depends only on the input
x (t)
for values of
Consider then the characterization of the system by the convolution integral:
y (t)
We replace the time variable
= x (t) h (t) =
by a xed value
y (t0 )
x ( ) h (t ) d.
(4.41)
t0 :
For this input to depend only on values of time
x ( ) h (t0 ) d.
t0
(where
(4.42)
represents a second time variable), we
must have its contribution to the integral being nulled out by requiring the value of the impulse response
h (t0 ) = 0
> t0 . By making a change of variables t = t0 (so that = t0 t), this

h (t) = 0 for t0 t > t0 ; this means that we require h (t) = 0 for t < 0.
that an LTI system with impulse response h (t) is causal if and only if h (t) = 0 for all
for values of
means that we require

Thus, we obtain
t < 0.
4.5.3 BIBO Stability

x (t) is bounded (that is, if there exists M <
|x (t) | < M for all t) then we have that the system output is also bounded (that is, if there exists
N < such that |y (t) | < N for all t). As before, consider then the characterization of the system by the
Recall that a system is BIBO stable if whenever the input

such that
convolution integral:
y (t)
= x (t) h (t) =
x ( ) h (t ) d.
(4.43)
We apply absolute value to both sides and use the straightforward bound on the absolute value of an integral:
R
R

|y (t)| = x ( ) h (t ) d |x ( ) h (t ) |d
R
R
= |x ( ) ||h (t ) |d M |h (t ) |d
R
= M |h (t ) |d,
x (t). Thus, if we have that the integral has a nite

|y (t) |, which implies BIBO stability. R
an LTI system with impulse response h (t) is BIBO stable if and only if |h (t) |
where the next-to-last step uses the boundedness of
value, then we have established a nite upper bound on

Thus, we obtain that
is nite.
7 This
(4.44)
63
4.5.4 Summary
The derivations above show that it is signicantly easier to verify whether a system is causal and/or BIBO
stable when it is linear and time-invariant.
evaluations of the system's impulse response
In such a case, we simply need to perform straightforward
h (t).
64

SYSTEMS
Chapter 5
Introduction to Fourier Analysis
5.1 Introduction to Fourier Analysis
5.1.1 Fourier's Daring Leap

Fourier postulated around 1807 that any periodic signal (equivalently nite length signal) can be built up
as an innite linear combination of harmonic sinusoidal waves.
Given the collection
2
B = {ej T
any nite-energy function
x (t)
nt
}n=
(5.1)
can be approximated arbitrarily closely by
x (t) =
C n ej T
nt
(5.2)
n=
Now, The issue of exact convergence did bring Fourier
much criticism from the French Academy of Science
(Laplace, Lagrange, Monge and LaCroix comprised the review committee) for several years after its presentation on 1807. It was not resolved for almost a century, and its resolution is interesting and important to
understand from a practical viewpoint.
Fourier analysis is fundamental to understanding the behavior of signals and systems. This is a result of
3
the fact that sinusoids are Eigenfunctions (Section 5.3) of linear, time-invariant (LTI)
systems. This is to
say that if we pass any particular sinusoid through a LTI system, we get a scaled version of that same sinusoid
on the output. Then, since Fourier analysis allows us to redene the signals in terms of a combination of
sinusoids, all we need to do is determine how any given system acts on each possible sinusoid (its
function) and we have a complete understanding of the system.
transfer
Furthermore, since we are able to dene
the passage of sinusoids through a system as the multiplication of that sinusoid by its scaling factor, we can
convert the passage of any signal through a system from convolution (Section 4.3) (in time) to multiplication
(in frequency). These ideas are what give Fourier analysis its power.
Now, after hopefully having sold you on the value of this method of analysis, we must examine exactly
what we mean by Fourier analysis. The four Fourier transforms that comprise this analysis are the Fourier
Series (Section 5.4), Continuous-Time Fourier Transform (Section 6.2), Discrete-Time Fourier Transform
(Section 9.3) and Discrete Fourier Transform (Section 10.1). All of these transforms act essentially the same
way, by converting a signal in time to an equivalent signal in frequency (sinusoids).
on the nature of a specic signal
i.e.
However, depending
whether it is nite- or innite-length and whether it is discrete- or
continuous-time) there is an appropriate transform to convert the signal into the frequency domain.

2 http://www-groups.dcs.st-and.ac.uk/history/Mathematicians/Fourier.html
3 "Continuous Time Systems" <http://legacy.cnx.org/content/m10855/latest/>

65
CHAPTER 5. INTRODUCTION TO FOURIER ANALYSIS
66
5.2 Continuous Time Periodic Signals
5.2.1 Introduction
This module describes the type of signals acted on by the Continuous Time Fourier Series.
5.2.2 Periodic Signals

When a function repeats itself exactly after some given period, or cycle, we say it's
function can be mathematically dened as:
periodic.
periodic
f (t) = f (t + mT ) m Z
T >0
where
represents the
(5.3)
fundamental period of the signal, which is the smallest positive value of T
for the signal to repeat. Because of this, you may also see a signal referred to as a T-periodic signal. Any
periodic with period T.

periodic functions (with period T ) two dierent ways:
as functions on all of R
function that satises this equation is said to be

We can think of
1.
Figure 5.1:
Continuous time periodic function over all of R where f (t0 ) = f (t0 + T )
2. or, we can cut out all of the redundancy, and think of them as functions on an interval
more generally,
[a, a + T ]).
[0, T ]
(or,
If we know the signal is T-periodic then all the information of the signal is
captured by the above interval.
Figure 5.2:
An
Remove the redundancy of the period function so that f (t) is undened outside [0, T ].
aperiodic CT function f (t), on the other hand, does not repeat for any T
R; i.e.
such that this equation (5.3) holds.
4 This
there exists no
67
5.2.3 Demonstration
Here's an example demonstrating a
periodic
sinusoidal signal with various frequencies, amplitudes and
phase delays:
Interact (when online) with a Mathematica CDF demonstrating a Periodic Sinusoidal Signal
with various frequencies, amplitudes, and phase delays. To download, right click and save le as .cdf.
Figure 5.3:
To learn the full concept behind periodicity, see the video below.
Khan Lecture on Periodic Signals

This media object is a Flash object. Please view or download it at
<http://www.youtube.com/v/tJW_a6JeXD8&rel=0&color1=0xb1b1b1&color2=0xd0d0d0&hl=en_US&feature=player_em
Figure 5.4:
video from Khan Academy
5.2.4 Conclusion
A periodic signal is completely dened by its values in one period, such as the interval [0,T].
68
5.3 Eigenfunctions of Continuous-Time LTI Systems
5.3.1 Introduction
Prior to reading this module, the reader should already have some experience with linear algebra and should
specically be familiar with the eigenvectors and eigenvalues of square matrices.
A linear time invariant
system is a linear operator dened on a function space that commutes with every time shift operator on
that function space. Thus, we can also consider the eigenvector functions, or eigenfunctions, of a system.
The concept of an eigenfunction is closely tied to the concept of an eigenvector in linear algebra. Eigen is
german for "self": the eigenfunction of a system is a function that, when fed to the system, produces in
the output a copy of the function, perhaps rescaled.
if
H (f ) = f
for some scalar
More concretely,
is an eigenfunction for a system
It is particularly easy to calculate the output of a system when an
eigenfunction is the input as the output is simply the eigenfunction scaled by the associated eigenvalue. As
will be shown, continuous time complex exponentials serve as eigenfunctions of linear time invariant systems
operating on continuous time signals.
5.3.2 Eigenfunctions of LTI Systems

Consider a linear time invariant system
with impulse response
continuous time signals. Recall that the output
H (x (t))
h operating on some space of innite length

x (t) is given by the
of the system for a given input
continuous time convolution of the impulse response with the input
h ( ) x (t ) d.
H (x (t)) =
(5.4)
Now consider the input
x (t) = est
where
s C.
H (est )
Computing the output for this input,
h ( ) es(t ) d
h ( ) est es d
R
est
h ( ) e
(5.5)
d.
Thus,

H est = s est
(5.6)
where
h ( ) es d
s =
(5.7)
is the eigenvalue corresponding to the eigenfunction
est .
There are some additional points that should be mentioned.

eigenfunctions of a linear time invariant system not described by
discussion has been somewhat formally loose as
est
Note that, there still may be additional
est
for some
s C.
Furthermore, the above
may or may not belong to the space on which the system
operates. However, for our purposes, complex exponentials will be accepted as eigenfunctions of linear time
invariant systems. A similar argument using continuous time circular convolution would also hold for spaces
nite length signals.
5.3.3 Eigenfunction of LTI Systems Summary

As has been shown, continuous time complex exponential are eigenfunctions of linear time invariant systems
operating on continuous time signals. Thus, it is particularly simple to calculate the output of a linear time
5 This

69
invariant system for a complex exponential input as the result is a complex exponential output scaled by the
associated eigenvalue. Consequently, representations of continuous time signals in terms of continuous time
complex exponentials provide an advantage when studying signals. As will be explained later, this is what
is accomplished by the continuous time Fourier transform and continuous time Fourier series, which apply
to aperiodic and periodic signals respectively.
5.4 Continuous Time Fourier Series (CTFS)
5.4.1 Introduction
In this module, we will derive an expansion for continuous-time, periodic functions, and in doing so, derive
the
Continuous Time Fourier Series (CTFS).
Since complex exponentials
are eigenfunctions of linear time-invariant (LTI) systems (Section 5.3), cal-
culating the output of an LTI system
H (s) C
given
est
as an input amounts to simple multiplication, where
is the eigenvalue corresponding to s. As shown in the gure, a simple exponential input would
yield the output
y (t) = H (s) est
Figure 5.5:
(5.8)
Simple LTI system.
Using this and the fact that
is linear, calculating
y (t) for combinations of complex exponentials is also
straightforward.
c1 es1 t + c2 es2 t c1 H (s1 ) es1 t + c2 H (s2 ) es2 t

X
cn esn t
n
The action of
cn H (sn ) esn t
on an input such as those in the two equations above is easy to explain.
pendently scales each exponential component es
nt
we can write a function
f (t)
by a dierent complex number
H (sn ) C.
inde-
as a combination of complex exponentials it allows us to easily calculate the
output of a system.
6 This content
7 "Continuous
As such, if
is available online at <http://legacy.cnx.org/content/m47348/1.2/>.

Time Complex Exponential" <http://legacy.cnx.org/content/m10060/latest/>
70
5.4.2 Fourier Series Synthesis

8
Joseph Fourier
demonstrated that an arbitrary periodic function
of harmonic complex sinusoids
x (t) =
x (t) can be written as a linear combination
cn ej2f0 nt
(5.9)
n=
1
T is the fundamental frequency. For almost all x (t) of practical interest, there exists cn to make
2
(5.9) true. If x (t) is nite energy ( x (t) L [0, T ]), then the equality in (5.9) holds in the sense of energy
where
f0 =
convergence; if
x (t)
is continuous, then (5.9) holds pointwise. Also, if
x (t)
meets some mild conditions (the
Dirichlet conditions), then (5.9) holds pointwise everywhere except at points of discontinuity.
cn - called the Fourier coecients - tell us "how much" of the sinusoid ej2f0 nt is in x (t). The formula
x (t) as a sum of complex exponentials, each of which is easily processed by an LTI system (since it
The
shows
is an eigenfunction of
ej2f0 nt , n Z
every LTI system).
Mathematically, it tells us that the set of complex exponentials
form a basis for the space of T-periodic continuous time functions.
Example 5.1
We know from Euler's formula that
cos (2f t) + sin (2f t) =
1j j2f t
2 e
1+j j2f t
.
2 e
71
5.4.3 Synthesis with Sinusoids Demonstration
Figure 5.6: Interact(when online) with a Mathematica CDF demonstrating sinusoid synthesis. To
download, right click and save as .cdf.
Guitar Oscillations on an iPhone

<http://www.youtube.com/v/TKF6nFzpHBU?version=3&hl=en_US>
Figure 5.7
72
5.4.4 Fourier Series Analysis

Finding the coecients of the Fourier series expansion involves some algebraic manipulation of the synthesis
formula. First of all we will multiply both sides of the equation by
f (t) e(j2f0 kt) =
e(j2f0 kt) ,
where
k Z.
cn ej2f0 nt e(j2f0 kt)
(5.10)
n=
Now integrate both sides over a given period,
T:
T
(j2f0 kt)
f (t) e
Z
dt =
cn ej2f0 nt e(j2f0 kt) dt
(5.11)
n=
On the right-hand side we can switch the summation and integral and factor the constant out of the integral.
f (t) e
(j2f0 kt)
dt =
cn
ej2f0 (nk)t dt
(5.12)
n=
RT
ej2f0 (nk)t dt,

0
on the right-hand side of the above equation. For this integral we will need to consider two cases: n = k and
Now that we have made this seemingly more complicated, let us focus on just the integral,
n 6= k .
For
n=k
we will have:
ej2f0 (nk)t dt = T , n = k
(5.13)
0
For
n 6= k ,
Z
we will have:
ej2f0 (nk)t dt =
0
But
sin (2f0 (n k) t) dt , n 6= k
cos (2f0 (n k) t)
Z
cos (2f0 (n k) t) dt + j
(5.14)
has an integer number of periods,
n k,
between
and
T.
Imagine a graph of the
cosine; because it has an integer number of periods, there are equal areas above and below the x-axis of the
graph. This statement holds true for
sin (2f0 (n k) t)
Z
as well. What this means is
cos (2f0 (n k) t) dt = 0
(5.15)
0
which also holds for the integral involving the sine function. Therefore, we conclude the following about
our integral of interest:
T
ej2f0 (nk)t dt =
0
if
n=k
(5.16)
otherwise
We plug in this result in (5.12) to see if we can nish nding an equation for our Fourier coecients. Using
the facts that we have just proven above, we can see that the only time (5.12) will have a nonzero result is
when
and
are equal:
f (t) e(j2f0 nt) dt = T cn , n = k
(5.17)
0
Finally, we have our general equation for the Fourier coecients:
1
cn =
T
f (t) e(j2f0 nt) dt
(5.18)
73
Example 5.2
Consider the square wave function given by
x (t) = {
on the unit interval
1/2
t 1/2
1/2
t > 1/2
(5.19)
t Z [0, 1).
ck
R1
x (t) ej2kt dt
0
R1
1 j2kt
dt 1/2 12 ej2kt dt
2e
j (1+ejk )
=
2k
=
R 1/2
0
(5.20)
Thus, the Fourier coecients of this function found using the Fourier series analysis formula are
ck = {
j/k
kodd
keven
(5.21)
5.4.5 Fourier Series Summary

Because complex exponentials are eigenfunctions of LTI systems, it is often useful to represent signals using
a set of complex exponentials as a basis. The continuous time Fourier series synthesis formula expresses a
continuous time, periodic function as the sum of continuous time, discrete frequency complex exponentials.
f (t) =
cn ej2f0 nt
(5.22)
n=
The continuous time Fourier series analysis formula gives the coecients of the Fourier series expansion.
cn =
In both of these equations
f0 =
1
T
f (t) e(j2f0 nt) dt
1
T is the fundamental frequency.
(5.23)
74
Chapter 6
Continuous Time Fourier Transform
(CTFT)
6.1 Continuous Time Aperiodic Signals
6.1.1 Introduction
This
module
describes
the
type
of
signals
acted
on
by
the
Continuous
Time
Fourier
Transform.
6.1.2 Periodic and Aperiodic Signals

periodic.
f (t) = f (t + mT ) m Z
where
T >0
represents the
periodic
(6.1)
fundamental period of the signal, which is the smallest positive value of T
for the signal to repeat. Because of this, you may also see a signal referred to as a T-periodic signal. Any
periodic with period T.

aperiodic CT function f (t) does not repeat for any T R; i.e.
function that satises this equation is said to be

An
there exists no
such that this
equation (6.1) holds.

Suppose we have such an aperiodic function
fT o (t) , where f (t) is repeated every T0 seconds.
f (t)
f (t) called
T0 , we obtain a precise model of
. We can construct a periodic extension of
If we take the limit as
an aperiodic signal for which all rules that govern periodic signals can be applied, including Fourier Analysis
(with an important modication). For more detail on this distinction, see the module on the
Time Fourier Transform.

1 This

75
Continuous
CHAPTER 6. CONTINUOUS TIME FOURIER TRANSFORM (CTFT)
76
6.1.3 Aperiodic Signal Demonstration
Interact (when online) with a Mathematica CDF demonstrating Periodic versus Aperiodic
Signals.To download, right-click and save as .cdf.
Figure 6.1:
6.1.4 Conclusion
Any aperiodic signal can be dened by an innite sum of periodic functions, a useful denition that makes
it possible to use Fourier Analysis on it by assuming all frequencies are present in the signal.
6.2 Continuous Time Fourier Transform (CTFT)
6.2.1 Introduction
In this module, we will derive an expansion for any arbitrary continuous-time function, and in doing so,
derive the
Continuous Time Fourier Transform (CTFT).
are eigenfunctions of linear time-invariant (LTI) systems (Section 5.3), cal-
culating the output of an LTI system
H (s) C
given
est
is the eigenvalue corresponding to s. As shown in the gure, a simple exponential input would
yield the output
y (t) = H (s) est

y (t) for combinations of complex exponentials is also
straightforward.
c1 es1 t + c2 es2 t c1 H (s1 ) es1 t + c2 H (s2 ) es2 t

2 This content
3 "Continuous
(6.2)
is available online at <http://legacy.cnx.org/content/m47319/1.5/>.

77
cn esn t
n
The action of
cn H (sn ) esn t
pendently scales each exponential component es
nt
f (t)
H (sn ) C.
inde-
As such, if
output of a system.
Now, we will look to use the power of complex exponentials to see how we may represent arbitrary signals
in terms of a set of simpler functions by superposition of a number of complex exponentials. Below we will
present the
Continuous-Time Fourier Transform
(CTFT), commonly referred to as just the Fourier
Transform (FT).
6.2.2 Fourier Synthesis

4
Joseph Fourier
demonstrated that an arbitrary periodic signal
of harmonic complex sinusoids
s (t) =
cn ej T
s (t)
can be written as a linear combination
nt
(6.3)
n=
where
T is the fundamental
s (t) is nite energy,
true. If
period. For almost all
s (t)
of practical interest, there exists
cn
to make (6.3)
then the equality in (6.3) holds in the sense of energy convergence; if
continuous, then (6.3) holds pointwise. Also, if
s (t)
s (t)
is
meets some mild conditions (the Dirichlet conditions),
then (6.3) holds pointwise everywhere except at points of discontinuity.

2
cn - called the Fourier coecients - tell us "how much" of the sinusoid ej T nt is in s (t). The formula
s (t) as a sum of complex exponentials, each of which is easily processed by an LTI system (since it
The
shows
n 2
o
ej T nt , n Z
every LTI system).
form a basis for the space of
T -periodic
continuous time functions. Because the CTFT
deals with nonperiodic signals, we must nd a way to include all real frequencies in the general equations.
For the CTFT we simply let
go to innity.
This will also change the summation over integers to an
integration over real numbers.
Example 6.1
We know from Euler's formula that
cos (2f t) + sin (2f t) =
1j j2f t
2 e
1+j j2f t
.
2 e
6.2.3 The Fourier Transform

We have seen how all signals
s (t)
can be decomposed in terms of the sinusoids
ej2f t .
We also know that
these sinusoids are complex exponentials, and so they are eigenfunctions of LTI systems; recall as well that
such eigenfunctions easily pass through LTI systems. Thus, we can use these two principles to easily obtain
the output to any signal for any system: First, we obtain the Fourier coecients of the signal to decompose
it as the sum of scaled sinusoids; next, we run each scaled sinusoid through the system, which in essence
scales (multiplies) it by the sinusoid's eigenvalue; and nally we sum together all the outputs (thanks to
linearity and superposition) to obtain the output. What remains to be shown is the way to easily compute
the coecients and eigenvalues of a signal and a system. Both of these problems are solved using the Fourier
Transform.
Continuous-Time Fourier Transform

Z
S (f ) =
s (t) e(j2f t) dt
(6.4)
78
Inverse CTFT
S (f ) ej2f t df
s (t) =
(6.5)
For signals, the CTFT provides the Fourier coecients that are attached to sinusoids to represent the signal
as in (6.3). The inverse CTFT then provides the signal as the linear combination of all sinusoids with the
corresponding weights, extending (6.2). For systems, the CTFT can provide the eigenvalues for all sinusoids
when applied to the impulse response function
h (t).
Applying the CTFT to the input signal and the impulse
response allows for very easy computation of the system output, as we will soon observe.
warning: It is not uncommon to see the above formula written slightly dierent.
One of the
most common dierences is the way that the exponential is written. The above equations use the
f in the exponential. However, in some cases the radial frequency is used

= 2f . Therefore, you may often see the expressions above with in place of 2f
frequency variable
instead, where
in the exponentials. Click here
for an overview of the notation used in Connexion's DSP modules.
6.2.4 CTFT Denition Demonstration
Interact (when online) with a Mathematica CDF demonstrating Continuous Time Fourier
Transform. To Download, right-click and save as .cdf.
Figure 6.2:
5 "DSP
notation" <http://legacy.cnx.org/content/m10161/latest/>
79
6.2.5 Example Problems

Exercise 6.2.1
(Solution on p. 89.)
Find the Fourier Transform (CTFT) of the function
e(t) if t 0
x (t) =
0 otherwise
(6.6)
Exercise 6.2.2
Find the inverse Fourier transform of the ideal lowpass lter dened by
1
X (f ) =
0
6.3 Properties of the CTFT
if
|f | M
(6.7)
otherwise
6.3.1 Introduction
This module will look at some of the basic properties of the Continuous-Time Fourier Transform (CTFT).
6.3.2 Summary Table of Fourier Transform Properties

Table of Fourier Transform Properties
Operation Name
Linearity (Section 6.3.3.1:
Signal ( x (t) )
Transform ( X (f ) )
ax1 t + bx2 t
aX1 f + bX2 f
(Sec-
x (t)
X (f )
Dual-
X (t)
x (f )
x (t)
1
|| X
x (t )
X (f ) e(j2f )
Lin-
earity)
Scalar
Multiplication
tion 6.3.3.1: Linearity)

Duality (Section 6.3.3.2:
ity)
Time
Scaling
(Section
6.3.3.3:

f
Time Scaling)
Time Shift (Section 6.3.3.4: Time
Shifting)
continued on next page
6 This

80
Convolution
in
Time
(Sec-
x1 (t) x2 (t)
X1 (f ) X2 (f )
x1 (t) x2 (t)
X1 (f ) X2 (f )
dn
dtn x (t)
(j2f ) X (f )
tion 6.3.3.5: Convolution)

Convolution in Frequency (Section 6.3.3.5: Convolution)
Dierentiation
(Section
6.3.3.6:
Time Dierentiation)
Parseval's
Theorem
(Sec-
(|x (t) |) dt
(|X (f ) |) df
tion 6.3.3.7: Parseval's Relation)

Modulation
(Section
(Frequency
6.3.3.8:
Shift)
x (t) ej2t
X (f )
x (t)
X (f ) = X (f )
Modulation
(Frequency Shift))
Symmetry for Real Signals (Sec-
is real
tion 6.3.3.9: Symmetry for Real

Signals)
Table 6.1
6.3.3 Discussion of Fourier Transform Properties

6.3.3.1 Linearity
The combined addition and scalar multiplication properties in the table above demonstrate the basic property
of linearity. What you should see is that if one takes the Fourier transform of a linear combination of signals
then it will be the same as the linear combination of the Fourier transforms of each of the individual signals.
This is crucial when using a table (Section 6.4) of transforms to nd the transform of a more complicated
signal.
Example 6.2
We will begin with the following signal:
z (t) = ax1 (t) + bx2 (t)
(6.8)
Z (f ) = aX1 (f ) + bX2 (f )
(6.9)
6.3.3.2 Duality
Duality is a property that can make life quite easy when solving problems involving Fourier transforms.
Basically what this property says is that since a rectangular function in time is a sinc function in frequency,
then a sinc function in time will be a rectangular function in frequency. This is a direct result of the similarity
between the forward CTFT and the inverse CTFT. The only dierence is a frequency reversal.

This property deals with the eect on the frequency-domain representation of a signal if the time variable
is altered. The most important concept to understand for the time scaling property is that signals that are
narrow in time will be broad in frequency and
vice versa.
The simplest example of this is a delta function,
81
a unit pulse
with a
very
small duration, in time that becomes an innite-length constant function in
frequency.
The table above shows this idea for the general transformation from the time-domain to the frequencydomain of a signal. You should be able to easily notice that these equations show the relationship mentioned
previously: if the time variable is increased then the frequency range will be decreased.

Time shifting shows that a shift in time is equivalent to a linear phase shift in frequency. Since the frequency
content depends only on the shape of a signal, which is unchanged in a time shift, then only the phase
spectrum will be altered. This property is proven below:
Example 6.3
We will begin by letting
z (t) = x (t ). Now let us take the Fourier

z (t).
Z
Z (f ) =
x (t ) e(j2f t) dt
transform with the previous
expression substituted in for
(6.10)
Dene
= t .
Through the calculations below, you can see that only the variable in the
exponential are altered thus only changing the phase in the frequency domain.
Z (f )
= e
= e
x () e(j2f (+ )) d
R
x () e(j2f ) d
(j2f )
(j2f )
(6.11)
X (f )
6.3.3.5 Convolution
Convolution is one of the big reasons for converting signals to the frequency domain, since convolution in
time becomes multiplication in frequency.
This property is also another excellent example of symmetry
between time and frequency. It also shows that there may be little to gain by changing to the frequency
domain when multiplication in time is involved.
We will introduce the convolution integral here, but if you have not seen this before or need to refresh your
memory, then look at the continuous-time convolution (Section 6.5) module for a more in depth explanation
and derivation.
y (t)
=
=
x1 (t) x2 (t)
R
f ( ) f2 (t ) d
1
(6.12)
6.3.3.6 Time Dierentiation

8
Since LTI
systems can be represented in terms of dierential equations, it is apparent with this property
that converting to the frequency domain may allow us to convert these complicated dierential equations to
simpler equations involving multiplication and addition. This is often looked at in more detail during the
9
study of the Laplace Transform .
7 "Elemental Signals": Section Pulse <http://legacy.cnx.org/content/m0004/latest/#pulsedef>

8 "System Classications and Properties" <http://legacy.cnx.org/content/m10084/latest/>
9 "The Laplace Transform" <http://legacy.cnx.org/content/m10110/latest/>
82
6.3.3.7 Parseval's Relation

Z
(|x (t) |) dt =
(|X (f ) |) df
(6.13)
6.3.3.8 Modulation (Frequency Shift)

Modulation is absolutely imperative to communications applications. Being able to shift a signal to a dierent
frequency, allows us to take advantage of dierent parts of the electromagnetic spectrum is what allows us
to transmit television, radio and other applications through the same space without signicant interference.
The proof of the frequency shift property is very similar to that of the time shift (Section 6.3.3.4: Time
Shifting); however, here we would use the inverse Fourier transform in place of the Fourier transform. Since
we went through the steps in the previous, time-shift proof, below we will just show the initial and nal step
to this proof:
X (f ) ej2f t df
z (t) =
(6.14)
z (t) = f (t) ej2t
(6.15)
6.3.3.9 Symmetry for Real Signals

For signals that are real-valued, we have that
X (f )
x (t) = x (t).
Thus, we can evaluate
x (t) ej2f t dt
x (t) e(j2(f )t) dt
(6.16)
= X (f )

The following online resources provide interactive explanations of the properties of the CTFT:
Continuous-Time Fourier Transform Properties Applet (Internet Explorer)
Continuous-Time Fourier Transform Properties Applet (Other Browsers)
10
11
6.3.5 Properties Demonstration

An interactive example demonstration of the properties is included below:
This media object is a LabVIEW VI. Please view or download it at
<CTFTSPlab.llb>
Figure 6.3:
Interactive Signal Processing Laboratory Virtual Instrument created using NI's Labview.
10 http://www.jhu.edu/signals/ctftprops-mathml/index.htm
11 http://www.jhu.edu/signals/ctftprops/indexCTFTprops.htm
83
6.4 Common Fourier Transforms
12
6.4.1 Common CTFT Pairs
12 This

84
Time Domain Signal

e
(at)
u (t)
at
e u (t)
e(a|t|)
te
(at)
u (t)
n (at)
t e
u (t)
Frequency Domain Signal
Condition
1
a+j2f
1
aj2f
2a
a2 +(2f )2
1
(a+j2f )2
n!
(a+j2f )n+1
a>0
(t)
(f )
j2f0 t
a>0
a>0
a>0
(f f0 )
cos (2f0 t)
sin (2f0 t)
u (t)
sgn (t)
cos (2f0 t) u (t)
sin (2f0 t) u (t)
e(at) sin (2f0 t) u (t)
e
a>0
(at)
cos (2f0 t) u (t)

p 2t = u (t + ) u (t )
1
2 ( (f f0 ) + (f + f0 ))
j
2 ( (f + f0 ) (f f0 ))
1
1
2 (f ) + j2f
1
jf
1
4 ( (f f0 ) + (f + f0 ))
jf
2(f0 2 f 2 )
1
4j ( (f f0 ) (f + f0 ))
f0
2(f0 2 f 2 )
2f0
(a+j2f )2 +(2f0 )2
a+j2f
(a+j2f )2 +(2f0 )2
)
2 sin(2f
= 2 sinc (2f )
2f
0 t)
(20 ) sin(2f
=
2f0 t
(2f0 ) sinc (2f0 t)

t
=

t
t
t
+ 1 u +1 u +
t + 1 u t u t 1
u (f + f0 ) u (f f0 ) = p
f0 sinc2 (f0 t)
+
+
a>0
a>0

f
2f0
sinc2 (f )

u ff0 + 1 u ff0 +

ff0 + 1 u ff0 u ff0 1 =

ff0
f
f0
+1
85
P
e
n=
t2
2
2
P
f0 n= (f nf0 )
2
2e2(f )
(t nT )
1
T
f0 =
Table 6.2
Notes
p (t)
t.
is the pulse function for arbitrary real-valued
p (t) = {
(t)
if
|t| > 1/2
if
|t| 1/2
is the triangle function for arbitrary real-valued
1+t
(t) = { 1 t
(6.17)
t.
if
1t0
if
0<t1
(6.18)
otherwise
6.5 Continuous Time Convolution and the CTFT
13
6.5.1 Introduction
This module discusses convolution of continuous signals in the time and frequency domains.
6.5.2 Continuous Time Fourier Transform

The CTFT (Section 6.2) transforms a innite-length continuous signal in the time domain into an innitelength continuous signal in the frequency domain.
CTFT
Z
X (f ) =
x (f ) e(j2f t) dt
(6.19)
X (f ) ej2f t df
(6.20)
Inverse CTFT
x (t) =
6.5.3 Convolution Integral

The convolution integral expresses the output of an LTI system based on an input signal,
system's impulse response,
h (t).
The
convolution integral is expressed as

Z
x (t),
and the
x ( ) h (t ) d
y (t) =
(6.21)
Convolution is such an important tool that it is represented by the symbol
and can be written as
y (t) = x (t) h (t)
(6.22)
Convolution is commutative. For more information on the characteristics of the convolution integral, read
about the Properties of Convolution (Section 4.4).
13 This

86
6.5.4 Demonstration
Interact (when online) with a Mathematica CDF demonstrating Use of the CTFT in signal
denoising. To Download, right-click and save target as .cdf.
Figure 6.4:
87
6.5.5 Convolution Theorem

Let
and
be two functions with convolution
f g ..
Let
be the Fourier transform operator. Then
F (f g) = F (f ) F (g)
(6.23)
F (f g) = F (f ) F (g)
(6.24)
By applying the inverse Fourier transform
F 1 ,
we can write:
f g = F 1 (F (f ) F (g))
(6.25)
6.5.6 Conclusion
The Fourier transform of a convolution is the pointwise product of Fourier transforms.
In other words,
convolution in one domain (e.g., time domain) corresponds to point-wise multiplication in the other domain
(e.g., frequency domain).
6.6 Frequency-Domain Analysis of Linear Time-Invariant Systems
14
Convolution is the standard tool for the analysis of linear time-invariance (LTI) systems.
an LTI system, the impulse response
(t)) is the only

y (t) = x (t) h (t). Recall,
an impulse
h (t)
Recall that for
(which, as its name indicates, is the output of the system for
system knowledge required to compute the output to an arbitrary input as

also, that convolution involves a nontrivial computation involving shifting,
multiplication, and accumulation.
Thus, it becomes highly convenient to compute the output of the LTI
system in the frequency domain by applying a Fourier transform to the LTI system equation, which results
Y (f ) = X (f ) H (f ). Here, Y (f ), X (f ), and H (f ) denote the Fourier transforms of y (t), x (t), and

h (t), respectively. In summary, the Fourier transform allows us to obtain the output of a LTI system by
in
performing simple multiplication instead of a complicated convolution.

We can also obtain a signicant amount of intuition about LTI systems by considering them from the
frequency domain. Since Fourier transforms are complex-valued, we can rewrite the multiplication process
|Y (f ) | = |X (f ) H (f ) | = (|X (f ) ||H (f ) |), and a sum of phases, (Y (f )) =

(X (f )) + (H (f )).
Recall that |X (f ) |, intuitively, represents the "magnitude" of the presence of a complex sinusoid of
frequency f in the input signal x (t); similarly, |Y (f ) | represents the "magnitude" of the presence of the
same complex sinusoid in the output signal y (t). Therefore, we can argue that due to the pointwise nature of
as a product of magnitudes,
function multiplication the eect of the LTI system is to scale the "magnitude" of each complex sinusoid in
the input independently to obtain the output, with the scaling factors being given by the Fourier transform
|H (f ) |, and these factors being distinct for each possible frequency f .

(X (f )) represents the delay of the complex sinusoid of frequency f when it is
present in the input signal x (t); similarly, (Y (f )) represents the delay of the complex sinusoid of frequency
f when it is present in the output signal y (t). Therefore, we can argue that the LTI system imposes an
additional delay of (H (f )) to each complex sinusoid present in the input to obtain the delay's output of the
same complex sinusoid, and this factor (H (f )) is specic to each possible frequency due to the pointwise
of the impulse response
Similarly, recall that
nature of function addition.

Remarkably, these are the only two things that an LTI system can do to an input signal, and these
operations are specic to each frequency of the input signal's Fourier decomposition.
that an LTI system can remove a specic frequency
from an input signal by setting
Note, for example,
H (f ) = 0;
however,
an LTI system cannot "add" new frequencies to an input signal if they were not already present, and an
14 This

88
LTI system cannot change the values of the frequencies present in the signal (that is, an LTI system cannot
perform modulation; by the same token, a modulator is not an LTI system).
It is also worth noting that the delay of a complex sinusoid with a given phase is linearly dependent on
its frequency. Thus, a delay system (which we know is LTI) is characterized by a phase
be linearly dependent on the value of the frequency
f.
(H (f ))
that will
This is contrast with an LTI system whose phase
response is independent of the frequency, i.e., with constant
(H (f )); such a system will actually delay each

f.
complex exponential with a dierent delay that is inversely proportional to the frequency
89
Solutions to Exercises in Chapter 6

Solution to Exercise 6.2.1 (p. 79)
In order to calculate the Fourier transform, all we need to use is (6.4) (Continuous-Time Fourier Transform),
15
complex exponentials
, and basic calculus.
X (f )
0
0
X (f ) =
x (t) e(j2f t) dt
e(t) e(j2f t) dt
1
+j2f
1
+ j2f

Here we will use (6.5) (Inverse CTFT) to nd the inverse FT given that
(6.27)
t 6= 0.
RM
ej2f t df
M
1 j2f t M
|f =M
2t e
1
t sin (2M t)
(6.28)
x (t) = (2M ) (sinc (2M t))
(6.29)
x (t)
=
=
=
15 "Continuous
(6.26)
e(t)(+j2f ) dt

90
Chapter 7
Discrete-Time Signals
7.1 Common Discrete Time Signals
7.1.1 Introduction
Before looking at this module, hopefully you have an idea of what a signal is and what basic classications
and properties a signal can have. In review, a signal is a function dened with respect to an independent
variable.
This variable is often time but could represent any number of things.
Mathematically, discrete
time analog signals have discrete independent variables and continuous dependent variables. This module
will describe some useful discrete time analog signals.
7.1.2 Important Discrete Time Signals

7.1.2.1 Sinusoids
One of the most important elemental signal that you will deal with is the real-valued sinusoid.
In its
discrete-time form, we write the general expression as
Acos (n + )
where
is the amplitude,
(7.1)
is the phase. Because

2
is a rational number.
is the frequency, and
resulting function is only periodic if
only takes integer values, the
Discrete-Time Cosine Signal

sn
1
Figure 7.1:
A discrete-time cosine signal is plotted as a stem plot.
Note that the equation representation for a discrete time sinusoid waveform is not unique.
1 This

91
CHAPTER 7. DISCRETE-TIME SIGNALS
92

As important as the general sinusoid, the
complex exponential function will become a critical part of your
study of signals and systems. Its general discrete form is written as
Aesn
where
(7.2)
s = +j , is a complex number in terms of , the attenuation constant, and the angular frequency.
The discrete time complex exponentials have the following property.
ejn = ej(+2)n
(7.3)
Given this property, if we have a complex exponential with frequency

to a complex exponential with frequency
+ 2 ,
then this signal "aliases"
implying that the equation representations of discrete complex
exponentials are not unique.
7.1.2.3 Unit Impulses

The second-most important discrete-time signal is the
1
[n] =
0
unit sample, which is dened as

n=0
if
(7.4)
otherwise
Unit Sample
n
1
n
Figure 7.2:
The unit sample.
More detail is provided in the section on the discrete time impulse function. For now, it suces to say
that this signal is crucially important in the study of discrete signals, as it allows the sifting property to be
used in signal representation and signal decomposition.
7.1.2.4 Unit Step

Another very basic signal is the
unit-step function dened as
0
u [n] =
1
if
n<0
if
n0
(7.5)
93
Figure 7.3:
Discrete-Time Unit-Step Function
The step function is a useful tool for testing and for dening other signals. For example, when dierent
shifted versions of the step function are multiplied by other signals, one can select a certain portion of the
signal and zero out the rest.
7.1.3 Common Discrete Time Signals Summary

Some of the most important and most frequently encountered signals have been discussed in this module.
There are, of course, many other signals of signicant consequence not discussed here. As you will see later,
many of the other more complicated signals will be studied in terms of those listed here. Especially take
note of the complex exponentials and unit impulse functions, which will be the key focus of several topics
included in this course.
7.2 Energy and Power of Discrete-Time Signals
7.2.1 Signal Energy

7.2.1.1 Discrete signals
For time discrete signals the "area under the squared signal" makes no sense, so we will have to use another
energy deniton. We dene energy as the sum of the squared magnitude of the samples. Mathematically
Energy - Discrete time signal:
Example 7.1
Given the sequence
y [l] = bl u [l],
Ed =
where
n=
u [n]
(|x [n] |)
is the unit step function.
Find the energy of the
sequence.
y [l] as a geometric series.

use the formula for the sum of a geometric
P Thus we can
1
Ed = l=0 y 2 [l] = 1b
2 . This expression is only valid for |b| < 1.
larger |b|, the series will diverge. The signal y [l] then has innite energy. So let's have
We recognize
series and we obtain the energy,

If we have a
a look at power...
7.2.2 Signal Power

Our denition of energy seems reasonable, and it is. However, what if the signal does not decay fast enough?
In this case we have innite energy for any such signal. Does this mean that a fty hertz sine wave feeding
into your headphones is as strong as the fty hertz sine wave coming out of your outlet? Obviously not.
This is what leads us to the idea of
2 This
signal power, which in such cases is a more adequate description.

94
Figure 7.4:
Signal with ininite energy
7.2.2.1 Discrete signals

For time discrete signals we dene power as energy per sample.
Power - Discrete time:
1
N 2N +1
Pd = lim
PN
n=N
(|x [n] |)
For periodic discrete-time signals, the integral need only be dened over one period:
Power - Discrete time periodic signal with period
N 0 : Pd =
Example 7.2
1
N0
PN0 1
n=0
(|x [n] |)

1
n , shown in Figure 7.5, calculate the power for one period.
x [n] = sin 10

P20
2 1
1
3
discrete sine we get Pd =
n=1 sin
20
10 n = 0.500. Download power_sine.m
Given the signal

For the
plots and calculation.
Figure 7.5:
3 See
Discrete time sine.
the le at <http://legacy.cnx.org/content/m11526/latest/power_sine.m>

for
95
7.3 Discrete-Time Signal Operations
7.3.1 Introduction
This module will look at two signal operations aecting the time parameter of the signal, time shifting
and time scaling. While they appear at rst to be straightforward extensions of the continuous-time signal
operations, there are some intricacies that are particular to discrete-time signals.
7.3.2 Manipulating the Time Parameter

Time shifting is, as the name suggests, the shifting of a signal in time. This is done by adding or subtracting
an integer quantity of the shift to the time variable in the function. Subtracting a xed positive quantity
from the time variable will shift the signal to the right (delay) by the subtracted quantity, while adding a
xed positive amount to the time variable will shift the signal to the left (advance) by the added quantity.
Figure 7.6:
f [n 3] moves (delays) f to the right by 3.

Time scaling compresses or dilates a signal by multiplying the time variable by some quantity. If that quantity
is greater than one, the signal becomes narrower and the operation is called decimation. In contrast, if the
quantity is less than one, the signal becomes wider and the operation is called expansion or interpolation,
depending on how the gaps between values are lled.
7.3.2.2.1 Decimation
In decimation, the input of the signal is changed to be
f [cn]
. The quantity used for decimation
must be
an integer so that the input takes values for which a discrete function is properly dened. The decimated
signal
f [cn]
corresponds to the original signal
f [n]
where only each
sample is preserved (including
and so we are throwing away samples of the signal (or decimating it).
4 This
f [0]),
96
Figure 7.7:
f [2n] decimates f by 2.
7.3.2.2.2 Expansion
In expansion, the input of the signal is changed to be
for integer values of the input
signal
signal
n.
n
c
. We know that the signal
f [n]
is dened only
Thus, in the expanded signal we can only place the entries of the original
f at values of n that are multiples of c. In other words, we are spacing the values of the discrete-time
c 1 entries away from each other. Since the signal is undened elsewhere, the standard convention
in expansion is to ll in the undetermined values with zeros.
Figure 7.8:
n
2
expands f by 2.
7.3.2.2.3 Interpolation
In practice, we may know specic information about the signal of interest that allows us to provide good
estimates of the entries of
n
c
that are missing after expansion. For example, we may know that the signal
is supposed to be piecewise linear, and so knowing the values of

infer the values for
between
P
and
n
c
at
n=
m
c and at
n=
P
c
allows us to
. This process of inferring the undened values is known as
interpolation. The rule described above is known as polar interpolation; although more sophisticated rules
exist for interpolating values, linear interpolation will suce for our explanation in this module.
97
Figure 7.9:
n
2
with interpolation lls in the missing values of the expansion using linear extensions.
Example 7.3
Given
f [n] we woul like to plot f [an b].
(a)
Figure 7.10:
n ab
f
to get
The gure below describes a method to accomplish this.
(b)
(c)
(a)
` Beginwith f [n] (b) Then replace n with an to get f [an] (c) Finally, replace n with
a n ab = f [an b]
7.3.2.3 Time Reversal

A natural question to consider when learning about time scaling is: What happens when the time variable
is multiplied by a negative number? The answer to this is time reversal. This operation is the reversal of
the time axis, or ipping the signal over the y-axis.
Figure 7.11:
Reverse the time axis
98
7.3.3 Signal Operations Summary

Some common operations on signals aect the time parameter of the signal. One of these is time shifting in
which a quantity is added to the time parameter in order to advance or delay the signal. Another is the time
scaling in which the time parameter is multiplied by a quantity in order to expand or decimate the signal in
time. In the event that the quantity involved in the latter operation is negative, time reversal occurs.
7.4 Discrete Time Impulse Function
7.4.1 Introduction
In engineering, we often deal with the idea of an action occurring at a point.
Whether it be a force at
a point in space or some other signal at a point in time, it becomes worth while to develop some way of
quantitatively dening this. This leads us to the idea of a unit impulse, probably the second most important
function, next to the complex exponential, in this systems and signals course.
7.4.2 Unit Sample Function

The
unit sample function,
often referred to as the unit impulse or delta function, is the function that
denes the idea of a unit impulse in discrete time. There are not nearly as many intricacies involved in its
denition as there are in the denition of the Dirac delta function, the continuous time impulse function.
The unit sample function simply takes a value of one at n=0 and a value of zero elsewhere. The impulse
function is often written as
(n).
1
[n] =
0
if
n=0
(7.6)
otherwise
Unit Sample
n
1
n
Figure 7.12:
The unit sample.
Below we will briey list a few important properties of the unit impulse without going into detail of their
proofs.
Unit Impulse Properties
1
[n] = ||
[n]
[n] = [n]
[n] = u [n] u [n 1]
f [n] [n] = f [0] [n]
5 This
99
f [n] [n] =
n=
f [0] [n] = f [0]
n=
[n] = f [0]
(7.7)
n=
7.4.3 Discrete Time Impulse Response Demonstration
Figure 7.13:
Function.
Interact(when online) with a Mathematica CDF demonstrating the Discrete Time Impulse
7.4.4 Discrete Time Unit Impulse Summary

The discrete time unit impulse function, also known as the unit sample function, is of great importance
to the study of signals and systems.
The function takes a value of one at time n=0 and a value of zero
100
elsewhere. It has several important properties that will appear again when studying systems.
7.5 Discrete Time Complex Exponential
7.5.1 Introduction
Complex exponentials are some of the most important functions in our study of signals and systems. Their
importance stems from their status as eigenfunctions of linear time invariant systems; as such, it can be
both convenient and insightful to represent signals in terms of complex exponentials. Before proceeding, you
should be familiar with complex numbers.
7.5.2 The Discrete Time Complex Exponential

The complex exponential function will become a critical part of your study of signals and systems. Its general
discrete form is written as
zn
(7.8)
z is a complex number. Recalling the polar expression of complex numbers, z can be expressed in terms
n
|z| and its angle (or argument) in the complex plane: z = |z|ej . Thus z n = (|z|) ejn .
In the context of complex exponentials, is referred to as frequency. For the time being, let's consider
complex exponentials for which |z| = 1.
where
of its magnitude
These discrete time complex exponentials have the following property, which will become evident through
discussion of Euler's formula.
ejn = ej(+2)n
(7.9)
Given this property, if we have a complex exponential with frequency

to a complex exponential with frequency
+ 2 ,
then this signal "aliases"
implying that the equation representations of discrete complex
exponentials are not unique.
7.5.2.2 Euler's Formula

The mathematician Euler proved an important identity relating complex exponentials to trigonometric functions. Specically, he discovered the eponymously named identity, Euler's formula, which states that for any
real number
x,
ejx = cos (x) + jsin (x)
(7.10)
which can be proven as follows.

In order to prove Euler's formula, we start by evaluating the Taylor series for
converges for all complex
z,
at
z = jx.
about
z = 0,
which
The result is
ejx
=
=
ez
k=0
k x2k
(2k)!
(1)
k=0
+j
(jx)k
k!
P
k=0
k x2k+1
(2k+1)!
(1)
(7.11)
= cos (x) + jsin (x)

because the second expression contains the Taylor series for
for all real
x.
Choosing
cos (x) and sin (x) about t = 0,
x = n,
we have:
ejn = cos (n) + jsin (n)

6 This
which converge
Thus, the desired result is proven.

(7.12)
101
which breaks a discrete time complex exponential into its real part and imaginary part. Using this formula,
we can also derive the following relationships.
1 jn 1 jn
e
+ e
2
2
(7.13)
1 jn
1
e
ejn
2j
2j
(7.14)
cos (n) =
sin (n) =
7.5.2.3 Real and Imaginary Parts of Complex Exponentials

Now let's return to the more general case of complex exponentials,
zn.
Recall that
z n = (|z|) ejn .
We can
express this in terms of its real and imaginary parts:
Re{z n } = (|z|) cos (n)
(7.15)
Im{z n } = (|z|) sin (n)

We see now that the magnitude of
establishes an exponential envelope to the signal, with
rate of the sinusoidal oscillation within the envelope.
(7.16)
controlling
102
(a)
(b)
(c)
(a) If |z| < 1, we have the case of a decaying exponential envelope. (b) If |z| > 1, we
have the case of a growing exponential envelope. (c) If |z| = 1, we have the case of a constant envelope.
Figure 7.14:
103
7.5.3 Discrete Complex Exponential Demonstration
Interact (when online) with a Mathematica CDF demonstrating the Discrete Time Complex Exponential. To Download, right-click and save target as .cdf.
Figure 7.15:
104
7.5.4 Discrete Time Complex Exponential Summary

Discrete time complex exponentials are signals of great importance to the study of signals and systems. They
can be related to sinusoids through Euler's formula, which identies the real and imaginary parts of complex
exponentials. Eulers formula reveals that, in general, the real and imaginary parts of complex exponentials
are sinusoids multiplied by real exponentials.
Chapter 8
Time Domain Analysis of Discrete Time
Systems
8.1 Discrete Time Systems
8.1.1 Introduction
As you already now know, a discrete time system operates on a discrete time signal input and produces a
discrete time signal output. There are numerous examples of useful discrete time systems in digital signal
processing, such as digital lters for images or sound.
The class of discrete time systems that are both
linear and time invariant, known as discrete time LTI systems, is of particular interest as the properties of
linearity and time invariance together allow the use of some of the most important and powerful tools in
signal processing.
8.1.2 Discrete Time Systems

8.1.2.1 Linearity and Time Invariance
A system
is said to be linear if it satises two important conditions. The rst, additivity, states for every
pair of signals
signal
x, y
that
and scalar
H (x + y) = H (x) + H (y).
H (ax) = aH (x). It
we have
The second, homogeneity of degree one, states for every

is clear that these conditions can be combined together
into a single condition for linearity. Thus, a system is said to be linear if for every signals
a, b
x, y
and scalars
we have that
H (ax + by) = aH (x) + bH (y) .
(8.1)
Linearity is a particularly important property of systems as it allows us to leverage the powerful tools of
linear algebra, such as bases, eigenvectors, and eigenvalues, in their study.
A system
is said to be time invariant if a time shift of an input produces the corresponding shifted
output. In other, more precise words, the system
T Z.
commutes with the time shift operator
ST
for every
That is,
ST H = HST .
(8.2)
Time invariance is desirable because it eases computation while mirroring our intuition that, all else equal,
physical systems should react the same to identical inputs at dierent times.
When a system exhibits both of these important properties it opens. As will be explained and proven
in subsequent modules, computation of the system output for a given input becomes a simple matter of
1 This

105
CHAPTER 8. TIME DOMAIN ANALYSIS OF DISCRETE TIME SYSTEMS
106
convolving the input with the system's impulse response signal. Also proven later, the fact that complex
exponential are eigenvectors of linear time invariant systems will encourage the use of frequency domain
tools such as the various Fouier transforms and associated transfer functions, to describe the behavior of
linear time invariant systems.
Example 8.1
Consider the system
in which
H (f [n]) = 2f [n]
for all signals
f.
f, g
and scalars
(8.3)
a, b
H (af [n] + bg [n]) = 2 (af [n] + bg [n]) = a2f [n] + b2g [n] = aH (f [n]) + bH (g [n])
for all integers
n.
Thus,
is a linear system. For all integers
and signals
(8.4)
f,
ST (H (f [n])) = ST (2f [n]) = 2f [n T ] = H (f [n T ]) = H (ST (f [n]))

for all integers
n.
Thus,
is a time invariant system.
Therefore,
(8.5)
is a linear time invariant
system.
8.1.2.2 Causality
The causality property requires that a system's output depends only on past and present values of the input.
For a discrete-time system, this means that the value of the output
depend on values of the input
x [n]
for
y [n0 ]
at a specic time
n0
can only
n < n0 .
8.1.2.3 Stability
There are several denitions of stability, but the one that will be used most frequently in this course will
be bounded input, bounded output (BIBO) stability. In this context, a stable system is one in which the
output is bounded if the input is also bounded. Similarly, an unstable system is one in which at least one
bounded input produces an unbounded output.
In order to understand this concept, we must rst look more closely into exactly what we mean by
bounded. A bounded signal is any signal such that there exists a value such that the absolute value of the
signal is never greater than some value. Since this value is arbitrary, what we mean is that at no point can
the signal tend to innity, including the end behavior.
A bounded signal
f [n]
is a signal for which there exists a constant
such that for all
we have that
|f [n] | < A
Representing this mathematically, a stable system must have the following property, where
input and
y (n)
Mx and My
x (n)
is the
is the output. The output must satisfy the condition
|y [n] | My <
(8.6)
|x [n] | Mx <
(8.7)
both represent a set of nite positive numbers and these relationships hold for all of
the system is unstable.
n.
Otherwise,
107
8.1.3 Discrete Time Systems Summary

Many useful discrete time systems will be encountered in a study of signals and systems. This course is most
interested in those that demonstrate both the linearity property and the time invariance property, which
together enable the use of some of the most powerful tools of signal processing. It is often useful to describe
them in terms of rates of change through linear constant coecient dierence equations, a type of recurrence
relation.
8.2 Discrete Time Impulse Response
8.2.1 Introduction
The output of a discrete time LTI system is completely determined by the input and the system's response
to a unit impulse.
System Output
We can determine the system's output, y[n], if we know the system's impulse response,
h[n], and the input, x[n].
Figure 8.1:
The output for a unit impulse input is called the impulse response.
Figure 8.2
2 This
108
(a)
(b)
Figure 8.3
8.2.1.1 Example Impulses

Since we are considering discrete time signals and systems, an ideal impulse is easy to simulate on a computer
or some other digital device. It is simply a signal that is
at the point
n = 0,
and
everywhere else.
8.2.2 LTI Systems and Impulse Responses

8.2.2.1 Finding System Outputs
By the sifting property of impulses, any signal can be decomposed in terms of an innite sum of shifted,
scaled impulses.
x [n]
The function
k [n] = [n k]
x [k] k [n]
x [k] [n k]
peaks up where
k=
k=
n = k.
(8.8)
109
(a)
(b)
Figure 8.4
Since we know the response of the system to an impulse and any signal can be decomposed into impulses,
all we need to do to nd the response of the system to any signal is to decompose the signal into impulses,
calculate the system's output for every impulse and add the outputs back together.
known as
Convolution.
Since we are in
This is the process
Discrete Time, this is the Discrete Time Convolution Sum.
8.2.2.2 Finding Impulse Responses

a. Apply an impulse input signal to the system and record the output
b. Use Fourier methods
We will assume that
h [n]
is given for now.
The goal is now to compute the output
y [n]
given the impulse response
h [n]
and the input
x [n].
Figure 8.5
8.2.3 Impulse Response Summary

When a system is "shocked" by a delta function, it produces an output known as its impulse response. For
an LTI system, the impulse response completely determines the output of the system given any arbitrary
input. The output can be found using discrete time convolution.
110
8.3 Discrete-Time Convolution
8.3.1 Introduction
output a system produces for a given input signal. It can be shown that a linear time invariant system is
completely characterized by its impulse response. The sifting property of the discrete time impulse function
tells us that the input signal to a system can be represented as a sum of scaled and shifted unit impulses.
Thus, by linearity, it would seem reasonable to compute of the output signal as the sum of scaled and shifted
unit impulse responses. That is exactly what the operation of convolution accomplishes. Hence, convolution
can be used to determine a linear time invariant system's output from knowledge of the input and the impulse
response.
8.3.2 Convolution and Circular Convolution

8.3.2.1 Convolution
8.3.2.1.1 Operation Denition
Discrete time convolution is an operation on two discrete time signals dened by the integral
(f g) [n] =
f [k] g [n k]
(8.9)
k=
f, g
for all signals
dened on
Z.
It is important to note that the operation of convolution is commutative,
meaning that
f g =gf
for all signals
f, g
dened on
Z.
(8.10)
Thus, the convolution operation could have been just as easily stated using
the equivalent denition
(f g) [n] =
f [n k] g [k]
(8.11)
k=
for all signals
f, g
dened on
Z.
Convolution has several other important properties not listed here but
explained and derived in a later module.

a system input signal
H (x).
h.
Given
First, we note that the
input can be expressed as the convolution
x [n] =
x [k] [n k]
(8.12)
k=
by the sifting property of the unit impulse function. By linearity
Hx [n] =
x [k] H [n k] .
k=
3 This

(8.13)
111
Since
H [n k]
Hx [n] =
h [n k],
x [k] h [n k] = (x h) [n] .
(8.14)
k=
Hence, convolution has been dened such that the output of a linear time invariant system is given by the
convolution of the system input with the system unit impulse response.

It is often helpful to be able to visualize the computation of a convolution in terms of graphical processes.
Consider the convolution of two functions
f, g
(f g) [n] =
given by
f [k] g [n k] =
k=
f [n k] g [k] .
(8.15)
k=
The rst step in graphically understanding the operation of convolution is to plot each of the functions.
Next, one of the functions must be selected, and its plot reected across the
same function must be shifted left by
t.
k=0
axis. For each real t, that
The product of the two resulting plots is then constructed. Finally,
the area under the resulting curve is computed.
Example 8.2
Recall that the impulse response for a discrete time echoing feedback system with gain
h [n] = an u [n] ,
is
(8.16)
and consider the response to an input signal that is another exponential
x [n] = bn u [n] .
(8.17)
We know that the output for this input is given by the convolution of the impulse response with
the input signal
y [n] = x [n] h [n] .
(8.18)
We would like to compute this operation by beginning in a way that minimizes the algebraic
complexity of the expression. However, in this case, each possible coice is equally simple. Thus, we
would like to compute
y [n] =
ak u [k] bnk u [n k] .
(8.19)
k=
The step functions can be used to further simplify this sum. Therefore,
y [n] = 0
for
n<0
(8.20)
and
y [n] =
n
X
(ab)
(8.21)
k=0
for
n 0.
Hence, provided
ab 6= 1,
we have that
y [n] = {
n<0
1(ab)n+1
1(ab)
n0
(8.22)
112
8.3.2.2 Circular Convolution

Discrete time circular convolution is an operation on two nite length or periodic discrete time signals dened
by the integral
(f g) [n] =
N
1
X
f [k] g [n k]
(8.23)
k=0
for all signals
f, g
dened on
Z [0, N 1]
where
^ ^
f, g
are periodic extensions of
and
g.
It is important to
note that the operation of circular convolution is commutative, meaning that
f g =gf
for all signals
f, g
Z [0, N 1].
dened on
(8.24)
Thus, the circular convolution operation could have been just as
easily stated using the equivalent denition
(f g) [n] =
N
1
X
f [n k] g [k]
(8.25)
k=0
for all signals
f, g
dened on
^ ^
Z [0, N 1] where f , g
are periodic extensions of
and
g.
Circular convolution
has several other important properties not listed here but explained and derived in a later module.
Alternatively, discrete time circular convolution can be expressed as the sum of two summations given
by
(f g) [n] =
n
X
f [k] g [n k] +
f, g
dened on
f [k] g [n k + N ]
(8.26)
k=n+1
k=0
for all signals
N
1
X
Z [0, N 1].
Meaningful examples of computing discrete time circular convolutions in the time domain would involve
complicated algebraic manipulations dealing with the wrap around behavior, which would ultimately be
more confusing than helpful. Thus, none will be provided in this section. Of course, example computations
in the time domain are easy to program and demonstrate. However, disrete time circular convolutions are
more easily computed using frequency domain tools as will be shown in the discrete time Fourier series
section.

a nite or periodic system input signal
h.
H (x).
Given
First,
we note that the input can be expressed as the circular convolution
x (n) =
N
1
X
x [k] [n k]
(8.27)
k=0
by the sifting property of the unit impulse function. By linearity,
Hx [n] =
N
1
X
x [k] H [n k] .
k=0
(8.28)
113
Since
H (n k)
Hx [n] =
N
1
X
h (n k),
x [k] h [n k] = (x h) [n] .
(8.29)
k=0
Hence, circular convolution has been dened such that the output of a linear time invariant system is given
by the convolution of the system input with the system unit impulse response.

It is often helpful to be able to visualize the computation of a circular convolution in terms of graphical
processes. Consider the circular convolution of two nite length functions
(f g) [n] =
N
1
X
f [k] g [n k] =
k=0
N
1
X
f, g
given by
f [n k] g [k] .
(8.30)
k=0
The rst step in graphically understanding the operation of convolution is to plot each of the periodic
extensions of the functions. Next, one of the functions must be selected, and its plot reected across the
k=0
axis. For each
k Z [0, N 1],
that same function must be shifted left by
k . The product of the two

Z [0, N 1] is computed.
resulting plots is then constructed. Finally, the area under the resulting curve on

The following website provides an interactive Java applet illustrating discrete-time convolution:
4
Discrete Joy of Convolution (Johns Hopkins University)
4 http://www.jhu.edu/signals/discreteconv/index.html
114
8.3.4 Interactive Element
Interact (when online) with the Mathematica CDF demonstrating Discrete Linear ConvoAvailable
for free
Connexions
lution. To download,
right
clickatand
save le <http://legacy.cnx.org/content/col11557/1.10>
as .cdf
Figure 8.6:
115
8.3.5 Convolution Summary

output signal of a linear time invariant system for a given input signal with knowledge of the system's unit
impulse response. The operation of discrete time convolution is dened such that it performs this function
for innite length discrete time signals and systems. The operation of discrete time circular convolution is
dened such that it performs this function for nite length and periodic discrete time signals. In each case,
the output of the system is the convolution or circular convolution of the input signal with the unit impulse
response.
8.4 Properties of Discrete Time Convolution
8.4.1 Introduction
We have already shown the important role that discrete time convolution plays in signal processing. This
section provides discussion and proof of some of the important properties of discrete time convolution.
Analogous properties can be shown for discrete time circular convolution with trivial modication of the
proofs provided except where explicitly noted otherwise.
8.4.2 Discrete Time Convolution Properties

8.4.2.1 Associativity
The operation of convolution is associative.
That is, for all discrete time signals
f1 , f2 , f3
the following
relationship holds.
f1 (f2 f3 ) = (f1 f2 ) f3
(8.31)
(f1 (f2 f3 )) [n]

=
P
k1 =
k2 = f1 [k1 ] f2 [k2 ] f3 [(n k1 ) k2 ]
P
P
k1 =
k2 = f1 [k1 ] f2 [(k1 + k2 ) k1 ] f3 [n (k1 +
P
P
= k3 = k1 = f1 [k1 ] f2 [k3 k1 ] f3 [n k3 ]
=
k2 )]
(8.32)
= ((f1 f2 ) f3 ) [n]
k3 = k1 + k2 .
8.4.2.2 Commutativity
The operation of convolution is commutative.
That is, for all discrete time signals
f1 , f2
the following
relationship holds.
f1 f2 = f2 f1
(8.33)
(f1 f2 ) [n]
f1 [k1 ] f2 [n k1 ]
f1 [n k2 ] f2 [k2 ]
k1 =
k2 =
= (f2 f1 ) [n]
5 This
k2 = n k1 .

(8.34)
116
8.4.2.3 Distribitivity
The operation of convolution is distributive over the operation of addition.
signals
f1 , f2 , f3
That is, for all discrete time
f1 (f2 + f3 ) = f1 f2 + f1 f3
(8.35)
(f1 (f2 + f3 )) [n]

=
f1 [k] (f2 [n k] + f3 [n k])

P
k= f1 [k] f2 [n k] +
k= f1 [k] f3 [n k]
=
k=
(8.36)
= (f1 f2 + f1 f3 ) [n]
8.4.2.4 Multilinearity
The operation of convolution is linear in each of the two function variables.
Additivity in each variable
results from distributivity of convolution over addition. Homogenity of order one in each varible results from
the fact that for all discrete time signals
f1 , f2
and scalars
a (f1 f2 ) = (af1 ) f2 = f1 (af2 )
(8.37)
(a (f1 f2 )) [n]
P
= a k= f1 [k] f2 [n k]
P
= k= (af1 [k]) f2 [n k]
= ((af1 ) f2 ) [n]
P
= k= f1 [k] (af2 [n k])
(8.38)
= (f1 (af2 )) [n]

8.4.2.5 Conjugation
The operation of convolution has the following property for all discrete time signals
f1 , f2 .
f1 f2 = f1 f2
(8.39)

f1 f2 [n]
f1 [k] f2 [n k]
f1 [k] f2 [n k]
k=
k=
f1 [k] f2 [n k]

= f1 f2 [n]
k=
(8.40)
117
8.4.2.6 Time Shift

time shift operator with
f1 , f2
where
ST
is the
T Z.
ST (f1 f2 ) = (ST f1 ) f2 = f1 (ST f2 )
(8.41)
ST (f1 f2 ) [n]
k=
f2 [k] f1 [(n T ) k]
k=
f2 [k] ST f1 [n k]
= ((ST f1 ) f2 ) [n]
=
(8.42)
k= f1 [k] f2 [(n T ) k]
P
k= f1 [k] ST f2 [n k]
= f1 (ST f2 ) [n]
8.4.2.7 Impulse Convolution

where
is the unit
sample funciton.
f =f
(8.43)
(f ) [n]
=f
k= f [k] [n k]
P
[n] k= [n k]
(8.44)
= f [n]
8.4.2.8 Width
gives the duration of a signal
f1 , f2 where Duration (f )
f.
Duration (f1 f2 ) = Duration (f1 ) + Duration (f2 ) 1

. In order to show this informally, note that
(f1 f2 ) [n]
is nonzero for all
(8.45)
for which there is a
such that
f1 [k] f2 [n k]
is nonzero. When viewing one function as reversed and sliding past the other, it is easy to
see that such a
exists for all
on an interval of length
Duration (f1 ) + Duration (f2 ) 1.
Note that this
is not always true of circular convolution of nite length and periodic signals as there is then a maximum
possible duration within a period.
8.4.3 Convolution Properties Summary

As can be seen the operation of discrete time convolution has several important properties that have been
listed and proven in this module. With silight modications to proofs, most of these also extend to discrete
time circular convolution as well and the cases in which exceptions occur have been noted above.
identities will be useful to keep in mind as the reader continues to study signals and systems.
These
118
8.5 Causality and Stability of Discrete-Time Linear Time-Invariant

Systems
6
8.5.1 Introduction
We have previously dened the system properties of causality and bounded-input bounded-output (BIBO)
stability.
We have also determined that a linear time-invariant (LTI) system is completely determined
by its impulse response
y [n] = x [n] h [n].
h [n]:
the output
y [n]
for an arbitrary input
x [n]
is obtained via convolution as
It should not be surprising then that one can determine whether an LTI system is causal
or BIBO stable simply by inspecting its impulse response
h [n].
8.5.2 Causality
Recall that a system is causal if its output
n n0 .
y [n0 ]
n0
at time
depends only on the input
x [n]
for values of
Consider then the characterization of the system by the convolution sum:
y [n]
We replace the time variable
= x [n] h [n] =
by a xed value
y [n0 ]
m=
x [m] h [n m] .
(8.46)
t0 :
m=
For this input to depend only on values of time
x [m] h [n0 m] .
m n0
(where
(8.47)
represents a second time variable), we
must have its contribution to the integral being nulled out by requiring the value of the impulse response
h [n0 m] = 0
m > n0 . By making a change of variables n = n0 m (so that m = n0 n),

h [n] = 0 for n0 n > n0 ; this means that we require h [n] = 0 for n < 0.
an LTI system with impulse response h [n] is causal if and only if h [n] = 0 for all
for values of
this means that we require

Thus, we obtain that
n < 0.
8.5.3 BIBO Stability

x [n] is bounded (that is, if there exists M <
|x [n] | < M for all n) then we have that the system output is also bounded (that is, if there exists
N < such that |y [n] | < N for all t). As before, consider then the characterization of the system by the
Recall that a system is BIBO stable if whenever the input
such that
convolution integral:
y [n]
= x [n] h [n] =
m=
x [m] h [n m] .
(8.48)
We apply absolute value to both sides and use the triangle inequality on a sum:
P
P
|y (n)| = m= x [m] h [n m] m= |x [m] h [n m] |
P
P
= m= |x [m] ||h [n m] | m= M |h [n m] |
P
= M m= |h [n m] |,
x [n]. Thus, if we have that the integral has a nite

|y [n] |, which implies BIBO stability.
an LTI system with impulse response h [n] is BIBO stable if and only if
where the next-to-last step uses the boundedness of
value, then we have established a nite upper bound on
PThus,
m=
6 This
we obtain that
|h [n] | is nite.
(8.49)
119
8.5.4 Summary
The derivations above show that it is signicantly easier to verify whether a system is causal and/or BIBO
stable when it is linear and time-invariant.
evaluations of the system's impulse response
In such a case, we simply need to perform straightforward
h [n].
120
Chapter 9
Discrete Time Fourier Transform
(DTFT)
9.1 Discrete Time Aperiodic Signals
9.1.1 Introduction
This module describes the type of signals acted on by the Discrete Time Fourier Transform.
9.1.2 Periodic and Aperiodic Signals

periodic.
periodic
f [n] = f [n + mN ] m Z
where
N > 0
represents the
fundamental period
(9.1)
of the signal, which is the smallest positive value of
for the signal to repeat. Because of this, you may also see a signal referred to as an
Any function that satises this equation is said to be
periodic with period N .
N -periodic
signal.
Periodic signals in discrete
time repeats themselves in each cycle. However, only integers are allowed as time variable in discrete time.
We denote signals in such case as
f [n],
where
discrete-time periodic signal with period N :

1 This
takes value over the integers.
Here's an example of a

121
CHAPTER 9. DISCRETE TIME FOURIER TRANSFORM (DTFT)
122
discrete-time periodic signal
Figure 9.1:
Notice the function is the same after a time shift of N
periodic functions (with period N ) two dierent ways:

as functions on all of R
We can think of
1.
Figure 9.2:
discrete time periodic function over all of R where f [n0 ] = f [n0 + N ]
2. or, we can cut out all of the redundancy, and think of them as functions on an interval
[0, N ]
(or,
123
more generally,
[a, a + N ]).
If we know the signal is
N -periodic
then all the information of the signal
is captured by the above interval.
Figure 9.3:
An
Remove the redundancy of the period function so that f [n] is undened outside [0, N ].
aperiodic DT function, however, f [n] does not repeat for any N
R; i.e.
there exists no
such
that this equation (9.1) holds. This broader class of signals can only be acted upon by the DTFT.
Suppose we have such an aperiodic function
fN o [n]
, where
f [n]
is repeated every
N0
f [n]
. We can construct a periodic extension of
seconds. If we take the limit as
N0 ,
f [n]
called
we obtain a precise
model of an aperiodic signal for which all rules that govern periodic signals can be applied, including Fourier
Analysis (with an important modication).
Discete Time Fourier Transform.
For more detail on this distinction, see the module on the
124
9.1.3 Aperiodic Signal Demonstration
Click on the above thumbnail image (when online) to download an interactive Mathematica
Player testing Periodic versus Aperiodic Signals. To download, right-click and save as .cdf.
Figure 9.4:
9.1.4 Conclusion
A discrete periodic signal is completely dened by its values in one period, such as the interval
[0, N ].
Any
aperiodic signal can be dened as an innite sum of periodic functions, a useful denition that makes it
possible to use Fourier Analysis on it by assuming all frequencies are present in the signal.
9.2 Eigenfunctions of Discrete Time LTI Systems
9.2.1 Introduction
Prior to reading this module, the reader should already have some experience with linear algebra and should
specically be familiar with the eigenvectors and eigenvalues of linear operators.
A linear time invariant
system is a linear operator dened on a function space that commutes with every time shift operator on
that function space. Thus, we can also consider the eigenvector functions, or eigenfunctions, of a system.
It is particularly easy to calculate the output of a system when an eigenfunction is the input as the output
is simply the eigenfunction scaled by the associated eigenvalue.
As will be shown, discrete time complex
exponentials serve as eigenfunctions of linear time invariant systems operating on discrete time signals.
2 This
125
9.2.2 Eigenfunctions of LTI Systems

Consider a linear time invariant system
with impulse response
discrete time signals. Recall that the output
H (x [n])
h operating on some space of innite length

x [n] is given by the
of the system for a given input
discrete time convolution of the impulse response with the input
H (x [n]) =
h [k] x [n k] .
(9.2)
k=
Now consider the input
x [n] = esn
where
s C.
H (esn )
=
=
=
Computing the output for this input,
k=
h [k] es(nk)
sn sk
k= h [k] e e
P
esn k= h [k] esk .
(9.3)
Thus,
H (esn ) = s esn
(9.4)
where
s =
h [k] esk
(9.5)
k=
is the eigenvalue corresponding to the eigenvector
esn .
There are some additional points that should be mentioned.

eigenvalues of a linear time invariant system not described by
discussion has been somewhat formally loose as
esn
esn
Note that, there still may be additional

for some
s C.
Furthermore, the above
may or may not belong to the space on which the system
operates. However, for our purposes, complex exponentials will be accepted as eigenvectors of linear time
invariant systems. A similar argument using discrete time circular convolution would also hold for spaces
nite length signals.
9.2.3 Eigenfunction of LTI Systems Summary

As has been shown, discrete time complex exponential are eigenfunctions of linear time invariant systems
operating on discrete time signals. Thus, it is particularly simple to calculate the output of a linear time
invariant system for a complex exponential input as the result is a complex exponential output scaled by
the associated eigenvalue. Consequently, representations of discrete time signals in terms of discrete time
complex exponentials provide an advantage when studying signals. As will be explained later, this is what
is accomplished by the discrete time Fourier transform and discrete time Fourier series, which apply to
aperiodic and periodic signals respectively.
9.3 Discrete Time Fourier Transform (DTFT)
9.3.1 Introduction
In this module, we will derive an expansion for arbitrary discrete-time functions, and in doing so, derive the
Discrete Time Fourier Transform (DTFT).

3 This
126

output of an LTI system
are eigenfunctions of linear time-invariant (LTI) systems , calculating the
given
eigenvalue corresponding to
ejn
H () C is the
As shown in the gure, a simple exponential input would yield the output
y [n] = H () ejn
Figure 9.5:
(9.6)
Simple LTI system.
y [n]
for combinations of
complex exponentials is
also straightforward.
c1 ej1 n + c2 ej2 n c1 H (1 ) ej1 n + c2 H (2 ) ej1 n

X
cl ejl n
l
The action of
cl H (l ) ejl n
H indepenH (l ) C. As such, if
dently scales each exponential component ej n

l
y [n]
output of a system.
Now, we will look to use the power of complex exponentials to see how we may represent arbitrary signals
in terms of a set of simpler functions by superposition of a number of complex exponentials. Below we will
present the
Discrete-Time Fourier Transform
(DTFT). Because the DTFT deals with nonperiodic
signals, we must nd a way to include all real frequencies in the general equations. For the DTFT we simply
let N go to innity. This will also change the summation over integers to an integration over real numbers.
9.3.2 DTFT synthesis

It can be demonstrated that an arbitrary
N -periodic
Discrete Time-periodic function
f [n]
can be written
as a linear combination of harmonic complex sinusoids
f [n] =
N
1
X
ck ej0 kn
(9.7)
k=0
2
N is the fundamental frequency. For almost all f [n] of practical interest, there exists cn to
make (9.7) true. If f [n] is nite energy, then the equality in (9.7) holds in the sense of energy convergence;
where
0 =
with discrete-time signals, there are no concerns for divergence as there are with continuous-time signals.
4 "Continuous Time
5 "Eigenfunctions of
Complex Exponential" <http://legacy.cnx.org/content/m10060/latest/>

LTI Systems" <http://legacy.cnx.org/content/m10500/latest/>
127
cn - called the Fourier coecients - tell us "how much" of the sinusoid ej0 kn is in f [n]. The formula
f [n] as a sum of complex exponentials, each of which is easily processed by an LTI system (since it
The
shows
ej0 kn , k Z
every LTI system).
form a basis for the space of N-periodic discrete time functions.
9.3.2.1 Equations
Discrete-Time Fourier Transform
X () =
f [n] e(jn)
(9.8)
X () ejn d
(9.9)
n=
Inverse DTFT
1
x [n] =
2
warning: It is not uncommon to see the above formula written slightly dierent. One of the most
common dierences is the way that the exponential is written. The above equations use the radial
in the exponential, where = 2F , but it is also common to include the more

j2F n, in the exponential. Sometimes DTFT notation is expressed as X ej ,
6
that it is not a CTFT (which is denoted as X ()). Click here for an overview of
frequency variable
explicit expression,
to make it clear
the notation used in Connexion's DSP modules.
6 "DSP
notation" <http://legacy.cnx.org/content/m10161/latest/>
128
9.3.3 DTFT Denition demonstration
Click on the above thumbnail image (when online) to download an interactive Mathematica
Player demonstrating Discrete Time Fourier Transform. To Download, right-click and save target as .cdf.
Figure 9.6:
9.3.4 DTFT Summary

Because complex exponentials are eigenfunctions of LTI systems, it is often useful to represent signals using
a set of complex exponentials as a basis. The discrete time Fourier transform synthesis formula expresses a
discrete time, aperiodic function as the innite sum of continuous frequency complex exponentials.
X () =
x [n] e(jn)
(9.10)
X () ejn d
(9.11)
n=
x [n] =
9.4 Properties of the DTFT
1
2
9.4.1 Introduction
8
This module will look at some of the basic properties of the Discrete-Time Fourier Transform
note:
(DTFT).
We will be discussing these properties for aperiodic, discrete-time signals but understand
that very similar properties hold for continuous-time signals and periodic signals as well.

8 "Discrete Time Fourier Transform (DTFT)" <http://legacy.cnx.org/content/m10108/latest/>
129
9.4.2 Table of DTFT Properties

Discrete-Time Fourier Transform Properties
Linearity
Sequence Domain
Frequency Domain
a1 s1 [n] + a2 s2 [n]
a1 S1 () + a2 S2 ()
Conjugate Symmetry
s [n]
S ()
Time Scaling (Expansion)
sc [n] = {
s [n/c]
0
if
n/c
is integer
S (c)
otherwise
Time Reversal
s [n]
S ()
Time Delay
s [n n0 ]
e(jn0 ) S ()
Multiplication by n
ns [n]
P
j dS()
d
Sum
n=
s [n]
S (0)
R
1
Value at Origin
s [0]
Time Convolution
s1 [n] s2 [n]
Frequency Convolution
Parseval's Theorem
s1 [n] s2 [n]
P
2
n= (|s [n] |)
S1 () S2 ()
R
1
2 2 S1 (u) S2 ( u) d
R
2
1
2 (|S () |) d
Complex Modulation
ej0 n s [n]
S ( 0 )
Amplitude Modulation
s [n] cos (0 n)
S(0 )+S(+0 )
2
S(0 )S(+0 )
2j
s [n] sin (0 n)
S () d
Table 9.1: Discrete-time Fourier transform properties and relations.
9.4.3 Discussion of Fourier Transform Properties

9.4.3.1 Linearity
The combined addition and scalar multiplication properties in the table above demonstrate the basic property
of linearity. What you should see is that if one takes the Fourier transform of a linear combination of signals
then it will be the same as the linear combination of the Fourier transforms of each of the individual signals.
This is crucial when using a table
of transforms to nd the transform of a more complicated signal.
Example 9.1
We will begin with the following signal:
z [n] = af1 [n] + bf2 [n]
(9.12)
Now, after we take the Fourier transform, shown in the equation below, notice that the linear
combination of the terms is unaected by the transform.
Z () = aF1 () + bF2 ()
9 "Common
Fourier Transforms" <http://legacy.cnx.org/content/m10099/latest/>
(9.13)
130
9.4.3.2 Symmetry
Symmetry is a property that can make life quite easy when solving problems involving Fourier transforms.
Basically what this property says is that since a rectangular function in time is a sinc function in frequency,
then a sinc function in time will be a rectangular function in frequency. This is a direct result of the similarity
between the forward DTFT and the inverse DTFT. The only dierence is the scaling by
and a frequency
reversal.

This property deals with the eect on the frequency-domain representation of a signal if the time variable
is altered. The most important concept to understand for the time scaling property is that signals that are
narrow in time will be broad in frequency and
10
a unit pulse
with a
very
vice versa.
The simplest example of this is a delta function,
small duration, in time that becomes an innite-length constant function in
frequency.
In contrast to the CTFT property, the DTFT time scaling property is available only for expansion in
the time domain. This is because decimation discards samples of the original signal and therefore there is
no unique relationship between the original signal and a decimated signal (that is, a decimated signal could
correspond to many original signals) that would provide a single transformation between the original DTFT
and the "decimated" DTFT. The intuition from CTFT still holds for expansion: expanding the signal in
time compacts the DTFT in frequency.

Time shifting shows that a shift in time is equivalent to a linear phase shift in frequency. Since the frequency
content depends only on the shape of a signal, which is unchanged in a time shift, then only the phase
spectrum will be altered. This property is proven below:
Example 9.2
We will begin by letting
z [n] = f [n ].
z [n].
Now let us take the Fourier transform with the previous
expression substituted in for
Z () =
f [n ] e(jn)
(9.14)
n=
Now let us make a simple change of variables, where
= n .
Through the calculations below,
you can see that only the variable in the exponential is altered thus only changing the phase in the
frequency domain.
Z ()
=
=
(j(+))
= f [] e
P
e(j) = f [] e(j)
(j)
= e
(9.15)
F ()
9.4.3.5 Convolution
Convolution is one of the big reasons for converting signals to the frequency domain, since convolution in
time becomes multiplication in frequency.
This property is also another excellent example of symmetry
between time and frequency. It also shows that there may be little to gain by changing to the frequency
domain when multiplication in time is involved.
10 "Elemental
Signals": Section Pulse <http://legacy.cnx.org/content/m0004/latest/#pulsedef>
131
We will introduce the convolution integral here, but if you have not seen this before or need to refresh
your memory, then look at the discrete-time convolution
11
module for a more in depth explanation and
derivation.
= f1 [n] f2 [n]
P
=
= f1 [] f2 [n ]
y [n]
(9.16)
9.4.3.6 Time Dierentiation

12
Since LTI
systems can be represented in terms of dierential equations, it is apparent with this property
that converting to the frequency domain may allow us to convert these complicated dierential equations to
simpler equations involving multiplication and addition. This is often looked at in more detail during the
study of the Z Transform
13
9.4.3.7 Parseval's Relation
(|f [n] |) =
(|F () |) d
(9.17)
n=
Parseval's relation tells us that the energy of a signal is equal to the energy of its Fourier transform.
Figure 9.7
9.4.3.8 Modulation (Frequency Shift)

Modulation is absolutely imperative to communications applications. Being able to shift a signal to a dierent
frequency, allows us to take advantage of dierent parts of the electromagnetic spectrum is what allows us
to transmit television, radio and other applications through the same space without signicant interference.
The proof of the frequency shift property is very similar to that of the time shift (Section 9.4.3.4: Time
Shifting); however, here we would use the inverse Fourier transform in place of the Fourier transform. Since
we went through the steps in the previous, time-shift proof, below we will just show the initial and nal step
to this proof:
z [n] =
1
2
F ( ) ejn d
(9.18)
z [n] = f [n] ejn

11 "Discrete Time Convolution" <http://legacy.cnx.org/content/m10087/latest/>
12 "System Classications and Properties" <http://legacy.cnx.org/content/m10084/latest/>
13 "The Laplace Transform" <http://legacy.cnx.org/content/m10110/latest/>
(9.19)
132

The following online resources provide interactive explanations of the properties of the CTFT:
Discrete-Time Fourier Transform Properties (Johns Hopkins University)
14
9.4.5 Properties Demonstration

An interactive example demonstration of the properties is included below:
<CTFTSPlab.llb>
Interactive Signal Processing Laboratory Virtual Instrument created using NI's Labview.
Figure 9.8:
9.5 Common Discrete Time Fourier Transforms
15
9.5.1 Common DTFTs

Time Domain
x [n]
j0 n
Frequency Domain
X ()
( 2m)
( 0 2m)
m=
m=
[n]
[n M ]
P
m= [n M m]
ejM
P
ejM m
=
m=

P
1
k
k=
M
2
M
P
1
+
( + 2k)
k=
1ej
u [n]
an u [n]
an u [ (n + 1)]
a|n|
nan u [n]
sin (an)
cos (an)
c 2
2
sinc2
c n
Notes
real number
integer
integer
1
if |a| < 1
1aej
1
if |a| > 1
1aej
1a2
if |a| < 1
12acos()+a2
aej
if |a| < 1
(ej a)2
P
j m= [ ( + a 2m) (real
a number
2m)]a
P
m= [ ( a 2m) + ( +
real
a number
2m)] a

P
2m
real number c ,
m=
2c
0 < c
14 http://www.jhu.edu/signals/dtftprops/indexDTFTprops.htm
133
sinc
c n
m= p
u [n] u [n M ]
c
(n+a) {cos [c
(n + a)]
sinc [c (n + a)]}
n
1
n2
[(1) 1]
0
(1)n
n
2m
2c
real number
sin(M/2) j(M 1)/2

e
sin(/2)

j p
eja
c
integer
c , 0 < c
real number
c , 0 < c 1
||
n=0
dierentiator lter
elsewhere
n odd
2
n
n even
<0
{ 0
=0
Hilbert Transform
>0
Table 9.2
Notes
p (t)
is the pulse function for arbitrary real-valued
p (t) = {
(t)
t.
0
if
|t| > 1/2
if
|t| 1/2
is the triangle function for arbitrary real-valued
(9.20)
t.
1+t
if
(t) = { 1 t
1t0
if
0<t1
(9.21)
otherwise
9.6 Discrete Time Convolution and the DTFT
16
9.6.1 Introduction
This module discusses convolution of discrete signals in the time and frequency domains.
9.6.2 The Discrete-Time Convolution

9.6.2.1 Discrete Time Fourier Transform
The DTFT transforms an innite-length discrete signal in the time domain into an nite-length (or
2 -
periodic) continuous signal in the frequency domain.
DTFT
X () =
x [n] e(jn)
(9.22)
X () ejn d
(9.23)
n=
Inverse DTFT
x [n] =
16 This
1
2

134
9.6.2.2 Demonstration
Interact (when online) with a Mathematica CDF demonstrating the Discrete Convolution.
To Download, right-click and save as .cdf.
Figure 9.9:
135
9.6.2.3 Convolution Sum

As mentioned above, the convolution sum provides a concise, mathematical way to express the output of
an LTI system based on an arbitrary discrete-time input signal and the system's impulse response.
convolution sum is expressed as
y [n] =
x [k] h [n k]
The
(9.24)
k=
As with continuous-time, convolution is represented by the symbol *, and can be written as
y [n] = x [n] h [n]
(9.25)
Convolution is commutative. For more information on the characteristics of convolution, read about the
Properties of Convolution (Section 4.4).
9.6.2.4 Convolution Theorem

Let
and
be two functions with convolution
f g.
Let
be the Fourier transform operator. Then
F (f g) = F (f ) F (g)
(9.26)
F (f g) = F (f ) F (g)
(9.27)
By applying the inverse Fourier transform
F 1 ,
we can write:
f g = F 1 (F (f ) F (g))
(9.28)
9.6.3 Conclusion
The Fourier transform of a convolution is the pointwise product of Fourier transforms.
In other words,
convolution in one domain (e.g., time domain) corresponds to point-wise multiplication in the other domain
(e.g., frequency domain).
136
Chapter 10
Computing Fourier Transforms
10.1 Discrete Fourier Transform (DFT)
10.1.1 Introduction
The discrete-time Fourier transform (DTFT) can be evaluated when we have an analytic expression for the
signal. This is a good match to the digital world, where signals are discrete-time and quantized, and thus
it is crucial to implement transforms like the DTFT in a computer. The formula for the DTFT is a sum,
which conceptually can be easily computed save for two issues.
Signal duration:
The sum extends over the signal's duration, which must be nite to compute the
signal's spectrum.
It is exceedingly dicult to store an innite-length signal in any case, so we'll
assume that the signal extends over
Continuous frequency:
[0, N 1].
Subtler than the signal duration issue is the fact that the frequency variable is
continuous: It may only need to span one period, like
[12, 12]
or
[0, 1],
but the DTFT formula as it
stands requires evaluating the spectra at all frequencies within a period.
10.1.2 Periodic Discrete-Time Signals

As a way to bridge nite and innite-length signals, we can consider the periodization of a signal
xN [n]
of
x [n] = x [n N ]
for all
n.
nite length
N.
The signal is periodized to
x [n]
by repeating its values so that
One can also write this relationship using a convolution with a train of impulses:
x [n]
= xN [n mod N ] = xN [n]
k=
[n kN ] .
(10.1)
Recall that each convolution with a shifted impulse will contribute one copy of the original signal, therefore providing periodization.
denote the DTFTs of
k=
factor of
x [n]
and
Let us consider the eect of periodization in the frequency domain.
xN [n]
by
X ()
= XN () F
X ()
We
respectively. Note also that the impulse train

that has been expanded in time by a

P
[n kN ] = XN () m= (N 2m)

P
= XN () N1 m= 2m
.
N
k=
In words, the DTFT of the periodic signal
1 This
XN (),
[n kN ] can be interpreted as a constant function f [n] = 1

N . Using the properties of the DTFT, we have:
X ()
signal
and
X ()
and a train of impulses with spacing
(10.2)
is equal to the product of the DTFT of the nite-length
2/N .
This means that
X ()
is nonzero only for values

137
CHAPTER 10. COMPUTING FOURIER TRANSFORMS
138
of
that are multiples of
dened by a set of
2/N .
Additionally, recall that a DTFT is
values of the DTFT
2 -periodic;
thus,
X ()
is uniquely
XN ().
Conceptually, we see that periodization in time is reected as sampling in frequency, and that the DTFT
of a
N -periodic signal (completely determined by its N
values in time) is completely determined by
distinct
values of its DTFT. This discrete, nite-length relationship between a signal and its Fourier transform gives
rise to the discrete Fourier transform (DFT).
10.1.3 Discrete Fourier Transform

Following this narrative, we can consider the DFT as a sampling of the DTFT of the nite-length signal
x [n]
with a spacing of
2/N
between samples. Instead of using the DTFT frequency
we will index the
values of the DFT with integers:
X [k] =
PN 1
n=0
x [n] e
j2kn
N
(10.3)
X ()
Similarly, we can dene the inverse transform by plugging in
into the inverse DTFT, which due to
its sampled nature turns the corresponding integral over the range of frequencies
x [n] =
1
N
PN 1
n=0
X [k] e
j2kn
N
[0, 2]
into a sum:
(10.4)
This pair of equations provide the denition for the forward and inverse DFT, respectively.
10.1.4 Computing the Discrete Fourier Transform

One would like to simplify the computation process as much as possible, since DFTs are computed so often
in science and engineering. One step in this direction can be made by realizing that both the forward and
inverse DFT transforms relate to a complex number
the signal
N.
wN = ej2/N
that is dependent only on the length of
With this notation, we can then write the pair of transforms as
X [k] =
PN 1
n=0
kn
x [n] (wN )
, x [n] =
PN 1
k=0
kn
(10.5)
X [k] (wN ) .
One may observe that the equation for these two transforms are very reminiscent of one another: in fact,
X [k] = DF T (x [n]), then one could write

1
DF
T
(X
[k])
- that is, one only need to invert X [k]
N
in time, run it through the DFT, and divide the output by N to obtain an equivalent IDFT. The DFT
if we write the DFT transform as a "system" using the notation
the inverse transform as
x [n] = IDF T (X [k]) =
has many additional useful properties that signicantly simplify it computation and that are leveraged by
modern implementations such as the Fast Fourier Transform (FFT).
10.1.5 Summary
By restricting our attention to nite-length signals, it is possible to dene a version of the Fourier transform
that also has nite length and that can be easily obtained by a computer. This new discrete Fourier transform
is intimately related with the discrete-time Fourier transform and can be interpreted as performing a sampling
operation in the frequency domain.
10.2 DFT: Fast Fourier Transform
We now have a way of computing the spectrum for an arbitrary signal: The Discrete Fourier Transform
(DFT)
computes the spectrum at
equally spaced frequencies from a length-
sequence. An issue that
never arises in analog "computation," like that performed by a circuit, is how much work it takes to perform
2 This content is available online at

3 "Discrete Fourier Transform", (1)
<http://legacy.cnx.org/content/m0504/2.9/>.
: Discrete Fourier transform <http://legacy.cnx.org/content/m0502/latest/#eqn1>
139
the signal processing operation such as ltering. In computation, this consideration translates to the number
of basic computational steps required to perform the needed processing.
the
complexity,
The number of steps, known as
becomes equivalent to how long the computation takes (how long must we wait for an
answer). Complexity is not so much tied to specic computers or programming languages but to how many
steps are required on any computer. Thus, a procedure's stated complexity says that the time taken will be
proportional to some function of the amount of data used in the computation and the amount demanded.
For example, consider the formula for the discrete Fourier transform. For each frequency we chose, we
must multiply each signal value by a complex number and add together the results. For a real-valued signal,
each real-times-complex multiplication requires two real multiplications, meaning we have
2N
multiplications
to perform. To add the results together, we must keep the real and imaginary parts separate. Adding
numbers requires
N 1
2N + 2 (N 1) = 4N 2
computations is N (4N 2).
additions. Consequently, each frequency requires
computational steps. As we have
frequencies, the total number of
basic
In complexity calculations, we only worry about what happens as the data lengths increase, and take the
dominant termhere the
4N 2
termas reecting how much work is involved in making the computation.
As multiplicative constants don't matter since we are making a "proportional to" evaluation, we nd the
DFT is an
O N2
computational procedure. This notation is read "order
N -squared".
Thus, if we double
the length of the data, we would expect that the computation time to approximately quadruple.
Exercise 10.2.1
In making the complexity evaluation for the DFT, we assumed the data to be real. Three questions emerge.
First of all, the spectra of such signals have conjugate symmetry, meaning that
N
2 + 1, ..., N + 1 in the DFT ) can be computed from the

corresponding positive frequency components. Does this symmetry change the DFT's complexity?
negative frequency components (k
Secondly, suppose the data are complex-valued; what is the DFT's complexity now?
Finally, a less important but interesting question is suppose we want
of
N;
frequency values instead
now what is the complexity?
10.3 The Fast Fourier Transform (FFT)
10.3.1 Introduction
The Fast Fourier Transform (FFT) is an ecient O(NlogN) algorithm for calculating DFTs The FFT
symmetries in the
exploits
matrix to take a "divide and conquer" approach. We will rst discuss deriving the
actual FFT algorithm, some of its implications for the DFT, and a speed comparison to drive home the
importance of this powerful algorithm.
10.3.2 Deriving the FFT

To derive the FFT, we assume that the signal's duration is a power of two:
N = 2l
. Consider what happens
to the even-numbered and odd-numbered elements of the sequence in the DFT calculation.
4 "Discrete Fourier Transform", (1) : Discrete Fourier transform <http://legacy.cnx.org/content/m0502/latest/#eqn1>

6 "Fast Fourier Transform (FFT)" <http://legacy.cnx.org/content/m10250/latest/>
140
22k
s [0] + s [2] e(j) N

2(2+1)k
+ s [3] e(j) N
+
2(N 2)k
+ s [N 2] e(j) N
2(N 2+1)k
N
+ s [N 1] e(j)
2 ( N
2 1)k
(j)
(j) 2k
N
N
2
2
s [0]
+
s [2] e
+
+ s [N 2] e
2 ( N
1
k
)
2
(j) 2k
(j)
N
N
s [1] + s [3] e
e (j2k)
2 + + s [N 1] e
2
N
S [k]
=
(j) 2k
N
s [1] e
Each term in square brackets has the
form of a
+
=
(10.6)
N
2 -length DFT. The rst one is a DFT of the evenThe rst DFT is combined with the
numbered elements, and the second of the odd-numbered elements.

second multiplied by the complex exponential
frequency indices
k {0, . . . , N 1} .
(j2k)
N
. The half-length transforms are each evaluated at
Normally, the number of frequency indices in a DFT calculation range
between zero and the transform length minus one. The
computational advantage of the FFT comes from
recognizing the periodic nature of the discrete Fourier transform. The FFT simply reuses the computations
made in the half-length transforms and combines them through additions and the multiplication by
(j2k)
N
N
, which is not periodic over
2 , to rewrite the length-N DFT. Figure 10.1 (Length-8 DFT decomposition)
2
N
N
illustrates this decomposition. As it stands, we now compute two length2 transforms (complexity 2O
4
), multiply one of them by the complex exponential (complexity
O (N )
O (N )
), and add the results (complexity
). At this point, the total complexity is still dominated by the half-length DFT calculations, but the
proportionality coecient has been reduced.

Now for the fun. Because
N = 2l , each of the half-length transforms can be reduced to two quarter-length
transforms, each of these to two eighth-length ones, etc. This decomposition continues until we are left with
length-2 transforms. This transform is quite simple, involving only additions. Thus, the rst stage of the
N
2 length-2 transforms (see the bottom part of Figure 10.1 (Length-8 DFT decomposition)). Pairs
of these transforms are combined by adding one to the other multiplied by a complex exponential. Each pair
N
N
requires 4 additions and 4 multiplications, giving a total number of computations equaling 8
4 = 2 . This
number of computations does not change from stage to stage. Because the number of stages, the number of
FFT has
times the length can be divided by two, equals
log2 N
, the complexity of the FFT is
O (N logN )
141
Length-8 DFT decomposition
(a)
(b)
The initial decomposition of a length-8 DFT into the terms using even- and odd-indexed
inputs marks the rst phase of developing the FFT algorithm. When these half-length transforms are
successively decomposed, we are left with the diagram shown in the bottom panel that depicts the
length-8 FFT computation.
Figure 10.1:
Doing an example will make computational savings more obvious. Let's look at the details of a length-8
DFT. As shown on Figure 10.1 (Length-8 DFT decomposition), we rst decompose the DFT into two length4 DFTs, with the outputs added and subtracted together in pairs. Considering Figure 10.1 (Length-8 DFT
decomposition) as the frequency index goes from 0 through 7, we recycle values from the length-4 DFTs
into the nal calculation because of the periodicity of the DFT output. Examining how pairs of outputs are
collected together, we create the basic computational element known as a
buttery (Figure 10.2 (Buttery)).
142
Buttery
Figure 10.2:
The basic computational element of the fast Fourier transform is the buttery. It takes
two complex numbers, represented by a and b, and forms the quantities shown. Each buttery requires
one complex multiplication and two complex additions.
By considering together the computations involving common output frequencies from the two half-length
DFTs, we see that the two complex multiplies are related to each other, and we can reduce our computational
work even further. By further decomposing the length-4 DFTs into two length-2 DFTs and combining their
outputs, we arrive at the diagram summarizing the length-8 fast Fourier transform (Figure 10.1 (Length-8
DFT decomposition)).
Although most of the complex multiplies are quite simple (multiplying by
e(j)
means negating real and imaginary parts), let's count those for purposes of evaluating the complexity as full
N
2 = 4 complex multiplies and 2N = 16 additions for each stage and
3N
stages, making the number of basic computations
2 log2 N as predicted.
complex multiplies. We have
Exercise 10.3.1
log2 N = 3
Note that the ordering of the input sequence in the two parts of Figure 10.1 (Length-8 DFT
decomposition) aren't quite the same. Why not? How is the ordering determined?
10.3.2.1 FFT and the DFT

We now have a way of computing the spectrum for an arbitrary signal: The Discrete Fourier Transform
(DFT)
computes the spectrum at
equally spaced frequencies from a length-
sequence. An issue that
never arises in analog "computation," like that performed by a circuit, is how much work it takes to perform
the signal processing operation such as ltering. In computation, this consideration translates to the number
of basic computational steps required to perform the needed processing.
the
complexity,
The number of steps, known as
becomes equivalent to how long the computation takes (how long must we wait for an
answer). Complexity is not so much tied to specic computers or programming languages but to how many
steps are required on any computer. Thus, a procedure's stated complexity says that the time taken will be
proportional to some function of the amount of data used in the computation and the amount demanded.
For example, consider the formula for the discrete Fourier transform. For each frequency we chose, we
must multiply each signal value by a complex number and add together the results. For a real-valued signal,
each real-times-complex multiplication requires two real multiplications, meaning we have
2N
multiplications
to perform. To add the results together, we must keep the real and imaginary parts separate. Adding
numbers requires
N 1
2N + 2 (N 1) = 4N 2
computations is N (4N 2).
additions. Consequently, each frequency requires
computational steps. As we have
frequencies, the total number of
basic
In complexity calculations, we only worry about what happens as the data lengths increase, and take the
dominant termhere the
7 "Discrete
4N 2
termas reecting how much work is involved in making the computation.
Fourier Transform", (1) : Discrete Fourier transform <http://legacy.cnx.org/content/m0502/latest/#eqn1>

143
As multiplicative constants don't matter since we are making a "proportional to" evaluation, we nd the
O N2
DFT is an
computational procedure. This notation is read "order
N -squared".
Thus, if we double
the length of the data, we would expect that the computation time to approximately quadruple.
Exercise 10.3.2
In making the complexity evaluation for the DFT, we assumed the data to be real. Three questions emerge.
First of all, the spectra of such signals have conjugate symmetry, meaning that
negative frequency components (k
N
2 + 1, ..., N + 1 in the DFT ) can be computed from the

corresponding positive frequency components. Does this symmetry change the DFT's complexity?
Secondly, suppose the data are complex-valued; what is the DFT's complexity now?
Finally, a less important but interesting question is suppose we want
of
N;
frequency values instead
now what is the complexity?
10.3.3 Speed Comparison

How much better is
O (N logN )
than
O N2
This gure shows how much slower the computation time of an O(NlogN) process grows.
Figure 10.3:
10
100
1000
106
109
N2
100
104
106
1012
1018
N logN
200
3000
6 106
9 109
Table 10.1
Say you have a 1 MFLOP machine (a million "oating point" operations per second). Let
N = 1million =
106 .
An
An

O N 2 algorithm takes 1012 ors 106 seconds ' 11.5
O (N logN ) algorithm takes 6 106 Flors 6 seconds.
note:
N = 1million
is
days.
not unreasonable.
Example 10.1
3106 numbers for each

picture. So for two N
f [n] and h [n]. If computing f [n] ~ h [n] directly: O N 2 operations.
taking FFTs O (N logN )
3 megapixel digital camera spits out
8 "Discrete
point sequences
Fourier Transform", (1) : Discrete Fourier transform <http://legacy.cnx.org/content/m0502/latest/#eqn1>

144
O (N )
O (N logN ).
complexity is O (N logN ).
multiplying FFTs
inverse FFTs
the total
10.3.4 Conclusion
Other "fast" algorithms have been discovered, most of which make use of how many common factors the
transform length N has. In number theory, the number of prime factors a given integer has measures how
composite
it is.
The numbers 16 and 81 are highly composite (equaling
number 18 is less so (
1 2
2 3
), and 17 not at all (it's prime).
24
and
34
respectively), the
In over thirty years of Fourier transform
algorithm development, the original Cooley-Tukey algorithm is far and away the most frequently used. It
is so computationally ecient that power-of-two transform lengths are frequently used regardless of what
the actual length of the data. It is even well established that the FFT, alongside the digital computer, were
almost completely responsible for the "explosion" of DSP in the 60's.
145

When the signal is real-valued, we may only need half the spectral values, but the complexity remains
unchanged. If the data are complex-valued, which demands retaining all frequency values, the complexity is
again the same. When only
frequencies are needed, the complexity is
O (KN).
The upper panel has not used the FFT algorithm to compute the length-4 DFTs while the lower one has.
The ordering is determined by the algorithm.
When the signal is real-valued, we may only need half the spectral values, but the complexity remains
unchanged. If the data are complex-valued, which demands retaining all frequency values, the complexity is
again the same. When only
frequencies are needed, the complexity is
O (KN).
146
Chapter 11
Sampling and Reconstruction
11.1 Signal Sampling
11.1.1 Introduction
Digital computers can process discrete time signals using extremely exible and powerful algorithms. However, most signals of interest are continuous time signals, which is how data almost always appears in nature.
This module introduces the concepts behind converting continuous time signals into discrete time signals
through a process called sampling.
11.1.2 Sampling
Sampling a continuous time signal produces a discrete time signal by selecting the values of the continuous
x with sampling period

x [n] = x (nTs ) . The sampling angular frequency is then given
time signal at evenly spaced points in time. Thus, sampling a continuous time signal
Ts
gives the discrete time signal
by
fs = 1/Ts .
xs
dened by
It should be intuitively clear that multiple continuous time signals sampled at the same rate can produce
the same discrete time signal since uncountably many continuous time functions could be constructed that
connect the points on the graph of any discrete time function. Thus, sampling at a given rate does not result
in an injective relationship. Hence, sampling is, in general, not invertible.
Example 11.1
For instance, consider the signals
x, y
dened by
x (t) =
sin (t)
t
sin (5t)
t
period Ts = /2
y (t) =
and their sampled versions
x, y
with sampling
1 This
(11.2)
sin (n/2)
n/2
(11.3)
sin (n5/2)
.
n/2
(11.4)
x [n] =
y [n] =
(11.1)

147
CHAPTER 11. SAMPLING AND RECONSTRUCTION
148
Notice that since
sin (n5/2) = sin (n2 + n/2) = sin (n/2)
(11.5)
it follows that
y [n] =
Hence,
and
sin (n/2)
= x [n] .
n/2
(11.6)
provide an example of distinct functions with the same sampled versions at a
specic sampling rate.

It is also useful to consider the relationship between the frequency domain representations of the continuous
time function and its sampled versions. Consider a signal
x [n] = x (nTs ).
the discrete time signal
sampled with sampling period
Ts
to produce
Let us dene the continuous-time "impulse train"
p (t) =
(t nTs )
(11.7)
n=
Using Fourier series, it can be shown that
p (t) =
1 X j T2 kt
e s
Ts
(11.8)
k=
Multiplying
x (t)
by the impulse train yields the "continuous-time sampled signal"
xs (t)
xc (t)
= xc (t)
Taking the CTFT of
=
=
=
Xs (f )
P
1
Ts
n= (t nT )
P
2
kt
jT
s
k= e
(11.9)
xs (t),
Xs (f )
Notice that
xs (t):
R

P
2
kt
jT
1
s
x
(t)
e
e(j2f t) dt
c
k=
Ts
R
P
k
1
xc (t) e(j2(f Ts )t) dt
k=
Ts

P
1
k
k= Xc f Ts
Ts
(11.10)
Xc (f ).
We now investigate the
is a summation of scaled and shifted copies of
relationship between the CTFT and the DTFT:
X ()
=
=
=
=
where
xc (t) (t nTs ) e(jn)

X ()
(jn)
n= x [n] e
P
(jn)
n= xc (nTs ) e
R
P
n= xc (t) (t
R
P
n= xc (t) (t
is nonzero if and only if
=
=
=
n=
nTs ) dte(jn)
(11.11)
nTs ) e(jn) dt
t
Ts .
R
(j Tts )
dt
n= xc (t) (t nTs ) e
(j t )
R
P
T
s dt
xc (t) n= (t nTs ) e
R
(j Tts )
x (t) e
dt
s
(11.12)
149
where
xc (t)
(t nTs ) = xs (t)
n=
so that

X () = Xs
2Ts

(11.13)
Plugging (11.10) into (11.13) yields:

2k
1 X
Xc
X () =
Ts
2Ts
(11.14)
k=
Notice that the DTFT of the sampled signal
x [n]
is a summation of scaled, stretched, and shifted copies
xc (t).
of the CTFT of the continuous signal
11.1.3 Sampling Summary

Sampling a continuous time signal produces a discrete time signal by selecting the values of the continuous
time signal at equally spaced points in time. However, we have shown that this relationship is not injective as
multiple continuous time signals can be sampled at the same rate to produce the same discrete time signal.
This is related to a phenomenon called aliasing which will be discussed in later modules.
Consequently,
the sampling process is not, in general, invertible. Nevertheless, as will be shown in the module concerning
reconstruction, the continuous time signal can be recovered from its sampled version if some additional
assumptions hold.
11.2 Sampling Theorem
11.2.1 Introduction
With the introduction of the concept of signal sampling, which produces a discrete time signal by selecting
the values of the continuous time signal at evenly spaced points in time, it is now possible to discuss one
of the most important results in signal processing, the Nyquist-Shannon sampling theorem. Often simply
called the sampling theorem, this theorem concerns signals, known as bandlimited signals, with spectra that
are zero for all frequencies with absolute value greater than or equal to a certain level. The theorem implies
that there is a suciently high sampling rate at which a bandlimited signal can be recovered exactly from its
samples, which is an important step in the processing of continuous time signals using the tools of discrete
time signal processing.
11.2.2 Nyquist-Shannon Sampling Theorem

11.2.2.1 Statement of the Sampling Theorem
The Nyquist-Shannon sampling theorem concerns signals with continuous time Fourier transforms that are
only nonzero on the interval
(B, B).
(B, B)
for some constant
B.
Such a function is said to be bandlimited to
Essentially, the sampling theorem has already been implicitly introduced in the previous module
concerning sampling. Given a continuous time signals

that the spectrum
Xs
of sampled signal
xs
with continuous time Fourier transform
with sampling period
Xs () =
Ts

1 X
2k
X
.
Ts
2Ts
k=
2 This
X,
recall
is given by

(11.15)
150
x is bandlimited to (1/2Ts , 1/2Ts ), the period of Xs centered about the

X scaled in frequency since no aliasing occurs. This is illustrated in Figure 11.1.
(1/2Ts , 1/2Ts ) bandlimited continuous time signals sampled to the same signal, they
It had previously been noted that if

origin has the same form as
Hence, if any two
would have the same continuous time Fourier transform and thus be identical. Thus, for each discrete time
signal there is a unique
(1/2Ts , 1/2Ts ) bandlimited continuous time signal that samples to the discrete
Ts . Therefore, this (1/2Ts , 1/2Ts ) bandlimited signal can be found from
time signal with sampling period
the samples by inverting this bijection.

This is the essence of the sampling theorem. More formally, the sampling theorem states the following. If
x is bandlimited to (B, B), it is completely determined by its samples with sampling rate fs = 2B ,
x can be reconstructed exactly from its samples xs with sampling
rate fs = 2B . The angular frequency 4B is often called the angular Nyquist rate. Equivalently, this can be
stated in terms of the sampling period Ts = 1/fs . If a signal x is bandlimited to (B, B), it is completely
determined by its samples with sampling period Ts = 1/2B . That is to say, x can be reconstructed exactly
from its samples xs with sampling period Ts .
a signal
known as the Nyquist rate. That is to say,
The spectrum of a bandlimited signals is shown as well as the spectra of its samples
at rates above and below the Nyquist frequency. As is shown, no aliasing occurs above the Nyquist
frequency, and the period of the samples spectrum centered about the origin has the same form as the
spectrum of the original signal scaled in frequency. Below the Nyquist frequency, aliasing can occur and
causes the spectrum to take a dierent than the original spectrum.
Figure 11.1:
11.2.2.2 Proof of the Sampling Theorem

The above discussion has already shown the sampling theorem in an informal and intuitive way that could
easily be rened into a formal proof. However, the original proof of the sampling theorem, which will be
151
given here, provides the interesting observation that the samples of a signal with period
Ts
provide Fourier
(1/2Ts , 1/2Ts ).
and xs be its samples
series coecients for the original signal spectrum on

Let
represent
(1/2Ts , 1/2Ts ) bandlimited signal

with sampling period Ts . We can
x in terms of its spectrum X using the inverse continuous time Fourier transfrom and the fact that
be a
is bandlimited. The result is
x (t)
This representation of
R 1/2Ts
1/2Ts
X (f ) ej2f t df
may then be sampled with sampling period
xs [n]
(1/2Ts , 1/2Ts ),
= xs (nTs ) =
xs [n]
Noticing that this indicates that

the interval
is the
nth
R 1/2Ts
(11.16)
Ts
to produce
X (f ) ej2f nTs df
1/2Ts
(11.17)
X (f )
X (f ) and,
continuous time Fourier series coecient for
it is shown that the samples determine the original spectrum
on
by
extension, the original signal itself.
11.2.2.3 Perfect Reconstruction

Another way to show the sampling theorem is to derive the reconstruction formula that gives the original
signal
x
=x
from its samples
xs
Ts ,
provided
is bandlimited to
(1/2Ts , 1/2Ts ).
This is done in the module on perfect reconstruction. However, the result, known as the Whittaker-Shannon
reconstruction formula, will be stated here. If the requisite conditions hold, then the perfect reconstruction
is given by
x (t) =
xs [n] sinc (t/Ts n)
(11.18)
n=
where the sinc function is dened as
sinc (t) =
sin (t)
.
t
(11.19)
From this, it is clear that the set
{sinc (t/Ts n) |n Z}
forms an orthogonal basis for the set of
(1/2Ts , 1/2Ts )
(1/2Ts , 1/2Ts )
(11.20)
bandlimited signals, where the coecients of a
signal in this basis are its samples with sampling period
Ts .
11.2.3 Practical Implications

11.2.3.1 Discrete Time Processing of Continuous Time Signals
The Nyquist-Shannon Sampling Theorem and the Whittaker-Shannon Reconstruction formula enable discrete time processing of continuous time signals. Because any linear time invariant lter performs a multiplication in the frequency domain, the result of applying a linear time invariant lter to a bandlimited signal
is an output signal with the same bandlimit. Since sampling a bandlimited continuous time signal above
the Nyquist rate produces a discrete time signal with a spectrum of the same form as the original spectrum,
a discrete time lter could modify the samples spectrum and perfectly reconstruct the output to produce
the same result as a continuous time lter. This allows the use of digital computing power and exibility to
be leveraged in continuous time signal processing as well. This is more thouroughly described in the nal
module of this chapter.
152
11.2.3.2 Psychoacoustics
The properties of human physiology and psychology often inform design choices in technologies meant for
interactin with people. For instance, digital devices dealing with sound use sampling rates related to the
frequency range of human vocalizations and the frequency range of human auditory sensativity.
Because
most of the sounds in human speech concentrate most of their signal energy between 5 Hz and 4 kHz, most
telephone systems discard frequencies above 4 kHz and sample at a rate of 8 kHz. Discarding the frequencies
greater than or equal to 4 kHz through use of an anti-aliasing lter is important to avoid aliasing, which
would negatively impact the quality of the output sound as is described in a later module. Similarly, human
hearing is sensitive to frequencies between 20 Hz and 20 kHz. Therefore, sampling rates for general audio
waveforms placed on CDs were chosen to be greater than 40 kHz, and all frequency content greater than
or equal to some level is discarded. The particular value that was chosen, 44.1 kHz, was selected for other
reasons, but the sampling theorem and the range of human hearing provided a lower bound for the range of
choices.
11.2.4 Sampling Theorem Summary

The Nyquist-Shannon Sampling Theorem states that a signal bandlimited to
structed exactly from its samples with sampling period
Ts .
(1/2Ts , 1/2Ts )
can be recon-
The Whittaker-Shannon interpolation formula,
which will be further described in the section on perfect reconstruction, provides the reconstruction of the
unique
(1/2Ts , 1/2Ts ) bandlimited continuous time signal that samples to a given discrete time signal
Ts . This enables discrete time processing of continuous time signals, which has many
powerful applications.
11.3 Signal Reconstruction
11.3.1 Introduction
The sampling process produces a discrete time signal from a continuous time signal by examining the value
of the continuous time signal at equally spaced points in time. Reconstruction, also known as interpolation,
attempts to perform an opposite process that produces a continuous time signal coinciding with the points
of the discrete time signal. Because the sampling process for general sets of signals is not invertible, there
are numerous possible reconstructions from a given discrete time signal, each of which would sample to that
signal at the appropriate sampling rate. This module will introduce some of these reconstruction schemes.
11.3.2 Reconstruction
11.3.2.1 Reconstruction Process
The process of reconstruction, also commonly known as interpolation, produces a continuous time signal
that would sample to a given discrete time signal at a specic sampling rate. Reconstruction can be mathematically understood by rst generating a continuous time impulse train
ximp (t) =
xs [n] (t nTs )
(11.21)
n=
from the sampled signal
xs with sampling period Ts and then applying a lowpass lter G that satises certain
x
. If G has impulse response g , then the result of the reconstruction
conditions to produce an output signal
3 This
153
process, illustrated in Figure 11.2, is given by the following computation, the nal equation of which is used
to perform reconstruction in practice.
= (ximp g) (t)
x
(t)
R
= ximp ( ) g (t ) d
R P
= n= xs [n] ( nTs ) g (t ) d
R
P
= n= xs [n] ( nTs ) g (t ) d
P
= n= xs [n] g (t nTs )
Figure 11.2:
(11.22)
Block diagram of reconstruction process for a given lowpass lter G.
11.3.2.2 Reconstruction Filters

In order to guarantee that the reconstructed signal
was reconstructed using the sampling period
Ts ,
samples to the discrete time signal
the lowpass lter
can be expressed well in the time domain in terms of a condition on the impulse response
lter
G.
xs
from which it
must satisfy certain conditions. These
of the lowpass
The sucient condition to be a reconstruction lters that we will require is that, for all
g (nTs ) = {
This means that
that sampling
sampled at a rate
n=0
n 6= 0
Ts produces a
Ts results in
x
(nTs )
(11.23)
discrete time unit impulse signal. Therefore, it follows
m=
= [n] .
n Z,
xs [m] g (nTs mTs )
m= xs [m] g ((n m) Ts )
P
m= xs [m] [n m]
(11.24)
= xs [n] ,
which is the desired result for reconstruction lters.
11.3.2.3 Cardinal Basis Splines

Since there are many continuous time signals that sample to a given discrete time signal, additional constraints are required in order to identify a particular one of these.
For instance, we might require our
reconstruction to yield a spline of a certain degree, which is a signal described in piecewise parts by polynomials not exceeding that degree. Additionally, we might want to guarantee that the function and a certain
number of its derivatives are continuous.
154
This may be accomplished by restricting the result to the span of sets of certain splines, called basis
splines or B-splines. Specically, if a
nth
degree spline with continuous derivatives up to at least order
is required, then the desired function for a given
Ts
belongs to the span of
{Bn (t/Ts k) |k Z}
Bn = B0 Bn1
for
n1
n1
where
(11.25)
and
B0 (t) = {
1/2 < t < 1/2
otherwise
(11.26)
The basis splines Bn are shown in the above plots. Note that, except for the order 0 and
order 1 functions, these functions do not satisfy the conditions to be reconstruction lters. Also notice
that as the order increases, the functions approach the Gaussian function, which is exactly B .
Figure 11.3:
Bn do not satisfy the conditions to be a reconstruction lter for n 2 as is

Bn are useful in dening the cardinal basis splines, which do satisfy the
conditions to be reconstruction lters. If we let bn be the samples of Bn on the integers, it turns out that bn
1
1
has an inverse bn with respect to the operation of convolution for each n. This is to say that bn bn = .
The cardinal basis spline of order n for reconstruction with sampling period Ts is dened as
However, the basis splines
shown in Figure 11.3.
Still, the
n (t) =
b1
n [k] Bn (t/Ts k) .
k=
(11.27)
155
In order to conrm that this satises the condition to be a reconstruction lter, note that
n (mTs ) =

1
b1
n [k] Bn (m k) = bn bn (m) = (m) .
(11.28)
k=
n is a valid reconstruction lter. Since n is an nth degree spline with continuous derivatives up
n 1, the result of the reconstruction will be a nth degree spline with continuous derivatives up
n 1.
Thus,
order
order
to
to
Figure 11.4: The above plots show cardinal basis spline functions 0 , 1 , 2 , and . Note that the
functions satisfy the conditions to be reconstruction lters. Also, notice that as the order increases, the
cardinal basis splines approximate the sinc function, which is exactly . Additionally, these lters are
acausal.
The lowpass lter with impulse response equal to the cardinal basis spline
of order 0 is one of the
simplest examples of a reconstruction lter. It simply extends the value of the discrete time signal for half
the sampling period to each side of every sample, producing a piecewise constant reconstruction. Thus, the
result is discontinuous for all nonconstant discrete time signals.
Likewise, the lowpass lter with impulse response equal to the cardinal basis spline
1 of order 1 is another
of the simplest examples of a reconstruction lter. It simply joins the adjacent samples with a straight line,
producing a piecewise linear reconstruction.
In this way, the reconstruction is continuous for all possible
discrete time signals. However, unless the samples are collinear, the result has discontinuous rst derivatives.
In general, similar statements can be made for lowpass lters with impulse responses equal to cardinal
basis splines of any order.
Using the
nth
order cardinal basis spline
n ,
the result is a piecewise degree
polynomial. Furthermore, it has continuous derivatives up to at least order
samples are points on a polynomial of degree at most
n,
the derivative of order
n 1. However, unless
n will be discontinuous.
all
156
Reconstructions of the discrete time signal given in Figure 11.5 using several of these lters are shown
in Figure 11.6. As the order of the cardinal basis spline increases, notice that the reconstruction approaches
that of the innite order cardinal spline
the sinc function. As will be shown in the subsequent section
on perfect reconstruction, the lters with impulse response equal to the sinc function play an especially
important role in signal processing.
The above plot shows an example discrete time function. This discrete time function will
be reconstructed using sampling period Ts using several cardinal basis splines in Figure 11.6.
Figure 11.5:
157
The above plots show interpolations of the discrete time signal given in Figure 11.5 using
lowpass lters with impulse responses given by the cardinal basis splines shown in Figure 11.4. Notice
that the interpolations become increasingly smooth and approach the sinc interpolation as the order
increases.
Figure 11.6:
11.3.3 Reconstruction Summary

Reconstruction of a continuous time signal from a discrete time signal can be accomplished through several
schemes.
However, it is important to note that reconstruction is not the inverse of sampling and only
produces one possible continuous time signal that samples to a given discrete time signal.
As is covered
in the subsequent module, perfect reconstruction of a bandlimited continuous time signal from its sampled
version is possible using the Whittaker-Shannon reconstruction formula, which makes use of the ideal lowpass
lter and its sinc function impulse response, if the sampling rate is suciently high.
11.4 Perfect Reconstruction
11.4.1 Introduction
If certain additional assumptions about the original signal and sampling rate hold, then the original signal
can be recovered exactly from its samples using a particularly important type of lter. More specically, it
will be shown that if a bandlimited signal is sampled at a rate greater than twice its bandlimit, the WhittakerShannon reconstruction formula perfectly reconstructs the original signal. This formula makes use of the
ideal lowpass lter, which is related to the sinc function. This is extremely useful, as sampled versions of
4 This

158
continuous time signals can be ltered using discrete time signal processing, often in a computer. The results
may then be reconstructed to produce the same continuous time output as some desired continuous time
system.
11.4.2 Perfect Reconstruction

In order to understand the conditions for perfect reconstruction and the lter it employs, consider the
following.
As a beginning, a sucient condition under which perfect reconstruction is possible will be
discussed. Subsequently, the lter and process used for perfect reconstruction will be detailed.
Recall that the sampled version
xs
of a continuous time signal
x with sampling period Ts
has a spectrum
given by
Xs () =

2k
1 X
X
.
Ts
2Ts
(11.29)
k=
As before, note that if x is bandlimited to (1/2Ts , 1/2Ts ), meaning that X is only nonzero on
(1/2Ts , 1/2Ts ), then each period of Xs has the same form as X . Thus, we can identify the original
spectrum X from the spectrum of the samples Xs and, by extension, the original signal x from its samples
xs at rate Ts if x is bandlimited to (1/2Ts , 1/2Ts ).
If a signal x is bandlimited to (B, B), then it is also bandlimited to (fs /2, fs /2) provided that Ts <
1/2B . Thus, if we ensure that x is sampled to xs with suciently high sampling frequency fs = 1/Ts > 2B
and have a way of identifying the unique (fs /2, fs /2) bandlimited signal corresponding to a discrete time
signal at sampling period Ts , then xs can be used to reconstruct x
= x exactly. The frequency 2B is known
as the Nyquist rate. Therefore, the condition that the sampling rate fs = 1/Ts > 2B be greater than the
Nyquist rate is a sucient condition for perfect reconstruction to be possible.
The correct lter must also be known in order to perform perfect reconstruction.
The ideal lowpass
G (f ) = Ts (u (f + fs /2) u (f fs /2)), which is shown in Figure 11.7, removes all signal

content not in the frequency range (fs /2, fs /2). Therefore, application of this lter to the impulse train
P
n= xs [n] (t nTs ) results in an output bandlimited to (fs /2, fs /2).
We now only need to conrm that the impulse response g of the lter G satises our sucient condition
to be a reconstruction lter. The inverse Fourier transform of G (f ) is
lter dened by
g (t) = sinc (t/Ts ) = {
t=0
sin(t/Ts )
t/Ts
t 6= 0
(11.30)
which is shown in Figure 11.7. Hence,
g (nTs ) = sinc (n) = {

Therefore, the ideal lowpass lter
n=0
sin(n)
n
n 6= 0
n= xs [n] (t nTs )
n 6= 0
= [n] .
(11.31)
(fs /2, fs /2),
this lter always produces the unique
Ts
when
is input.
Therefore, we can always reconstruct any
Ts
n=0
bandlimited signal that samples to a given discrete time sequence at sampling period
the impulse train

period
is a valid reconstruction lter. Since it is a valid reconstruction lter
and always produces an output that is bandlimited to
(fs /2, fs /2)
={
(fs /2, fs /2)
bandlimited signal from its samples at sampling
by the formula
x (t) =
xs [n] sinc (t/Ts n) .
(11.32)
n=
This perfect reconstruction formula is known as the Whittaker-Shannon interpolation formula and is sometimes also called the cardinal series. In fact, the sinc function is the innite order cardinal basis spline
159
Consequently, the set
{sinc (t/Ts n) |n Z} forms a basis for the vector space of (fs /2, fs /2) bandlimited
signals where the signal samples provide the corresponding coecients. It is a simple exercise to show that
this basis is, in fact, an orthogonal basis.
Figure 11.7:
The above plots show the ideal lowpass lter and its inverse Fourier transform, the sinc
Figure 11.8:
The plots show an example discrete time signal and its Whittaker-Shannon sinc recon-
function.
struction.
11.4.3 Perfect Reconstruction Summary

This module has shown that bandlimited continuous time signals can be reconstructed exactly from their
samples provided that the sampling rate exceeds the Nyquist rate, which is twice the bandlimit.
The
Whittaker-Shannon reconstruction formula computes this perfect reconstruction using an ideal lowpass lter,
with the resulting signal being a sum of shifted sinc functions that are scaled by the sample values. Sampling
below the Nyquist rate can lead to aliasing which makes the original signal irrecoverable as is described in
the subsequent module.
The ability to perfectly reconstruct bandlimited signals has important practical
implications for the processing of continuous time signals using the tools of discrete time signal processing.
160
11.5 Aliasing Phenomena
11.5.1 Introduction
Through discussion of the Nyquist-Shannon sampling theorem and Whittaker-Shannon reconstruction formula, it has already been shown that a
at rate
fs = 1/Ts
(B, B) continuous time signal can be reconstructed from its samples

fs > 2B . Now, this module will investigate a problematic
via the sinc interpolation lter if
phenomenon, called aliasing, that can occur if this sucient condition for perfect reconstruction does not
hold. When aliasing occurs the spectrum of the samples has dierent form than the original signal spectrum,
so the samples cannot be used to reconstruct the original signal through Whittaker-Shannon interpolation.
11.5.2 Aliasing
Aliasing occurs when each period of the spectrum of the samples does not have the same form as the spectrum
of the original signal. Given a continuous time signals
that the spectrum
Xs
of sampled signal
xs
with continuous time Fourier transform
Ts
X,
recall
is given by

2k
1 X
X
.
Xs () =
Ts
2Ts
(11.33)
k=
As has already been mentioned several times, if

the same form as
X.
However, if
x is bandlimited to (fs /2, fs /2) then

eachperiod of Xs
x is not bandlimited to (fs /2, fs /2), then the X
2k
2Ts
has
can overlap and
sum together. This is illustrated in Figure 11.9 in which sampling above the Nyquist frequency produces
a samples spectrum of the same shape as the original signal, but sampling below the Nyquist frequency
produces a samples spectrum with very dierent shape. Whittaker-Shannon interpolation of each of these
sequences produces dierent results. The low frequencies not aected by the overlap are the same, but there
is noise content in the higher frequencies caused by aliasing. Higher frequency energy masquerades as low
energy content, a highly undesirable eect.
5 This
161
The spectrum of a bandlimited signals is shown as well as the spectra of its samples
at rates above and below the Nyquist frequency. As is shown, no aliasing occurs above the Nyquist
frequency, and the period of the samples spectrum centered about the origin has the same form as the
spectrum of the original signal scaled in frequency. Below the Nyquist frequency, aliasing can occur and
causes the spectrum to take a dierent than the original spectrum.
Figure 11.9:
Unlike when sampling above the Nyquist frequency, sampling below the Nyquist frequency does not yield
(B, B) bandlimited continuous time signals to the discrete time

x with spectrum X which overlaps and sums to Xs samples to xs . It should be intuitively
clear that there are very many (B, B) bandlimited signals that sample to a given discrete time signal below
an injective (one-to-one) function from the

signals. Any signal
the Nyquist frequency, as is demonstrated in Figure 11.10. It is quite easy to construct uncountably innite
families of such signals.
Aliasing obtains it name from the fact that multiple, in fact innitely many,
sample to the same discrete sequence if
fs < 2B .
(B, B) bandlimited signals
Thus, information about the original signal is lost in this
noninvertible process, and these dierent signals eectively assume the same identity, an alias. Hence, under
these conditions the Whittaker-Shannon interpolation formula will not produce a perfect reconstruction of
the original signal but will instead give the unique
(fs /2, fs /2)
bandlimited signal that samples to the
discrete sequence.
162
The spectrum of a discrete time signal xs , taken from Figure 11.9, is shown along with
the spectra of three (B, B) signals that sample to it at rate s < 2B . From the sampled signal alone,
it is impossible to tell which, if any, of these was sampled at rate s to produce xs . In fact, there are
innitely many (B, B) bandlimited signals that sample to xs at a sampling rate below the Nyquist rate.
Figure 11.10:

The following online resources provide interactive explanations of sampling, reconstruction, and aliasing:
Sampling and Reconstruction with Sound Output
Sampling and Reconstruction (Rice University)
7
8
An Introduction to Sampling (University of Houston)
6 http://cwx.prenhall.com/bookbind/pubbooks/cyganski/chapter0/medialib/SAMPLING_RECONS_SOUND/sampling.html
7 http://www.ece.rice.edu/dsp/courses/elec301/demos/applets/Reconst/index.html
8 http://www2.egr.uh.edu/glover/applets/Sampling/Sampling.html
163
11.5.4 Aliasing Demonstration
Figure 11.11: Interact (when online) with a Mathematica CDF demonstrating sampling and aliasing
for a sinusoid. To Download, right-click and save target as .cdf.
11.5.5 Aliasing Summary

Aliasing, essentially the signal processing version of identity theft, occurs when each period of the spectrum
of the samples does not have the same form as the spectrum of the original signal.
there can be innitely many
fs = 1/Ts < 2B
that samples to xs ,
(B, B)
As has been shown,
xs
(B, B) bandlimited
of xs , at rate fs 2B
bandlimited signals that sample to a given discrete time signal
at a rate
below the Nyquist frequency. However, there is a unique
signal
which is given by the Whittaker-Shannon interpolation
as no aliasing occurs above the Nyquist frequency. Unfortunately, suciently high sampling rates cannot
always be produced. Aliasing is detrimental to many signal processing applications, so in order to process
continuous time signals using discrete time tools, it is often necessary to nd ways to avoid it other than
increasing the sampling rate. Thus, anti-aliasing lters, are of practical importance.
11.6 Anti-Aliasing Filters
11.6.1 Introduction
It has been shown that a
rate
fs = 1/Ts 2B .
(B, B)
bandlimited signal can be perfectly reconstructed from its samples at a
However, it is not always practically possible to produce suciently high sampling
rates or to ensure that the input is bandlimited in real situations.
Aliasing, which manifests itself as a
dierence in shape between the periods of the samples signal spectrum and the original spectrum, would
occur without any further measures to correct this.
energy at frequencies above
fs /2
Thus, it often becomes necessary to lter out signal
in order to avoid the detrimental eects of aliasing.
This is the role of
the anti-aliasing lter, a lowpass lter applied before sampling to ensure that the signal is
(fs /2, fs /2)
bandlimited or at least nearly so.
11.6.2 Anti-Aliasing Filters

Aliasing can occur when a signal with energy at frequencies other that
(B, B)
is sampled at rate
fs < 2B .
Thus, when sampling below the Nyquist frequency, it is desirable to remove as much signal energy outside
the frequency range
(B, B) as possible while keeping as much signal energy in the frequency range (B, B)
fs /2 would be the optimal anti-
as possible. This suggests that the ideal lowpass lter with cuto frequency
aliasing lter to apply before sampling. While this is true, the ideal lowpass lter can only be approximated
in real situations.
9 This
164
In order to demonstrate the importance of anti-aliasing lters, consider the calculation of the error
energy between the original signal and its Whittaker-Shannon reconstruction from its samples taken with
and without the use of an anti-aliasing lter.
ltered signal where
Let
be the original signal and
fs /2.
is the ideal lowpass lter with cuto frequency
y = Gx
be the anti-alias
It is easy to show that the
reconstructed spectrum using no anti-aliasing lter is given by
(f ) = {
X
Ts Xs (Ts f ) |f | < fs /2
0
P
={
k=
X (f kfs ) |f | < fs /2
0
otherwise
otherwise
(11.34)
Thus, the reconstruction error spectrum for this case is
P

k=1 (X (f + kfs ) + X (f kfs )) |f | < fs /2
X X (f ) = {
.
X (f )
otherwise
(11.35)
Similarly, the reconstructed spectrum using the ideal lowpass anti-aliasing lter is given by
Y (f ) = Y (f ) = {
X (f ) |f | < fs /2
0
(11.36)
(11.37)
otherwise
Thus, the reconstruction error spectrum for this case is

X Y (f ) = {
Hence, by Parseval's theorem, it follows that

identical to that of the original signal
|f | < fs /2
X (f )
otherwise
||x y|| ||x x

||. Also note that the spectrum of Y
(fs /2, fs /2). This is graphically shown
in the frequency range
Figure 11.12.
is
in
165
The gure above illustrates the use of an anti-aliasing lter to improve the process of
sampling and reconstruction when using a sampling frequency below the Nyquist frequency. Notice that
when using an ideal lowpass anti-aliasing lter, the reconstructed signal spectrum has the same shape
as the original signal spectrum for all frequencies below half the sampling rate. This results in a lower
error energy when using the anti-aliasing lter, as can be seen by comparing the error spectra shown.
Figure 11.12:
11.6.3 Anti-Aliasing Filters Summary

As can be seen, anti-aliasing lters ensure that the signal is
(fs /2, fs /2)
bandlimited, or at least nearly so.
The optimal anti-aliasing lter would be the ideal lowpass lter with cuto frequency at
fs /2,
which would
ensure that the original signal spectrum and the reconstructed signal spectrum are equal on the interval
(fs /2, fs /2).
However, the ideal lowpass lter is not possible to implement in practice, and approximations
must be accepted instead.
Anti-aliasing lters are an important component of systems that implement
discrete time processing of continuous time signals, as will be shown in the subsequent module.
11.7 Changing Sampling Rates in Discrete Time
10
Up to this point, we have discussed the connection between continuous-time and discrete-time signals that
are captured by the concepts of sampling and reconstruction. In particular cases, we might be interested in
observing a signal under a variety of sampling rates. For example, the amount of memory or communication
bandwidth available for transmission or storage of a discrete-time signal might uctuate in time, which may
require increasing or decreasing the sampling rate (or sampling period) accordingly.
10 This
166
Changing the sampling frequency by reconstructing and sampling always works, but
sometimes it may be possible to do so working only in the discrete-time domain.
Figure 11.13:
Naively, if we have sampled the signal suciently often to be able to recover it (according to the Nyquist
criterion), then we can always return from the discrete-time signal to a continuous-time signal using reconstruction and then sample the signal at the new desired sampling rate.However, there are specic cases
where it is possible to modify the sampling rate of the signal without having to switch back to a continuoustime representation.
In other words, certain changes of sampling rate can be performed directly on the
discrete-time signal. We discuss three specic cases: downsampling, upsampling, and rational scaling.
11.7.1 Downsampling
fs and are asked to reduce the sampling frequency
fs
. When this change is translated to the sampling period T (as
k
1/T ), it is easy to see that the new sampling period T ' = kT will be k times larger than its original
Consider the case where we start with a sampling frequency

by an integer factor to a new value
fs =
fs' =
value. Therefore, the change in sampling frequency can be accounted for by taking the existing discrete signal
x [n]
(sampled at frequency
fs )
and decimating it by a factor of
re-sampled with sampling frequency
to obtain the new signal
x' [n] = x [kn]
'
fs .
We know that for both the old and new sampling frequencies the discrete-time Fourier transform of
the sampled signal will correspond to a periodized, frequency-scaled version of the continuous-time signal's
Fourier transform
to
= 2 .
XCT (f )
where the respective periods/sampling frequencies
fs
and
fs'
each gets mapped
We now compare how these two discrete-time transforms match to one another:

s
x [n] ejn = XCT f
,
2

P
s
= n= x [kn] ejn = XCT f
k2 .
(11.38)
By connecting the two equations through the right-most terms, it is easy to see that
X ' () = X (/k),
X ()
X ' ()
=
=
n=
n=
x' [n] ejn
i.e., that the downsampling performed expands the DTFT of the original signal
since we are still working with a discrete-time signal
x' [n],
x [n].
Note, however, that
the new transform must remain
2 -periodic,
and
so this expansion occurs for each copy of the spectrum around its center, but the copies stay stationary at
multiples of
2 .
167
Figure 11.14:
The downsampling operation in time, continuous frequency, and discrete frequency.
Note also that this result eectively provides us with a new property for the DTFT: decimation in the
time domain corresponds to a qualied expansion in the frequency domain, where the expansion is around
the center of each copy of the CT spectrum (
= 0, 2, 4, ...).
Finally, notice that in downsampling there is the risk that aliasing may occur when the new frequency
'
fs'
does not hold the Nyquist frequency criterion (fs 2f0 , where f0 is the bandwidth of the CT signal).
Noticeably, this is the rst time that we observe the possibility of aliasing directly in the discrete-time
domain.
Since aliasing may occur, it is good engineering practice to apply a (discrete-time) anti-aliasing
lter to the signal before decimation so that aliasing is prevented.

low-pass lter with cuto frequency
c = 2/k .
Such a lter will ideally be a perfect
The combination of an anti-aliasing lter and a decimator
is known in the community as a donwsampler, as shown below.
Figure 11.15:
A downsampler consists of an anti-aliasing lter and a decimator.
11.7.2 Upsampling
Now, consider the case where we start with a sampling frequency
frequency by an integer factor to a new value
period
(as
fs = 1/T ),
fs' = k fs .
fs
and are asked to increase the sampling
When this change is translated to the sampling
it is easy to see that the new sampling period
T ' = T /k
will be a fraction (1/k ) of
the original sampling period. Therefore, the change in sampling frequency requires the acquisition of new
samples in addition to those already available under the old sampling frequency. For this reason, this process
is commonly known as up sampling.
While at rst sight this may imply a demand to go back to the continuous-time signal, we must recall
that samples that are obtained with a sampling frequency greater than the Nyquist frequency contain all
information needed to recover the continuous-time signal, and so it should be possible to infer the new
168
samples directly from existing ones (as long as no aliasing has occurred). For this purpose, we will retrieve
the concept of an expanded discrete-time signal:
xk [n]
Notice that this signal
xk [n]
={
x [n/k]
0
if
n/k Z,
should match the upsampled signal
and our goal is to ll in the missing samples in
xk [n]
(11.39)
otherwise.
x' [n]
for indices that are multiples of
k,
currently having value zero. To see how this can be
done, we appeal to the DTFT of the expanded signal: recall that the time expansion property of the DTFT
Xk () = X (k), which in practice compresses the DTFT in frequency by a factor of k and

2/k -periodic. In contrast, the DTFT of the upsampled signal would remain 2 -periodic, while
simultaneously compressing each copy of the signal's spectrum by a factor of k .
gives us that
makes it
Figure 11.16:
The upsampling operation in time, continuous frequency, and discrete frequency.
This comparison illuminates a method to retrieve the upsampled signal
x' [n]
from the expanded signal
xk [n]: to apply a low-pass lter on the expanded signal xk [n] so that one of the k copies that appear
each 2 -length region of the DTFT is preserved. Such a lter will need a cuto frequency of fc = /k .
combination of an expander and a low-pass lter is known as an upsampler, as shown below.
Figure 11.17:
An upsampler consists of an expander and a low-pass lter.
over
The
169
11.7.3 Rational Sampling Frequency Scaling

A third case where changes in the sampling frequency can be resolved in the discrete-time domain is when
a
b fs .
Intuitively, one can see that such a change can be obtained by combining an upsampling by a factor of a
the ratio between the new and old sampling frequencies provides a rational number.
with a downsampling by a factor of
b.
That is,
fs' =
However, the order of these two operations is crucial.
Essentially, if downsampling is applied before upsampling, there is a chance that the downsampling antialiasing lter will remove a portion of the signal's spectrum that would alias but would have been shrunk
into the allowable region during the upsampling, and so the potential for unnecessary distortion is introduced.
In contrast, if upsampling is performed before downsampling, the cascade of the two systems will yield a
sequence of two low-pass lters, and implementing only the narrower lter would provide the same output
as the original cascade. This is illustrated in an example below.
fs = 15kHz, the signal's bandwidth is f0 = 5kHz

= 2 (5kHz/15kHz) = 2/3, and we are interested in resampling to
Consider the case where the original sampling frequency

(so that the DTFT bandwidth is
fs = 10kHz
(i.e., the minimum allowed sampling frequency under the Nyquist criterion).
applying downsampling before upsampling results in the following:
In this case,
170
Changing the sampling frequency by downsampling followed by upsampling. The downsampling anti-aliasing lter cut o part of the original spectrum.
Figure 11.18:
and so only the portion of the signal's spectrum below
2.5kHz (/3 in the original spectrum) survives the
process. In contrast, applying upsampling before downsampling allows the entire spectrum to go through,
eectively meeting the bound given by the Nyquist criterion.
171
Figure 11.19: Changing the sampling frequency by upsampling followed by downsampling. The entire
spectrum is preserved through the process, and one of the lowpass lters is redundant.
11.8 Discrete Time Processing of Continuous Time Signals
11
11.8.1 Introduction
Digital computers can process discrete time signals using extremely exible and powerful algorithms. However, most signals of interest are continuous time signals, which is how data almost always appears in nature.
Now that the theory supporting methods for generating a discrete time signal from a continuous time signal
through sampling and then perfectly reconstructing the original signal from its samples without error has
been discussed, it will be shown how this can be applied to implement continuous time, linear time invariant systems using discrete time, linear time invariant systems. This is of key importance to many modern
technologies as it allows the power of digital computing to be leveraged for processing of analog signals.
11 This
172
11.8.2 Discrete Time Processing of Continuous Time Signals

11.8.2.1 Process Structure
With the aim of processing continuous time signals using a discrete time system, we will now examine one of
the most common structures of digital signal processing technologies. As an overview of the approach taken,
the original continuous time signal
the samples spectrum
H2
time invariant lter

the bandlimit of
Xs
Xs ,
x is sampled to a discrete time signal xs
X.
Then a discrete time, linear
is applied, which modies the shape of the samples spectrum
to produce another signal
to produce a continuous time output signal
H1 .
in such a way that the periods of
is as close as possible in shape to the spectrum of
y,
ys .
Xs
but cannot increase
This is reconstructed with a suitable reconstruction lter
thus eectively implementing some continuous time system
This process is illustrated in Figure 11.20, and the spectra are shown for a specic case in Figure 11.21.
Figure 11.20:
shown.
A block diagram for processing of continuous time signals using discrete time systems is
Further discussion about each of these steps is necessary, and we will begin by discussing the analog to
digital converter, often denoted by ADC or A/D. It is clear that in order to process a continuous time signal
using discrete time techniques, we must sample the signal as an initial step. This is essentially the purpose of
the ADC, although there are practical issues that which will be discussed later. An ADC takes a continuous
time analog signal as input and produces a discrete time digital signal as output, with the ideal innite
precision case corresponding to sampling. As stated by the Nyquist-Shannon Sampling theorem, in order to
retain all information about the original signal, we usually wish sample above the Nyquist frequency
where the original signal is bandlimited to
(B, B).
fs 2B
When it is not possible to guarantee this condition, an
anti-aliasing lter should be used.

The discrete time lter is where the intentional modications to the signal information occur. This is
commonly done in digital computer software after the signal has been sampled by a hardware ADC and
before it is used by a hardware DAC to construct the output.
This allows the above setup to be quite
exible in the lter that it implements. If sampling above the Nyquist frequency the. Any modications
that the discrete lter makes to this shape can be passed on to a continuous time signal assuming perfect
reconstruction. Consequently, the process described will implement a continuous time, linear time invariant
lter. This will be explained in more mathematical detail in the subsequent section. As usual, there are, of
course, practical limitations that will be discussed later.
Finally, we will discuss the digital to analog converter, often denoted by DAC or D/A. Since continuous
time lters have continuous time inputs and continuous time outputs, we must construct a continuous time
signal from our ltered discrete time signal. Assuming that we have sampled a bandlimited at a suciently
high rate, in the ideal case this would be done using perfect reconstruction through the Whittaker-Shannon
interpolation formula. However, there are, once again, practical issues that prevent this from happening that
will be discussed later.
173
Spectra are shown in black for each step in implementing a continuous time lter using
a discrete time lter for a specic signal. The lter frequency responses are shown in blue, and both are
meant to have maximum value 1 in spite of the vertical scale that is meant only for the signal spectra.
Ideal ADCs and DACs are assumed.
Figure 11.21:
11.8.2.2 Discrete Time Filter

With some initial discussion of the process illustrated in Figure 11.20 complete, the relationship between
the continuous time, linear time invariant lter
be explored.
H1
and the discrete time, linear time invariant lter
H2
can
We will assume the use of ideal, innite precision ADCs and DACs that perform sampling
and perfect reconstruction, respectively, using a sampling rate

bandlimited to
(B, B).
fs = 1/Ts 2B
where the input signal
is
Note that these arguments fail if this condition is not met and aliasing occurs. In
that case, preapplication of an anti-aliasing lter is necessary for these arguments to hold.
Recall that we have already calculated the spectrum
Xs
of the samples
xs
given an input
x with spectrum
as

2k
1 X
X
.
Ts
2Ts
Xs () =
(11.40)
k=
Likewise, the spectrum
Ys
of the samples
ys
given an output
Ys () =
with spectrum
is

1 X
2k
Y
.
Ts
2Ts
(11.41)
k=
From the knowledge that
ys = (H1 x)s = H2 (xs ),

H1
k=
Because
is bandlimited to
H2 () =
2k
2Ts
H2
is

X
2k
2Ts
(1/2Ts , 1/2Ts ),
k=
More simply stated,

H1
it follows that
2k
2Ts

= H2 ()
X
k=

X
2k
2Ts

.
we may conclude that

(u ( (2k 1) ) u ( (2k + 1) )) .
(11.43)
H2 () = H1 (/2Ts ) for [, ).
H1 , the above equation solves the system
implement H2 . The lter H2 must be chosen such that it has a
periodic and
Given a specic continuous time, linear time invariant lter

design problem provided we know how to
(11.42)
174
frequency response where each period has the same shape as the frequency response of
H1 on (1/2Ts , 1/2Ts ).
This is illustrated in the frequency responses shown in Figure 11.21.

We might also want to consider the system analysis problem in which a specic discrete time, linear
time invariant lter
H2
is given, and we wish to describe the lter
(fs /2, fs /2)

H1 (f ) = H2 (2f /fs )
H1 .
There are many such lters, but we
can describe their frequency responses on
using the above equation. Isolating one period of
H2 ()
for
yields the conclusion that
be bandlimited to
(fs /2, fs /2),
f (fs /2, fs /2).
Because
was assumed to
the value of the frequency response elsewhere is irrelevant.
11.8.3 Practical Considerations

As mentioned before, there are several practical considerations that need to be addressed at each stage of
the process shown in Figure 11.20. Some of these will be briey addressed here, and a more complete model
of how discrete time processing of continuous time signals appears in Figure 11.22.
A more complete model of how discrete time processing of continuous time signals is
implemented in practice. Notice the addition of anti-aliasing and anti-imaging lters to promote input
and output bandlimitedness. The ADC is shown to perform sampling with quantization. The digital
lter is further specied to be causal. The DAC is shown to perform imperfect reconstruction, a zero
order hold in this case.
Figure 11.22:
11.8.3.1 Anti-Aliasing Filter

In reality, we cannot typically guarantee that the input signal will have a specic bandlimit, and suciently
high sampling rates cannot necessarily be produced. Since it is imperative that the higher frequency components not be allowed to masquerade as lower frequency components through aliasing, anti-aliasing lters
with cuto frequency less than or equal to
fs /2
must be used before the signal is fed into the ADC. The
block diagram in Figure 11.22 reects this addition.

As described in the previous section, an ideal lowpass lter removing all energy at frequencies above
fs /2
would be optimal. Of course, this is not achievable, so approximations of the ideal lowpass lter with low
gain above
fs /2
must be accepted. This means that some aliasing is inevitable, but it can be reduced to a
mostly insignicant level.
11.8.3.2 Signal Quantization

In our preceding discussion of discrete time processing of continuous time signals, we had assumed an ideal
case in which the ADC performs sampling exactly. However, while an ADC does convert a continuous time
signal to a discrete time signal, it also must convert analog values to digital values for use in a digital logic
device, a phenomenon called quantization. The ADC subsystem of the block diagram in Figure 11.22 reects
this addition.
175
The data obtained by the ADC must be stored in nitely many bits inside a digital logic device. Thus,
there are only nitely many values that a digital sample can take, specically
2N
where
is the number of
bits, while there are uncountably many values an analog sample can take. Hence something must be lost in
the quantization process. The result is that quantization limits both the range and precision of the output
of the ADC. Both are nite, and improving one at constant number of bits requires sacricing quality in the
other.
11.8.3.3 Filter Implementability

In real world circumstances, if the input signal is a function of time, the future values of the signal cannot
be used to calculate the output. Thus, the digital lter
H2
and the overall system
H1
must be causal. The
lter annotation in Figure 11.22 reects this addition. If the desired system is not causal but has impulse
response equal to zero before some time
t0 ,
a delay can be introduced to make it causal. However, if this
delay is excessive or the impulse response has innite length, a windowing scheme becomes necessary in order
to practically solve the problem. Multiplying by a window to decrease the length of the impulse response
can reduce the necessary delay and decrease computational requirements.
Take, for instance the case of the ideal lowpass lter. It is acausal and innite in length in both directions.
Thus, we must satisfy ourselves with an approximation. One might suggest that these approximations could
be achieved by truncating the sinc impulse response of the lowpass lter at one of its zeros, eectively
windowing it with a rectangular pulse.
However, doing so would produce poor results in the frequency
domain as the resulting convolution would signicantly spread the signal energy. Other windowing functions,
of which there are many, spread the signal less in the frequency domain and are thus much more useful for
producing these approximations.
11.8.3.4 Anti-Imaging Filter

In our preceding discussion of discrete time processing of continuous time signals, we had assumed an ideal
case in which the DAC performs perfect reconstruction. However, when considering practical matters, it is
important to remember that the sinc function, which is used for Whittaker-Shannon interpolation, is innite
in length and acausal. Hence, it would be impossible for an DAC to implement perfect reconstruction.
Instead, the DAC implements a causal zero order hold or other simple reconstruction scheme with respect
to the sampling rate
to
(fs /2, fs /2).
fs
used by the ADC. However, doing so will result in a function that is not bandlimited
Therefore, an additional lowpass lter, called an anti-imaging lter, must be applied to the
output. The process illustrated in Figure 11.22 reects these additions. The anti-imaging lter attempts to
bandlimit the signal to
(fs /2, fs /2),
so an ideal lowpass lter would be optimal. However, as has already
been stated, this is not possible. Therefore, approximations of the ideal lowpass lter with low gain above
fs /2
must be accepted.
The anti-imaging lter typically has the same characteristics as the anti-aliasing
lter.
11.8.4 Discrete Time Processing of Continuous Time Signals Summary

As has been show, the sampling and reconstruction can be used to implement continuous time systems
using discrete time systems, which is very powerful due to the versatility, exibility, and speed of digital
computers. However, there are a large number of practical considerations that must be taken into account
when attempting to accomplish this, including quantization noise and anti-aliasing in the analog to digital
converter, lter implementability in the discrete time lter, and reconstruction windowing and associated
issues in the digital to analog converter. Many modern technologies address these issues and make use of
this process.
176
Appendix: Mathematical Pot-Pourri

12.1 Basic Linear Algebra
This brief tutorial on some key terms in linear algebra is not meant to replace or be very helpful to those of
you trying to gain a deep insight into linear algebra. Rather, this brief introduction to some of the terms and
ideas of linear algebra is meant to provide a little background to those trying to get a better understanding
or learn about eigenvectors and eigenfunctions, which play a big role in deriving a few important ideas on
Signals and Systems. The goal of these concepts will be to provide a background for signal decomposition
2
and to lead up to the derivation of the Fourier Series .
12.1.1 Linear Independence

A set of vectors
{x1 , x2 , . . . , xk } , xi Cn
are
linearly independent if none of them can be written as
a linear combination of the others.
Denition 12.1: Linearly Independent

For a given set of vectors,
{x1 , x2 , . . . , xn },
they are linearly independent if
c1 x1 + c2 x2 + + cn xn = 0
only when
Example
c1 = c2 = = cn = 0
We are given the following two vectors:
x1 =
x2 =
These are
2
6
4
not linearly independent as proven by the following statement, which, by inspection,
can be seen to not adhere to the denition of linear independence stated above.
(x2 = 2x1 ) (2x1 + x2 = 0)

Another approach to reveal a vectors independence is by graphing the vectors. Looking at these two
vectors geometrically (as in Figure 12.1), one can again prove that these vectors are
independent.

2 "Fourier Series: Eigenfunction Approach" <http://legacy.cnx.org/content/m10496/latest/>

177
not linearly
APPENDIX
178
Graphical representation of two vectors that are not linearly independent.
Figure 12.1:
Example 12.1
x1 =
x2 =
These are
3
2
1
2
linearly independent since

c1 x1 = (c2 x2 )
only if
c1 = c2 = 0.
independent.
Based on the denition, this proof shows that these vectors are indeed linearly
Again, we could also graph these two vectors (see Figure 12.2) to check for linear
independence.
Figure 12.2:
Graphical representation of two vectors that are linearly independent.
Exercise 12.1.1
Are
{x1 , x2 , x3 }
linearly independent?
x1 =
3
2
APPENDIX
179
x2 =
x3 =
1
2
1
0
As we have seen in the two above examples, often times the independence of vectors can be easily seen
through a graph. However this may not be as easy when we are given three or more vectors. Can you easily
tell whether or not these vectors are independent from Figure 12.3. Probably not, which is why the method
used in the above solution becomes important.
Plot of the three vectors. Can be shown that a linear combination exists among the
three, and therefore they are not linear independent.
Figure 12.3:
Hint:
A set of
vectors in
Cn
cannot be linearly independent if
m > n.
12.1.2 Span
Denition 12.2: Span
The span
{x1 , x2 , . . . , xk }
{x1 , x2 , . . . , xk }
of a set of vectors
combination of
is the set of vectors that can be written as a linear
span ({x1 , . . . , xk }) = {1 x1 + 2 x2 + + k xk , i Cn }
Example
Given the vector
x1 =
the span of
x1
is a
Example
Given the vectors
line.
x1 =
3 "Subspaces",
3
2
Denition 2: "Span" <http://legacy.cnx.org/content/m10297/latest/#defn2>

APPENDIX
180
x2 =
the span of these vectors is
C2 .
12.1.3 Basis
Denition 12.3: Basis
A basis for
Cn
is a set of vectors that: (1) spans
Clearly, any set of
Cn
linearly independent vectors is a
Example 12.2
and (2) is linearly independent.

basis for Cn .
We are given the following vector
.
.
.
ei =
1
.
..
0
where the
1 is always in the ith place and the remaining values are zero.
Then the
basis for Cn is
{ei , i = [1, 2, . . . , n] }
note:
{ei , i = [1, 2, . . . , n] }
is called the
standard basis.
Example 12.3
h1 =
h2 =
{h1 , h2 }
is a basis for
1
1
1
1
C2 .
APPENDIX
181
Plot of basis for C2
Figure 12.4:
If
{b1 , . . . , b2 }
is a basis for
Cn ,
then we can express
any x Cn as a linear combination of the bi 's:
x = 1 b1 + 2 b2 + + n bn , i C
Example 12.4
Given the following vector,
x=
writing
in terms of
{e1 , e2 }
1
2
gives us
x = e1 + 2e2
Exercise 12.1.2
Try and write
{h1 , h2 }
in terms of
In the two basis examples above,
(dened in the previous example).
is the same vector in both cases, but we can express it in many dierent
ways (we give only two out of many, many possibilities). You can take this even further by extending this
idea of a basis to
note:
function spaces.
As mentioned in the introduction, these concepts of linear algebra will help prepare you
4
to understand the Fourier Series , which tells us that we can express periodic functions,
terms of their basis functions,
j0 nt
[Media Object]5
4 "Fourier Series: Eigenfunction Approach" <http://legacy.cnx.org/content/m10496/latest/>
5 This media object is a LabVIEW VI. Please view or download it at
<LinearAlgebraCalc3.llb>
f (t),
in
APPENDIX
182
Khan Lecture on Basis of a Subspace

<http://www.youtube.com/v/zntNi3ybfQ&rel=0&color1=0xb1b1b1&color2=0xd0d0d0&hl=en_US&feature=player_embedded&fs=1>
Figure 12.5:
video from Khan Academy, Basis of a Subspace - 20 min.
12.2 Linear Constant Coecient Dierence Equations
12.2.1 Introduction: Dierence Equations

In our study of signals and systems, it will often be useful to describe systems using equations involving
the rate of change in some quantity. In discrete time, this is modeled through dierence equations, which
are a specic type of recurrance relation. For instance, recall that the funds in an account with discretely
componded interest rate
will increase by
times the previous balance.
Thus, a discretely compounded
interest system is described by the rst order dierence equation shown in (12.1).
y (n) = (1 + r) y (n 1)
(12.1)
Given a suciently descriptive set of initial conditions or boundary conditions, if there is a solution to the
dierence equation, that solution is unique and describes the behavior of the system. Of course, the results
are only accurate to the degree that the model mirrors reality.
12.2.2 Linear Constant Coecient Dierence Equations

An important subclass of dierence equations is the set of linear constant coecient dierence equations.
These equations are of the form
Cy (n) = f (n)
where
(12.2)
is a dierence operator of the form given
C = cN DN + cN 1 DN 1 + ... + c1 D + c0
in which
(12.3)
is the rst dierence operator
D (y (n)) = y (n) y (n 1) .
Note that operators of this type satisfy the linearity conditions, and
(12.4)
c0 , ..., cn
are real constants.
However, (12.2) can easily be written as a linear constant coecient recurrence equation without dierence
operators.
Conversely, linear constant coecient recurrence equations can also be written in the form of
a dierence equation, so the two types of equations are dierent representations of the same relationship.
Although we will still call them linear constant coecient dierence equations in this course, we typically
will not write them using dierence operators. Instead, we will write them in the simpler recurrence relation
form
N
X
k=0
6 This
ak y (n k) =
M
X
bk x (n k)
k=0

(12.5)
APPENDIX
where
183
is the input to the system and
1
y (n) =
a0
is the output. This can be rearranged to nd
N
X
ak y (n k) +
k=1
M
X
y (n)
as
!
bk x (n k)
(12.6)
k=0
The forms provided by (12.5) and (12.6) will be used in the remainder of this course.
A similar concept for continuous time setting, dierential equations, is discussed in the chapter on time
domain analysis of continuous time systems.
There are many parallels between the discussion of linear
constant coecient ordinary dierential equations and linear constant coecient dierece equations.
12.2.3 Applications of Dierence Equations

Dierence equations can be used to describe many useful digital lters as described in the chapter discussing
the z-transform. An additional illustrative example is provided here.
Example 12.5
Recall that the Fibonacci sequence describes a (very unrealistic) model of what happens when a
pair rabbits get left alone in a black box...
The assumptions are that a pair of rabits never die
and produce a pair of ospring every month starting on their second month of life. This system is
dened by the recursion relation for the number of rabit pairs
y (n)
at month
y (n) = y (n 1) + y (n 2)
with the initial conditions
y (0) = 0 and y (1) = 1.
(12.7)
The result is a very fast growth in the sequence.
This is why we do not open black boxes.
12.2.4 Linear Constant Coecient Dierence Equations Summary

Dierence equations are an important mathematical tool for modeling discrete time systems. An important
subclass of these is the class of linear constant coecient dierence equations. Linear constant coecient
dierence equations are often particularly easy to solve as will be described in the module on solutions to
linear constant coecient dierence equations and are useful in describing a wide range of situations that
arise in electrical engineering and in other elds.
12.3 Solving Linear Constant Coecient Dierence Equations
12.3.1 Introduction
The approach to solving linear constant coecient dierence equations is to nd the general form of all
possible solutions to the equation and then apply a number of conditions to nd the appropriate solution.
The two main types of problems are initial value problems, which involve constraints on the solution at several
consecutive points, and boundary value problems, which involve constraints on the solution at nonconsecutive
points.
The number of initial conditions needed for an
N th
order dierence equation, which is the order of the
highest order dierence or the largest delay parameter of the output in the equation, is
N,
and a unique
solution is always guaranteed if these are supplied. Boundary value probelms can be slightly more complicated
and will not necessarily have a unique solution or even a solution at all for a given set of conditions. Thus,
this section will focus exclusively on initial value problems.
7 This
APPENDIX
184
12.3.2 Solving Linear Constant Coecient Dierence Equations

Consider some linear constant coecient dierence equation given by
Ay (n) = f (n),
in which
is a
dierence operator of the form
A = aN DN + aN 1 DN 1 + ... + a1 D + a0
where
(12.8)
is the rst dierence operator
D (y (n)) = y (n) y (n 1) .
yh (n)
Let
A,
yp (n) be two functions such that Ayh (n) = 0

L (yh (n) + yp (n)) = 0 + f (n) = f (n). Thus, the
and
note that
and
(12.9)
Ayp (n) = f (n).
By the linearity of
yg (n) to any
yh (n) to the
f (n).
form of the general solution
linear constant coecient ordinary dierential equation is the sum of a homogeneous solution
equation
Ay (n) = 0
and a particular solution
yp (n)
that is specic to the forcing function
We wish to determine the forms of the homogeneous and nonhomogeneous solutions in full generality in
order to avoid incorrectly restricting the form of the solution before applying any conditions. Otherwise, a
valid set of initial or boundary conditions might appear to have no corresponding solution trajectory. The
following sections discuss how to accomplish this for linear constant coecient dierence equations.
12.3.2.1 Finding the Homogeneous Solution

In order to nd the homogeneous solution to a dierence equation described by the recurrence relation
PN
PN
ak y (n k) = f (n), consider the dierence equation k=0 ak y (n k) = 0. We
PN
n
nk
have the form c for some complex constants c, . Since
= 0 for
k=0 ak c
k=0
tions
know that the solua solution it follows
that
cnN
N
X
ak N k = 0
(12.10)
k=0
so it also follows that
a0 N + ... + aN = 0.
(12.11)
Therefore, the solution exponential are the roots of the above polynomial, called the characteristic polynomial.
For equations of order two or more, there will be several roots. If all of the roots are distinct, then the
general form of the homogeneous solution is simply
yh (n) = c1 n1 + ... + c2 n2 .
(12.12)
If a root has multiplicity that is greater than one, the repeated solutions must be multiplied by each power
of
from 0 to one less than the root multipicity (in order to ensure linearly independent solutions). For
instance, if
had a multiplicity of 2 and
had multiplicity 3, the homogeneous solution would be
yh (n) = c1 n1 + c2 nn1 + c3 n2 + c4 nn2 + c5 n2 n2 .
(12.13)
Example 12.6
Recall that the Fibonacci sequence describes a (very unrealistic) model of what happens when a
pair rabbits get left alone in a black box...
The assumptions are that a pair of rabits never die
and produce a pair of ospring every month starting on their second month of life. This system is
dened by the recursion relation for the number of rabit pairs
y (n)
at month
y (n) y (n 1) y (n 2) = 0
(12.14)
APPENDIX
185
with the initial conditions
y (0) = 0
y (1) = 1.
and
Note that the forcing function is zero, so only the homogenous solution is needed. It is easy to
2 1 = 0, so there are two roots with multiplicity
1 5
2 . Thus, the solution is of the form
see that the characteristic polynomial is

one. These are
1 =
1+ 5
and
2
2 =
y (n) = c1
!n
!n
1 5
1+ 5
+ c2
.
2
2
(12.15)
Using the initial conditions, we determine that
c1 =
5
5
(12.16)
and
c2 =
5
.
5
(12.17)
Hence, the Fibonacci sequence is given by
5
y (n) =
5
!n
5
1+ 5
2
5
!n
1 5
.
2
(12.18)
12.3.2.2 Finding the Particular Solution

Finding the particular solution is a slightly more complicated task than nding the homogeneous solution. It
can be found through convolution of the input with the unit impulse response once the unit impulse response
is known.
Finding the particular solution ot a dierential equation is discussed further in the chapter
concerning the z-transform, which greatly simplies the procedure for solving linear constant coecient
dierential equations using frequency domain tools.
Example 12.7
Consider the following dierence equation describing a system with feedback
y (n) ay (n 1) = x (n) .
(12.19)
In order to nd the homogeneous solution, consider the dierence equation
y (n) ay (n 1) = 0.
It is easy to see that the characteristic polynomial is
(12.20)
a = 0,
so
=a
is the only root. Thus
the homogeneous solution is of the form
yh (n) = c1 an .
(12.21)
In order to nd the particular solution, consider the output for the
x (n) = (n)
unit impulse case
y (n) ay (n 1) = (n) .
By inspection, it is clear that the impulse response is
given
x (n)
a u (n).
(12.22)
Hence, the particular solution for a
is
yp (n) = x (n) (an u (n)) .
(12.23)
APPENDIX
186
Therefore, the general solution is
yg (n) = yh (n) + yp (n) = c1 an + x (n) (an u (n)) .
(12.24)
Initial conditions and a specic input can further tailor this solution to a specic situation.
12.3.3 Solving Dierence Equations Summary

Linear constant coecient dierence equations are useful for modeling a wide variety of discrete time systems.
The approach to solving them is to nd the general form of all possible solutions to the equation and then
apply a number of conditions to nd the appropriate solution.
This is done by nding the homogeneous
solution to the dierence equation that does not depend on the forcing function input and a particular
solution to the dierence equation that does depend on the forcing function input.
APPENDIX
187

By playing around with the vectors and doing a little trial and error, we will discover the following relationship:
x1 x2 + 2x3 = 0
Thus we have found a linear combination of these three vectors that equals zero without setting the coecients
equal to zero. Therefore, these vectors are
not linearly independent!

x=
1
3
h1 +
h2
2
2
APPENDIX
188
Appendix: Viewing Interactive Content

189
APPENDIX
190
13.1 Viewing Embedded LabVIEW Content in Connexions
13.1.1 Introduction
In order to view LabVIEW content embedded in Connexions modules, you must install and enable the LabVIEW 8.0 and 8.5 Local VI Execution Browser Plug-in for Windows. Step-by-step installation instructions
are given below. Once installation is complete, the placeholder box at the bottom of this module should
display a demo LabVIEW virtual instrument (VI).
13.1.2 Installing the LabVIEW Run-time Engine on Microsoft Windows

1. Download
and
install
the
LabVIEW
2
http://zone.ni.com/devzone/cda/tut/p/id/4346
2. Download
and
install
the
LabVIEW
3
http://zone.ni.com/devzone/cda/tut/p/id/6633
3. Dowload the
8.0
Runtime
Engine
found
at:
8.5
Runtime
Engine
found
at:
.
.
LVBrowserPlugin.ini le from http://zone.ni.com/devzone/cda/tut/p/id/82884 , and

My Documents\LabVIEW Data folder. (You may have to create this folder if it doesn't
place it in the
already exist.)
4. Restart your computer to complete the installation.
5. The placeholder box at the bottom of this module should now display a demo LabVIEW virtual
instrument (VI).
13.1.3 Example Virtual Instrument

<DFD_Utility.llb>
Figure
13.1:
Digital
lter
http://cnx.org/content/m13115/latest/5 .
design
LabVIEW
virtual
instrument
from
13.2 Getting Started With Mathematica
13.2.1 What is Mathematica?

Mathematica is a computational software program used in technical elds.
It is developed by Wolfram
Research. Mathematica makes it easy to visualize data and create GUIs in only a few lines of code.
13.2.2 How can I run, create, and nd Mathematica les?

Run
The free CDF Player
is available for running non-commercial Mathematica programs. The option exists

2 http://zone.ni.com/devzone/cda/tut/p/id/4346
5 http://cnx.org/content/m13115/latest/
7 http://www.wolfram.com/cdf-player/
APPENDIX
191
of downloading source les and running on your computer, but the CDF-player comes with a plug-in for
viewing dynamic content online on your web browser!
Create
Mathematica 8 is available for purchase
from Wolfram. Many universities (including Rice) and companies

9
already have a Mathematica license. Wolfram has a free, save-disabled 15-day trial version
Find
of Mathematica.
Wolfram has thousands of Mathematica programs (including source code) available at the Wolfram Demon10
strations Project
. Anyone can create and submit a Demonstration. Also, many other websites (including
Connexions) have a lot of Mathematica content.
13.2.3 What do I need to run interactive content?

Mathematica 8 is supported on Linux, Microsoft Windows, Mac OS X, and Solaris.
Mathematica's free
CDF-player is available for Windows and Mac OS X, and is in development for Linux; the CDF-Player
plugin is available for IE, Firefox, Chrome, Safari, and Opera.
13.2.4 How can I upload a Mathematica le to a Connexions module?

Go to the Files tab at the top of the module and upload your .cdf le, along with an (optional) screenshot
of the le in use. In order to generate a clean bracket-less screenshot, you should do the following:
Open your .cdf in Mathematica and left click on the bracket surrounding the manipulate command.
Click on Cell->Convert To->Bitmap.
Then click on File->Save Selection As, and save the image le in your desired image format.
Embed the les into the module in any way you like. Some tags you may nd helpful include image,
gure, download, and link (if linking to an .cdf le on another website). The best method is to create an
interactive gure, and include a fallback png image of the cdf le should the CDF image not render properly.
See the interactive demo/image below.
Convolution Demo
<figure id="demoonline">
<media id="CNXdemoonline" alt="timeshiftDemo">
<image mime-type="image/png" src="Convolutiondisplay-4.cdf" thumbnail="Convolution4.0Display.png" width
<object width="500" height="500" src="Convolutiondisplay-4.cdf" mime-type="application/vnd.wolfram.cdf"
<image mime-type="application/postscript" for="pdf" src="Convolution4.0Display.png" width="400"/>
</media>
<caption>Interact (when online) with a Mathematica CDF demonstrating Convolution. To Download, right-cl
</figure>
8 http://www.wolfram.com/products/mathematica/purchase.html
9 http://www.wolfram.com/products/mathematica/experience/request.cgi
10 http://demonstrations.wolfram.com/index.html
APPENDIX
192
Interact (when online) with a Mathematica CDF demonstrating Convolution. To Download, right-click and save target as .cdf.
Figure 13.2:
Alternatively, this is how it looks when you use a thumbnail link to a live online demo.
Click on the above thumbnail image (when online) to view an interactive Mathematica
Player demonstrating Convolution.
Figure 13.3:
13.2.5 How can I learn Mathematica?

Open Mathematica and go to the Getting Started section of the "Welcome to Mathematica" screen, or check
out Help: Documentation Center.
11
The Mathematica Learning Center
has lots of screencasts, how-tos, and tutorials.
When troubleshooting, the error messages are often unhelpful, so it's best to evaluate often so the problem
can be easily located. Search engines like Google are useful when you're looking for an explanation of specic
error messages.
11 http://www.wolfram.com/learningcenter/
GLOSSARY
193
Glossary
B Basis
Cn
A basis for
is a set of vectors that: (1) spans
Cn
and (2) is linearly independent.
L Linearly Independent
For a given set of vectors,
{x1 , x2 , . . . , xn },
they are linearly independent if
c1 x1 + c2 x2 + + cn xn = 0
only when
c1 = c2 = = cn = 0
Example:
x1 =
x2 =
These are
3
2
6
4
not linearly independent as proven by the following statement, which, by
inspection, can be seen to not adhere to the denition of linear independence stated above.
(x2 = 2x1 ) (2x1 + x2 = 0)

Another approach to reveal a vectors independence is by graphing the vectors. Looking at these
two vectors geometrically (as in
12
), one can again prove that these vectors are
not linearly
independent.
S Span
13
The span
{x1 , x2 , . . . , xk }
{x1 , x2 , . . . , xk }
of a set of vectors
combination of
is the set of vectors that can be written as a linear
span ({x1 , . . . , xk }) = {1 x1 + 2 x2 + + k xk , i Cn }
Example:
Given the vector
x1 =
the span of
x1
is a
3
2
line.
12 http://legacy.cnx.org/content/m10734/latest/
13 http://legacy.cnx.org/content/m10734/latest/
GLOSSARY
194
Example:
Given the vectors
x1 =
x2 =
the span of these vectors is
3
2
1
2
C2 .
INDEX
195
Index of Keywords and Terms

Keywords are listed by the section with that keyword (page numbers are in parentheses).
Keywords
do not necessarily appear in the text of the page. They are merely associated with that section.
apples, 1.1 (1)
Terms are referenced by the page they appear on. Ex.
acausal, 11.8(171)
continuous frequency, 6.2(76), 9.3(125)
ADC, 11.8(171)
continuous time, 2.1(17), 2.2(22), 4.1(51),
algebra, 1.2(5)
4.2(52), 54, 4.3(55), 5.1(65), 5.3(68),
alias, 11.5(160), 11.6(163)
5.4(69), 6.1(75), 6.2(76), 6.3(79),
aliasing, 11.5(160), 11.6(163), 11.8(171)
6.4(83), 11.1(147)
analog, 2.1(17), 18, 11.8(171)
Continuous Time Convolution Integral, 54
analysis, 5.4(69)
Continuous Time Fourier Series, 69
Anti-Aliasing, 11.6(163), 11.8(171)
Continuous Time Fourier Transform, 75, 76
anti-imaging, 11.8(171)
continuous-time, 4.1(51)
anticausal, 2.1(17), 11.8(171)
Continuous-Time Fourier Transform, 77
aperiodic, 2.1(17), 66, 6.1(75), 75,
Continuous-Time Systems, 4.5(62)
9.1(121), 123
converter, 11.8(171)
Convolution, 54, 4.3(55), 4.4(58), 6.3(79),
bandlimited, 11.8(171)
6.5(85), 109, 8.3(110), 8.4(115)
bases, 12.1(177)
convolution integral, 85
basis, 12.1(177), 180, 180, 180
Cooley-Tukey, 10.2(138)
BIBO stability, 3.2(39), 4.5(62), 8.5(118)
CTFS, 5.4(69)
BIBO stable, 3.2(39)

bits, 11.8(171)
block diagram, 3.1(37)
Ex.
apples, 1
CTFT, 6.2(76), 6.5(85)
DAC, 11.8(171)
bounded input bounded output stable,
de, 4.1(51)
3.2(39)
decimation, 7.3(95)
buttery, 141
decompose, 7.1(91)
cardinal, 11.3(152)
cascade, 3.1(37), 3.3(43)
causal, 2.1(17), 3.2(39), 11.8(171)
causality, 3.2(39), 4.5(62), 8.5(118)
common, 9.5(132)
complex, 2.2(22), 7.1(91)
complex exponential, 2.6(33), 5.3(68), 92,
7.5(100), 9.2(124)
complex exponentials, 126
complex numbers, 1.1(1), 1.2(5), 1.3(11)
complex plane, 2.6(33), 7.5(100)
complex-valued, 2.2(22), 7.1(91)
complexity, 139, 142
composite, 144
computational advantage, 140
considerations, 11.8(171)
Constant Coecient, 12.2(182), 12.3(183)
continuous, 17, 6.5(85), 11.8(171)
delta function, 8.2(107)

DFT, 10.1(137), 10.3(139)
dierence equation, 8.1(105)
Dierence Equations, 12.2(182), 12.3(183)
dierential, 4.1(51)
dierential equation, 54
dierential equations, 4.1(51)
digital, 2.1(17), 18, 11.8(171)
digital signal processing, 8.1(105)
dirac delta function, 2.2(22), 2.5(31), 31
Discete Time Fourier Transform, 123
discrete, 17, 11.8(171)
discrete convolution, 9.6(133)
Discrete Fourier Transform, 10.1(137)
discrete time, 2.1(17), 5.1(65), 8.2(107),
109, 8.3(110), 8.4(115), 9.1(121),
9.2(124), 9.3(125), 9.5(132), 9.6(133),
11.1(147)
Discrete Time Convolution Sum, 109
INDEX
196
Discrete Time Fourier Transform, 125

discrete-time, 7.1(91), 9.4(128)
Discrete-Time Fourier Transform, 126
homogeneous, 12.3(183)
imperfect, 11.8(171)
9.4(128)
implementability, 11.8(171)
discrete-time periodic signal, 121
impulse, 2.2(22), 2.5(31), 4.2(52),
discrete-time signals, 7.3(95), 11.7(165)
7.4(98), 8.2(107), 109
discrete-time systems, 8.1(105), 8.5(118)
impulse response, 4.2(52), 8.2(107),
downsampling, 11.7(165)
8.3(110), 12.3(183)
DSP, 8.1(105), 11.2(149)
impulse-like input signal, 54
DT, 8.3(110)
independence, 12.1(177)
DTFT, 9.3(125), 9.5(132), 9.6(133)
innite-length signal, 19
duality, 6.3(79)
initial value, 12.3(183)
dynamic content, 13.1(190)
input, 3.1(37)
eigenfunction, 5.3(68), 5.4(69), 9.2(124)
interpolation, 7.3(95), 11.3(152)
eigenvalue, 5.3(68), 9.2(124)
invariant, 11.8(171)
eigenvector, 5.3(68), 9.2(124)

ELEC 301, 12.2(182)
linear, 3.2(39), 3.3(43), 11.8(171),
1.3(11)
12.2(182), 12.3(183)
embedded, 13.1(190)
linear algebra, 12.1(177)
Energy, 2.4(28), 7.2(93)
linear independence, 12.1(177)
engineering, 1.1(1), 1.2(5), 1.3(11)
linear system, 4.1(51)
even signal, 2.1(17), 20
linear time invariant, 4.3(55), 5.3(68),
examples, 9.5(132)
9.2(124)
expansion, 7.3(95)
Linear Time-Invariant Systems, 8.5(118)
exponential, 2.2(22), 7.1(91)
linearity, 6.3(79)
fast Fourier transform, 10.2(138), 10.3(139)
linearly independent, 177, 177
feedback, 3.1(37)
lowpass, 11.3(152), 11.4(157), 11.6(163)
FFT, 10.2(138), 10.3(139)
LTI, 4.3(55), 5.3(68), 9.2(124)
lter, 11.3(152), 11.6(163), 11.8(171)
LTI Systems, 4.5(62), 6.6(87)
nite-length signal, 19
form, 140
fourier, 10.3(139)
Fourier methods, 54, 109
fourier series, 5.1(65), 5.4(69)
M Mathematica, 13.2(190)
modulation, 6.3(79)
nonlinear, 3.2(39)
6.3(79), 6.4(83), 6.5(85), 6.6(87),
not, 143
9.3(125), 9.4(128), 9.6(133), 10.1(137),
Nyquist, 11.1(147), 11.2(149), 11.6(163),
10.3(139)
11.8(171)
fourier transforms, 9.5(132)
Nyquist frequency, 11.2(149)
frequency, 11.5(160), 11.6(163), 11.8(171)

Frequency Domain, 9.4(128)
FT, 6.5(85)
function spaces, 181
functional, 37
fundamental period, 18, 66, 75, 121
noncausal, 2.1(17), 3.2(39)

nonhomogeneous, 12.3(183)
fourier transform, 5.1(65), 6.2(76),
G
H
LabVIEW, 13.1(190)
Laplace transform, 54, 5.1(65)
electrical engineering, 1.1(1), 1.2(5),
ideal, 11.3(152), 11.4(157)

imaging, 11.8(171)
Discrete-Time Fourier Transform properties,
Nyquist theorem, 11.2(149)
odd signal, 2.1(17), 20

output, 3.1(37)
parallel, 3.1(37), 3.3(43)

perfect, 11.4(157)
geometry, 1.1(1)
hold, 11.8(171)
period, 18, 5.2(66)

periodic, 2.1(17), 5.2(66), 66, 67, 75, 121
INDEX
Q
R
197
periodic function, 5.2(66), 66, 75, 121
sinusoid, 7.1(91)
periodic functions, 66, 122
solution, 12.3(183)
periodicity, 5.2(66)
span, 12.1(177), 179
Player, 13.2(190)
spectrum, 11.1(147)
plug-in, 13.1(190)
spline, 11.3(152)
Power, 2.4(28), 7.2(93)
stability, 3.2(39)
practical, 11.8(171)
stable, 3.2(39)
precision, 11.8(171)
standard basis, 180
processing, 11.8(171)
superposition, 3.3(43), 8.1(105)
properties, 8.4(115)
symbolic-valued signals, 7.1(91)
property, 4.4(58)
synthesis, 5.4(69)
proportional, 139, 142
system theory, 3.1(37)
pulse, 2.2(22)
systems, 4.3(55), 4.4(58), 5.1(65),
quantization, 11.8(171)
range, 11.8(171)
6.4(83), 7.1(91)
theorem, 11.2(149)
real-valued, 7.1(91)
time, 11.8(171)
real-valued signal conjugate symmetry,
time dierentiation, 6.3(79)
6.3(79)
time domain, 8.1(105), 8.4(115)
reconstruct, 11.4(157)
time invariant, 3.2(39)
reconstruction, 11.1(147), 11.2(149),
time reversal, 2.3(25), 7.3(95)
11.3(152), 11.4(157), 11.5(160),
time scaling, 2.3(25), 6.3(79), 7.3(95)
11.8(171)
t-periodic, 5.2(66)
time shifting, 2.3(25), 6.3(79), 7.3(95)

time varying, 3.2(39)
sample, 11.2(149), 11.5(160)
time-invariant, 3.3(43)
sampling, 11.1(147), 11.2(149), 11.4(157),
transfer function, 65, 5.4(69)
11.5(160), 11.6(163), 11.7(165),
triangle, 2.2(22)
11.8(171)
sampling rate, 11.7(165)
sampling theorem, 11.8(171)
Sequence-Domain, 9.4(128)
triangle function, 24
unit sample function, 98
sequences, 7.1(91)
unit step, 2.2(22)
Shannon, 11.1(147), 11.2(149), 11.3(152),
unit-pulse function, 24
11.4(157), 11.8(171)
unit-step function, 23, 92
shift-invariant systems, 8.1(105)

sifting property, 2.2(22)
signal, 3.1(37), 5.2(66)
unit sample, 7.1(91), 92, 7.4(98)
upsampling, 11.7(165)
vector space, 1.3(11)
signals, 2.2(22), 2.3(25), 2.5(31),
VI, 13.1(190)
2.6(33), 3.2(39), 4.3(55), 4.4(58),
virtual instrument, 13.1(190)
5.1(65), 6.1(75), 6.4(83), 7.1(91),

7.4(98), 7.5(100), 8.3(110), 9.1(121),
W Whittaker, 11.3(152), 11.4(157),

11.8(171)
11.8(171)
window, 11.8(171)
signals and systems, 2.1(17), 8.3(110)

sinc, 11.3(152), 11.4(157), 11.8(171)
sine, 7.1(91)
Wolfram, 13.2(190)
z transform, 5.1(65)
ATTRIBUTIONS
198
Attributions
Collection:
Signals and Systems
Edited by: Marco F. Duarte

URL: http://legacy.cnx.org/content/col11557/1.10/
License: http://creativecommons.org/licenses/by/4.0/
Module: "Complex Numbers: Geometry of Complex Numbers"
Used here as: "Geometry of Complex Numbers"
By: Louis Scharf
URL: http://legacy.cnx.org/content/m21411/1.6/
Pages: 1-5
Copyright: Louis Scharf
Module: "Complex Numbers: Algebra of Complex Numbers"
By: Marco F. Duarte
Pages: 5-11
Copyright: Marco F. Duarte
Based on: Complex Numbers: Algebra of Complex Numbers
By: Louis Scharf
Module: "Complex Numbers: Representing Complex Numbers in a Vector Space"
Used here as: "Representing Complex Numbers in a Vector Space"
By: Louis Scharf
Pages: 11-15
Copyright: Louis Scharf
Module: "Signal Classications and Properties"
By: Marco F. Duarte
Pages: 17-22
Based on: Signal Classications and Properties
By: Melissa Selik, Richard Baraniuk, Michael Haag
Module: "Common Continuous Time Signals"
By: Marco F. Duarte
Pages: 22-25
Based on: Common Continuous Time Signals
By: Melissa Selik, Richard Baraniuk
ATTRIBUTIONS
199
Module: "Signal Operations"

By: Marco F. Duarte
Pages: 25-28
Based on: Signal Operations
By: Richard Baraniuk
Module: "Energy and Power of Continuous-Time Signals"
By: Marco F. Duarte
Pages: 28-30
Based on: Energy and Power
By: Anders Gjendemsj, Melissa Selik, Richard Baraniuk
Module: "Continuous Time Impulse Function"
Pages: 31-33
Copyright: Melissa Selik, Richard Baraniuk, Adam Blair
Module: "Continuous-Time Complex Exponential"
By: Marco F. Duarte
Pages: 33-36
Based on: Continuous Time Complex Exponential
By: Richard Baraniuk, Stephen Kruzick
Module: "Introduction to Systems"
By: Don Johnson
Pages: 37-39
Copyright: Don Johnson
License: http://creativecommons.org/licenses/by/1.0
Module: "System Classications and Properties"
By: Marco F. Duarte
Pages: 39-43
Based on: System Classications and Properties
By: Melissa Selik, Richard Baraniuk, Stephen Kruzick
ATTRIBUTIONS
200
Module: "Linear Time Invariant Systems"
By: Thanos Antoulas, JP Slavinsky
Pages: 43-50
Copyright: Thanos Antoulas, JP Slavinsky
Module: "Continuous Time Systems"
By: Marco F. Duarte
Pages: 51-52
Based on: Continuous Time Systems
By: Michael Haag, Richard Baraniuk, Stephen Kruzick
Module: "Continuous Time Impulse Response"
By: Dante Soares
Pages: 52-55
Copyright: Dante Soares
Module: "Continuous-Time Convolution"
By: Marco F. Duarte
Pages: 55-58
Based on: Continuous Time Convolution
By: Melissa Selik, Richard Baraniuk, Stephen Kruzick, Dan Calderon
Module: "Properties of Continuous Time Convolution"
By: Melissa Selik, Richard Baraniuk, Stephen Kruzick
Pages: 58-61
Copyright: Melissa Selik, Richard Baraniuk, Stephen Kruzick
Module: "Causality and Stability of Continuous-Time Linear Time-Invariant Systems"
By: Marco F. Duarte
Pages: 62-63
ATTRIBUTIONS
201
Module: "Introduction to Fourier Analysis"

By: Marco F. Duarte
Page: 65
Based on: Introduction to Fourier Analysis
Module: "Continuous Time Periodic Signals"
By: Marco F. Duarte
Pages: 66-67
Based on: Continuous Time Periodic Signals
By: Michael Haag, Justin Romberg
Module: "Eigenfunctions of Continuous-Time LTI Systems"
By: Marco F. Duarte
Pages: 68-69
Based on: Eigenfunctions of Continuous Time LTI Systems
By: Justin Romberg, Stephen Kruzick
Module: "Continuous Time Fourier Series (CTFS)"
By: Marco F. Duarte
Pages: 69-73
Based on: Continuous Time Fourier Series (CTFS)
By: Stephen Kruzick, Dan Calderon
Module: "Continuous Time Aperiodic Signals"
By: Marco F. Duarte
Pages: 75-76
Based on: Continuous Time Aperiodic Signals
By: Stephen Kruzick
ATTRIBUTIONS
202
Module: "Continuous Time Fourier Transform (CTFT)"

By: Marco F. Duarte
Pages: 76-79
Based on: Continuous Time Fourier Transform (CTFT)
By: Richard Baraniuk, Melissa Selik
Module: "Properties of the CTFT"
By: Marco F. Duarte
Pages: 79-82
Based on: Properties of the CTFT
Module: "Common Fourier Transforms"
By: Marco F. Duarte
Pages: 83-85
Based on: Common Fourier Transforms
Module: "Continuous Time Convolution and the CTFT"
By: Marco F. Duarte
Pages: 85-87
Based on: Continuous Time Convolution and the CTFT
By: Stephen Kruzick
Module: "Frequency-Domain Analysis of Linear Time-Invariant Systems"
By: Marco F. Duarte
Pages: 87-88
ATTRIBUTIONS
203
Module: "Common Discrete Time Signals"

By: Marco F. Duarte
Pages: 91-93
Based on: Common Discrete Time Signals
By: Don Johnson, Stephen Kruzick
Module: "Energy and Power of Discrete-Time Signals"
By: Marco F. Duarte
Pages: 93-94
Based on: Energy and Power
By: Anders Gjendemsj, Melissa Selik, Richard Baraniuk
Module: "Discrete-Time Signal Operations"
By: Marco F. Duarte
Pages: 95-98
Based on: Signal Operations
Module: "Discrete Time Impulse Function"
By: Marco F. Duarte
Pages: 98-100
Based on: Discrete Time Impulse Function
By: Dan Calderon
Module: "Discrete Time Complex Exponential"
By: Dan Calderon, Richard Baraniuk, Stephen Kruzick, Matthew Moravec
Pages: 100-104
Copyright: Dan Calderon, Stephen Kruzick
Based on: The Complex Exponential
ATTRIBUTIONS
204
Module: "Discrete Time Systems"
By: Marco F. Duarte
Pages: 105-107
Based on: Discrete Time Systems
By: Don Johnson, Stephen Kruzick
Module: "Discrete Time Impulse Response"
By: Marco F. Duarte
Pages: 107-109
Based on: Discrete Time Impulse Response
By: Dante Soares
Module: "Discrete-Time Convolution"
By: Marco F. Duarte
Pages: 110-115
Based on: Discrete Time Convolution
By: Ricardo Radaelli-Sanchez, Richard Baraniuk, Stephen Kruzick, Catherine Elder
Module: "Properties of Discrete Time Convolution"
By: Marco F. Duarte
Pages: 115-117
Based on: Properties of Discrete Time Convolution
By: Stephen Kruzick
Module: "Causality and Stability of Discrete-Time Linear Time-Invariant Systems"
By: Marco F. Duarte
Pages: 118-119
ATTRIBUTIONS
205
Module: "Discrete Time Aperiodic Signals"

By: Marco F. Duarte, Natesh Ganesh
Pages: 121-124
Based on: Discrete Time Aperiodic Signals
Module: "Eigenfunctions of Discrete Time LTI Systems"
By: Marco F. Duarte
Pages: 124-125
Based on: Eigenfunctions of Discrete Time LTI Systems
Module: "Discrete Time Fourier Transform (DTFT)"
Pages: 125-128
Based on: Discrete Time Fourier Transform (DTFT)
Module: "Properties of the DTFT"
Pages: 128-132
Based on: Properties of the DTFT
By: Don Johnson
Module: "Common Discrete Time Fourier Transforms"
By: Marco F. Duarte
Pages: 132-133
Based on: Common Discrete Time Fourier Transforms
By: Stephen Kruzick
ATTRIBUTIONS
206
Module: "Discrete Time Convolution and the DTFT"
By: Marco F. Duarte
Pages: 133-135
Based on: Discrete Time Convolution and the DTFT
Module: "Discrete Fourier Transform (DFT)"
By: Marco F. Duarte
Pages: 137-138
Based on: Discrete Fourier Transform (DFT)
By: Don Johnson
Module: "DFT: Fast Fourier Transform"
By: Don Johnson
Pages: 138-139
Copyright: Don Johnson
Module: "The Fast Fourier Transform (FFT)"
By: Marco F. Duarte
Pages: 139-144
Based on: The Fast Fourier Transform (FFT)
By: Justin Romberg
Module: "Signal Sampling"
By: Marco F. Duarte
Pages: 147-149
Based on: Signal Sampling
By: Stephen Kruzick, Justin Romberg
ATTRIBUTIONS
207
Module: "Sampling Theorem"

By: Marco F. Duarte
Pages: 149-152
Based on: Sampling Theorem
Module: "Signal Reconstruction"
By: Marco F. Duarte
Pages: 152-157
Based on: Signal Reconstruction
By: Stephen Kruzick, Justin Romberg
Module: "Perfect Reconstruction"
By: Marco F. Duarte
Pages: 157-159
Based on: Perfect Reconstruction
By: Stephen Kruzick, Roy Ha, Justin Romberg
Module: "Aliasing Phenomena"
By: Marco F. Duarte
Pages: 160-163
Based on: Aliasing Phenomena
By: Justin Romberg, Don Johnson, Stephen Kruzick
Module: "Anti-Aliasing Filters"
By: Marco F. Duarte
Pages: 163-165
Based on: Anti-Aliasing Filters
ATTRIBUTIONS
208
Module: "Changing Sampling Rates in Discrete Time"
By: Marco F. Duarte
Pages: 165-171
Module: "Discrete Time Processing of Continuous Time Signals"
Pages: 171-175
Based on: Discrete Time Processing of Continuous Time Signals
Module: "Linear Algebra: The Basics"
Used here as: "Basic Linear Algebra"
By: Michael Haag, Justin Romberg
Pages: 177-182
Copyright: Michael Haag, Justin Romberg
Module: "Linear Constant Coecient Dierence Equations"
Pages: 182-183
Copyright: Richard Baraniuk, Stephen Kruzick
Module: "Solving Linear Constant Coecient Dierence Equations"
Pages: 183-186
Copyright: Richard Baraniuk, Stephen Kruzick
Module: "Viewing Embedded LabVIEW Content in Connexions"
By: Stephen Kruzick
Page: 190
Copyright: Stephen Kruzick
Based on: Viewing Embedded LabVIEW Content
By: Matthew Hutchinson
ATTRIBUTIONS
209
Module: "Getting Started With Mathematica"

By: Catherine Elder, Dan Calderon
Pages: 190-192
Copyright: Catherine Elder, Dan Calderon
Signals and Systems

This course deals with signals, systems, and transforms, from their theoretical mathematical foundations to
practical implementation in circuits and computer algorithms. At the conclusion of ECE 313, you should
have a deep understanding of the mathematics and practical issues of signals in continuous and discrete time,
linear time invariant systems, convolution, and Fourier transforms.
About OpenStax-CNX
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Signals and Systems

Signals and Systems

8.4 Properties of Discrete Time Convolution . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 115

11 Sampling and Reconstruction

13 Appendix: Viewing Interactive Content

Available for free at Connexions <http://legacy.cnx.org/content/col11557/1.10>

note: This module is part of the collection,

A First Course in Electrical and Computer Engineer-

is used to build complex numbers

in the following way:

In MATLAB, the variable

has real part

and imaginary part

z = Re [z] + jIm [z]

and the variable

is called the in-phase component of

The imaginary number

is called the quadrature component. We

representation of z , with real component x and imaginary component y . We

say that the Cartesian pair

axis and the vertical axis the imaginary axis.

The plane is called the complex plane.

The radius and

content is available online at <http://legacy.cnx.org/content/m21411/1.6/>.

CHAPTER 1. REVIEW OF PREREQUISITES: COMPLEX NUMBERS

Cartesian and Polar Representations of the Complex Number z

The original Cartesian representation is obtained from the radius

The complex number

may therefore be written as

coded with the Cartesian pair

The Complex Number cos + jsin

Available for free at Connexions <http://legacy.cnx.org/content/col11557/1.10>

The complex number cos

of such fundamental importance to our study of complex numbers

that we give it the special symbol

has radius 1 and angle

With the symbol

z . We say that the polar pair rcodes the

representation for the complex number

In this polar representation, we dene

In "Roots of Quadratic Equations"

and record the corresponding geometric codes:

we show that the denition

is just the familiar function

is more than symbolic.

evaluated at the imaginary argument

a complex exponential, meaning that it is an exponential with an imaginary argument.

j3, j4, j5.

ej[(/2)+m2] = j, ej[(3/2)+m2] = j, ej(0+m2) = 1,

Find the polar representation

for each of the following complex numbers:

Plot the points on the complex plane.

Find the Cartesian representation

for each of the following complex numbers:

Numbers: Roots of Quadratic Equations" <http://legacy.cnx.org/content/m21415/latest/>

CHAPTER 1. REVIEW OF PREREQUISITES: COMPLEX NUMBERS

Plot the points on the complex plane.

The following geometric codes represent complex numbers.

Decode each by writing down the

Demo 1.1 (MATLAB). Run the following

MATLAB program in order to compute and plot the complex number

= 2/360, 2 (2/360) , ..., turns

out complex numbers that lie at angle

has real part

and imaginary part

is called the in-phase component of

is called the quadrature component. We

axis and the vertical axis the imaginary axis.

The plane is called the complex plane.

In this polar representation, we dene

we show that the denition

a complex exponential, meaning that it is an exponential with an imaginary argument.

is computed from a parallelogram rule, wherein

complex conjugates is to change

is the radius, or magnitude, that we dened in "Geometry of Complex Numbers"

In the eld of complex numbers, the complex number

In this eld, the complex number