C H A P T E R

1
Signals, Sequences, and
Systems

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Engineers, scientists, and mathematicians all use linear systems

theory because it is the foundation for building many of the
things we use in our daily lives. The theory of linear systems
introduced in this text provides powerful tools for analysis and
design. Many communication, control, and signal-processing
systems can be approximated by linear mathematical models,
and by applying linear systems techniques to these models,
we can design and develop better systems and shorten the
production cycle. In addition, computer simulation plays a
central role in applying linear systems theory, and there are now
available powerful and easy-to-use software packages, such as
LabVIEW and MATLAB® . DTSFA1 m-files that run with
either of these packages are used extensively in this book.
Computers are frequently used as elements of systems. This
means that we need to consider systems whose signals change
only at discrete-time instants, called discrete-time systems,
and also those systems whose signals vary continuously with
time, called continuous-time or analog systems. Discrete-
time systems operate on sequences, whereas continuous-time
systems process analog signals. Frequently, the sequences of
Figure 1-1: Claude E. Shannon.
interest in discrete-time systems analysis come from sampling
continuous-time signals.
1 DTSFA is shorthand for Discrete-Time Systems: Fundamentals and
Applications.
2 CHAPTER 1 SIGNALS, SEQUENCES, AND SYSTEMS

We begin our study of linear systems theory by first x(t)

considering the ways in which signals and sequences are
described. After considering the basic "building blocks" of
signals and sequences, we turn our attention to their use in
describing signals and sequences that occur in real-life systems. t
There are at least three good reasons for doing this. First, it
is convenient to be able to describe sequences by analytical (a) Analog input signal
mathematical expressions using compact notation. Second,
analytical descriptions of sequences are advantageous when
using the transform methods introduced in Chapter 2 to analyze Continuous-
x(t) time y(t)
and design linear systems. Finally, an analytical description of
(analog)
a sequence makes system simulation much easier than using a signal
tabulation of values.
Many systems contain both analog and discrete-time (or (b) Continuous-time (analog) system
digital) elements. Thus, there is a need to be able to
convert analog signals to sequences and vice versa. These y(t)
conversions are accomplished by analog-to-digital converters
(ADCs) and digital-to-analog converters (DACs), respectively.
Following a discussion of analog-to-digital and digital-to-
analog conversion, we consider the sampling theorem, based t
on the work of Claude E. Shannon (Figure 1-1), Harry Nyquist,
and others, that is of fundamental importance in converting (c) Analog output signal
continuous-time signals to sequences. Finally, we take a look
at where the rest of the book is going. Figure 1-2: A continuous-time system with its input and output
signals.

1.1 Types of Systems

by −3.5, and so forth, corresponding to the samples
Continuous-time systems operate on and generate signals that n = {0, 1, 2, 3, 4, . . .}, as shown in Figure 1-4(a). In addition
may vary over the entire time interval rather than just at discrete to knowing the sequence values, we also need to know a time
times, and these signals also have amplitudes that vary over a reference, and we generally will use (arbitrarily) n = 0. Using
continuous range of values. Such signals are called analog
signals. Figure 1-2 shows a typical analog input signal x(t), r(t)
a continuous-time system, and an analog output signal y(t).
Signals occurring in nature are often analog. As designers
wanting to process and use these signals, however, we have
a choice of whether to process them using continuous-time
(analog) systems or discrete-time (digital) systems. If we t
choose a discrete-time system, the continuous-time signals
must be converted to a discrete-time and discrete-amplitude (a) Analog signal
format. This is often accomplished by sampling the analog
signals at equally spaced time intervals to obtain sequences, r[n]
as illustrated in Figure 1-3. Collectively, these sample values
make up the sequence r[n]. Observe that we use the notation (t),
as in x(t) and y(t), to denote functions of the continuous-time
variable t (signals) and [n], as in r[n], to represent functions of
the discrete-time variable n (sequences).
n
Discrete-time systems generally receive inputs at equally
spaced (uniform) time intervals. The inputs are simply
numbers, but the order in which they are received makes (b) Sequence from sampled analog signal
a difference, so we represent the numbers as sequences.
For example, x[n] = {7.5, −3.5, 2.5, 6.0, 0, . . .} represents Figure 1-3: Analog signal and sequence resulting from the sampling
a sequence with the value 7.5 occurring first, followed process.
1.2 HISTORICAL PERSPECTIVE 3

x[n] signals. We often want to process these signals by using a digital

7.5 6.0 computer which is a discrete-time system, so they must first be
converted to sequences. Also, we often may need to re-convert
2.5
the output sequence from a discrete-time system to an analog
1 signal. The means for accomplishing these conversions of a
n
0 2 3
continuous-time signal to a discrete-time sequence and vice
−3.5 versa appear in Figure 1-5 as an analog-to-digital converter
(ADC), and a digital-to-analog converter (DAC). Notice how
(a) Input sequence
the signals and sequences are labeled in this diagram. The
analog input signal is denoted as x(t). The output of the analog-
x[n] Discrete-time y[n] to-digital conversion process is the sequence x[n], which is the
system input to the discrete-time system that modifies this sequence to
provide the output sequence y[n]. Assuming that an analog
(b) Discrete-time system output is desired, the sequence y[n] is then converted to a
continuous-time signal y(t) by the digital-to-analog converter.
y[n] Later in this chapter we consider more details concerning
6.0 analog-to-digital and digital-to-analog conversion.
5.0
3.0 2.5 1.2 Historical Perspective
4
n
0 1 2 3 The foundations for linear systems theory and applications
−1.5
can be traced to the 17th, 18th, and 19th centuries with
−4.5 results developed by Newton, Gauss, Laplace, Euler, Fourier,
−7.0 Lagrange, Laurent, and many others. Initially, analysis was
done by analytical methods or hand calculations. These
(c) An output sequence approaches were applied to the analysis and design of analog
systems whose input signals vary continuously with time (or
Figure 1-4: A discrete-time system and its input and output sequences. with other variables, such as space) and produce output signals
that also vary continuously with time, space, or other variables.
the sequence x[n] in Figure 1-4(a) as the input to a discrete- Figure 1-2 shows a conceptual representation of a continuous-
time system represented in Figure 1-4(b) results in a different time (or analog system) together with representations of typical
sequence, such as the output y[n] shown in Figure 1-4(c). input and output signals. As time evolved and system
The abscissas of the graphs of x[n] and y[n] portray time. complexity grew, the need for fast, accurate, and economical
The sequences are shown only at a discrete set of values, computational approaches increased.
because a discrete-time system operates on or generates signals Analog computers were developed and widely used for
only at these discrete instants. Since the sequence values simulation of continuous-time systems and as components of
are often obtained by sampling a continuous-time (or analog) systems through the late 1940s and early 1950s. The earliest
waveform at uniform intervals, we refer to the integer values analog computers were mechanical systems, but advances in
of n as the sampling times and the amplitudes 7.5, −3.5, 2.5, electronics technology soon made electronic analog computers
6.0, . . ., as the sample values. We’ll see later that the time the dominant form. In the 1950s and 1960s continuing progress
interval separating the samples is an important parameter in the in solid-state electronics and the eventual development of
sampling process. integrated circuits combined with the need for smaller, faster,
Inputs to systems come from sensors, radars, sonars, or and more accurate computing systems propelled computing
recordings, for example. These are continuous-time (or analog) into digital domains. Unlike analog systems, digital systems

Analog-to- Digital-to-
x(t) x[n] Discrete- y[n] y(t)
Digital Analog
Time
Converter Converter
System
(ADC) (DAC)

Figure 1-5: Using a discrete-time system to process a continuous-time signal.

4 CHAPTER 1
receive sequences of numbers as inputs and process them
with a numerical procedure (an algorithm) to produce another
sequence of numbers as outputs. The elements of the input and
output sequences generally change at uniformly spaced times,
and the amplitudes of the sequence values are quantized (that
is, they can take on only a finite or discrete set of values that
depend on the wordlength of the digital hardware). Figure 1-4
illustrates a digital or discrete-time system2 along with typical
input and output sequences. Early applications of digital
computers focused on simulations of analog systems as a way to
evaluate designs before committing to build systems. Another
application was in seismic exploration, where data was recorded
from sensors and later fed into a digital computer and processed.
The processing often took much longer than the time duration
of the recorded data. This was not a serious drawback, since
seismic structures do not change significantly except perhaps
over relatively long time periods.
As technology progressed, however, it became apparent
that digital computers have attributes that make them very
desirable as system elements as well as computational aids
to carry out analysis and design procedures. Initially, digital
computers were physically large, relatively slow, had limited
memory, were expensive, and required significant electric
power compared to today’s computers. This began to change
in the 1960s, and today digital computers and special-purpose
digital-processing hardware are small, fast, reliable, have
abundant memory, are inexpensive, and require small amounts
of electric power.
Although the earliest analysis and design approaches for
discrete-time systems were largely based on “translations”
or adaptations of analog methods, it soon became apparent
that the attributes of digital systems offered opportunities to
dramatically expand the repertoire of useful techniques. Today,
the applications of discrete-time systems are everywhere.
Systems for control, signal processing, and communications
rely on discrete-time systems technology for both design and
implementation. Speech processing, for example, includes
techniques for storing, transmitting, enhancing, compressing,
synthesizing and recognizing the content of speech, and
identifying a speaker. Interpreting seismic signals is a valuable
tool in exploration for oil and subterranean surface structures.
Radar signal processing is relied upon in both civilian and
military applications to identify and track aircraft, satellites,
and space vehicles. We frequently see road signs warning
us that our car speed may be tracked by radar. Image
processing has become widespread in television, movies and
the visual media that surround us. Medical imaging is often
used to detect and diagnose illnesses. Of course, mobile
phones, audio players, and portable reading tablets also rely on
2Although
there are subtle differences of significance in some
circumstances, we will use the terms discrete-time systems and digital systems
interchangeably.
SIGNALS, SEQUENCES, AND SYSTEMS
signal processing technology. Control systems to manufacture
products, command robots, and move equipment and objects
also depend on digital processing. Digital computers (or
special-purpose digital hardware) are essential elements in all
of these systems.
Discrete-time systems have many advantages over analog
systems. For example, the accuracy of a digital processor
can be improved by increasing the wordlength of the system
or using floating-point processing, whereas analog processing
accuracy depends on component tolerances, and component
values may also vary with temperature. In addition, the
sensitivity to noise of a digital system is generally better
than for an analog system. Another positive feature of a
discrete-time system is the capability to make changes in
processing functions by modifying the software or firmware
of the system; changes in analog processing, on the other
hand, generally require hardware component changes. Digital
systems are also amenable to implementing adaptive, nonlinear,
and time-varying processing algorithms, which is a capability
not generally available with analog systems. In addition, digital
processing enables encryption and decryption of system inputs
and outputs—something that cannot be provided by an analog
system. As a consequence of these and other advantages, even
analog signals are usually converted to a discrete-time format
and processed digitally. Depending on the system application,
the output sequence may be converted (or reconstructed) to
an analog signal. For example, in an audio system with
digital processing, the output would drive speakers, and this
necessitates conversion of the discrete-time output to an analog
signal, as shown in Figure 1-5.
We note that (in spite of continuing advances in speed, cost,
capability, and size of digital hardware) software development
continues to be a challenge. In addition, the need for conversion
of continuous-time signals to and from discrete-time sequences
adds complexity and cost in the form of analog-to-digital
and digital-to-analog converters and associated analog filters.
Finally, applications requiring very high speed processing still
may be beyond the capabilities of digital hardware.
1.3 Signals and Sequences
We are now ready to consider in detail continuous-time signals
that occur in analog (or continuous-time) systems and discrete-
time sequences in discrete-time systems. We will observe many
similar characteristics in signals and sequences, but also some
differences.
1.3.1 Signals (Continuous-Time Functions)
We begin with several signals that occur frequently in
continuous-time system applications and describe these signals
using analytical expressions (equations) and corresponding
graphical representations.
1.3 SIGNALS AND SEQUENCES 5

Unit Impulse g(Δ,␴)

1
The unit impulse is an important signal in the study of Δ
continuous-time systems. An arbitrary analog signal can be
approximated by impulses, and knowing the response of a
system to an impulse input enables us to find the forced response
0 ␴
to all other inputs. Denoted by δ(σ ), the unit impulse is not Δ
really a function in the normal sense. It is generally defined as
the limit of a function or through its properties. One commonly
used definition is (a) Unit pulse approximation

␦(␴)
Unit impulse

δ(σ ) = lim g(, σ ) (1.1)

→0

0 ␴

where g(, σ ) is a function such as the pulse shown in

Figure 1-6(a). As  → 0, the height of this pulse becomes
infinitely large, its base approaches zero, and its area is always 1. (b) Unit impulse symbol
Thus, we think of an impulse as having zero duration, infinite
␦(t − 1)
height, and unit area. Multiplying an impulse by a constant A
simply makes its area equal to A. We represent a unit impulse by
the symbol shown in Figure 1-6(b), where the numeral 1 next 1
to the arrow indicates the impulse’s area. Of course, we can
have impulses occurring at various times with different areas,
as shown in Figures 1-6(c) and (d). Notice that we use time, t, 0 t
which is measured in seconds, as the independent variable, but 1
there are other possibilities such as distance, pressure, etc.
A useful property of impulses is the sifting property, which
(c) Shifted unit impulse
is
−2.5␦(t − 3.4)

Sifting property
∞
p(t)δ(t − t1 )dt = p(t1 ) (1.2)
−∞
3.4
0 t

at all values of t1 for which p(t) is continuous. This property is

−2.5
easily justified by using a limiting argument based on the pulse
approximation to the impulse shown in Figure 1-6(a). (d) Shifted and scaled impulse

Figure 1-6: Impulse functions.

Arbitrary Functions
shown. As the number of impulses approaches infinity and the
There is no analytical way of exactly describing an arbitrary interval between them approaches zero we have
analog waveshape such as the one in Figure 1-7(a); however,
equally spaced impulses can be used as an approximate ∞
representation, shown as s(t) in Figure 1-7(b). The impulses g(t) = g(τ )δ(t − τ )dτ, (1.3)
have the areas associated with the function g(t) in the intervals −∞
6 CHAPTER 1 SIGNALS, SEQUENCES, AND SYSTEMS

g(t) 2 2

1.5 1.5

u(t + 1) − u(t − 3)
1 1

u(t)
t 0.5 0.5

0 0
(a) An arbitrary signal
−0.5 −0.5
s(t)
−1 −1
−2 0 2 4 −2 0 2 4
Time, t Time, t
(a) Unit step function (b) Pulse formed by two
unit step functions
t
4 4
3.5 3.5
3 3

g(t) [u(t + 1) − u(t − 3)]

(b) Approximation of g(t) by a sum of impulses
2.5 2.5
Figure 1-7: Approximation of a signal by a sum of impulses. 2 2
g(t)

1.5 1.5
which is an exact representation of g(t) for all values of t. 1 1
0.5 0.5
Unit Step Function
0 0
Figure 1-8(a) pictures a unit step function that “turns on” at −0.5 −0.5
t = 0 and is defined by −1 −1
−2 0 2 4 −2 0 2 4
Time, t Time, t
Unit step function (c) A cosine with a (d) Signal g(t) multiplied
 dc component by pulse function
1, 0≤t
u(t) = (1.4) Figure 1-8: A unit step function and an application.
0, t <0

a way to "turn on" and "off" other functions. For example,

Figure 1-8(b) shows the pulse function [u(t + 1) − u(t − 3)]
Notice that the unit step is related to the unit impulse by
that turns on at t = −1 and off at t = 3. The signal consisting
t of the difference between a constant and a sinusoid
u(t) = δ(τ )dτ. (1.5)
g(t) = 2.5 − cos(5t) (1.7)
−∞
is plotted in Figure 1-8(c). Figure 1-8(d) shows the result of
Of course, we can scale and time-shift step functions to obtain
multiplying g(t) by the pulse signal of Figure 1-8(b). Thus, we
can think of step functions as mathematical switches.
General step function Comment:
 Figure 1-8 was obtained by running DTSFA file F1_8. It is
B, t0 ≤ t
Bu(t − t0 ) = , (1.6) recommended that you access this file and review the statements
0, t < t0 along with the explanatory comments. Then run this file
yourself using MathScript in LabVIEW or MATLAB® . As
we proceed, there will be many m-files available, and you are
which is equal to B for t0 ≤ t and zero otherwise. In addition encouraged to take advantage of them. Look for these files on
to being useful in their own right, step functions also provide the DTSFA website (see Preface).
1.3 SIGNALS AND SEQUENCES 7

Ramp Functions 20
18 g1(t)
A shifted ramp function g(t) with slope B is defined as g2(t)
16
14
g2(t) = 2e(−1.2t)

g1(t) and g2(t)

Shifted ramp function 12
10
g(t) = Bt − Bt0 = B · (t − t0 ). (1.8) 8
6
4 g1(t) = 2e(−0.5t)
Notice that this is the equation of a straight line with a slope
2
of B with an ordinate intercept of −Bt0 . Figure 1-9(a) shows
0
a ramp with B = 2 and t0 = 3. To obtain a unit ramp function
−2 0 2 4
that passes through the origin, has a slope of 1, and “begins” Time, t
or turns on at t = 0, we make B = 1, t0 = 0, and multiply by
(a) Negative exponents
u(t) giving
20
Unit ramp function g3(t)
18
 g4(t)
16
0, t < 0
r(t) = tu(t) = (1.9) 14 g4(t) = 1.5e(1.2t)
t, 0 ≤ t,

g3(t) and g4(t)

12
10
as shown in Figure 1-9(b). 8
6
Exponential Functions 4
g3(t) = 1.5e(0.5t)
A function given by 2
0
Exponential function −4 −2 0 2
Time, t
g(t) = Aeβt , (1.10) (b) Positive exponents

Figure 1-10: Real exponentials.

where e is the Naperian constant 2.718 . . ., is called an
exponential function. A and β are constants that may be real
or complex. Let’s consider several of the possibilities for A Real Exponential Functions
and β.
First, assume that A and β in Eq. (1.10) are both real numbers
4 6
g(t) = B(t − t0) r(t) = t u(t) giving a real exponential function. Exponentials for a few
2 = 2(t − 3) 5 values of A and β are shown in Figure 1-10.
0 4
−2 Complex Exponential Functions
3
g(t)

r(t)

−4
2 If A is real and β is imaginary, that is β = j ω0 with ω0 a real
−6 constant, we have
1
−8
0 g(t) = Aej ω0 t = Aej 2πf0 t . (1.11)
−10
−12 −1 In this case, g(t) is a complex-valued function of t. The quantity
−2 0 2 4 −2 0 2 4
Time, t Time, t ω0 is called the radian frequency and has units of radians/second
(a) Scaled, shifted (b) Unit ramp (rad/s); f0 is called the frequency and has units of hertz (Hz) or
ramp function function cycles/second.3 Clearly, ω0 = 2πf0 or f0 = ω0 /2π.
3A hertz has dimensions of cycles/second, and multiplication by 2π
Figure 1-9: Two ramp functions. radians/cycle yields a quantity with dimensions radians per second.
8 CHAPTER 1 SIGNALS, SEQUENCES, AND SYSTEMS

The function g(t) can be visualized as the vectors shown Im

on the complex planes of Figure 1-11 for various values of t.
Radius = A
Two situations are illustrated: one where ω0 = π/4 and another t = 2,10,…
where ω0 = 0.2. Notice that for the values of t selected both t = 3,11,… t = 1, 9,…
sets of vectors begin to repeat themselves after different values
t
of elapsed time. This repetition of a function is a property
called periodicity, and we say that g(t) is a periodic signal t = 4,12,… Re
with a period of T0 seconds. In general, if we have a signal t = 0, 8,…
p(t) for which there is an interval T0 such that
t = 5,13,… t = 7,15,…
Periodic signal t = 6,14,…

p(t + T0 ) = p(t) (1.12) (a) ␻0 = ␲/4

for all values of time, t, then p(t) is a periodic signal with Im

period T0 . Notice that T0 is the smallest positive value for
Radius = A t = 10␲ ,50␲ ,…
which the equality in Eq. (1.12) is satisfied. To determine if 4 4
g(t) in Eq. (1.11) is periodic, we consider whether or not there
t = 15␲ ,55␲ ,… t = 5␲ ,45␲ ,…
is a value of T0 for which 4 4 4 4
t
g(t + T0 ) = g(t). (1.13) t = 5␲,15␲,… Re
t = 0, 10␲,…
Using the properties of exponentials we have

Aej ω0 (t+T0 ) = Aej ω0 t t = 25␲ ,65␲ ,… t = 35␲ ,75␲ ,…

(1.14) 4 4 4 4
Aej ω0 t · ej ω0 T0 = Aej ω0 t , t = 30␲ ,70␲ ,…
4 4
and for this to be satisfied for all t requires that ej ω0 T0 = 1. (b) ␻0 = 0.2
This will be the case if and only if ej ω0 T0 = ej 2mπ , where m
is an integer. So we have that ω0 T0 = 2mπ, and since T0 is Figure 1-11: Representation of Aej ω0 t as rotating vectors.
the smallest positive value for which Eq. (1.12) is satisfied, we
select m = 1, and the period T0 is given by
If we now allow the constant A to be a complex quantity (i.e.,
T0 = 2π/ω0 or T0 = 1/f0 . (1.15) A = |A|ej ϕ ) again with β = j ω0 , we have a slightly different
situation. Again starting with g(t) = Aeβt , we have
For the values of ω0 shown in Figure 1-11(a) and (b), the periods
are T0 = 8 and T0 = 10π, respectively. Notice that these are the g(t) = |A|ej ϕ ej ω0 t
values of t when the vectors have made one complete rotation
= |A|ej (ω0 t+ϕ) (1.17)
and again coincide with the positive real axis.
= |A| cos(ω0 t + ϕ) + j |A| sin(ω0 t + ϕ).
From the Euler Identities (see Appendix B), we can write

Aej ω0 t = A cos(ω0 t) + j A sin(ω0 t) (1.16)
and we observe that {real part of [Aej ω0 t ]} = Re[Aej ω0 t ] =
A cos(ω0 t) and {imaginary part of [Aej ω0 t ]} = Im[Aej ω0 t ] =
A sin(ω0 t). From Figure 1-11 we see that the real parts of the
vectors representing Aej ω0 t are the projections of these vectors
onto the real (horizontal) axis and the imaginary parts are the
projections onto the imaginary (vertical) axis. Notice that the
angles made by these vectors with the positive real axis are the
values of ω0 t.
The effect of the phase shift ϕ on the diagrams in Figure 1-11
is to rotate each of the vectors by ϕ radians—counterclockwise
for ϕ equal to a positive value or clockwise for ϕ negative.
Figure 1-12 shows a plot generated by DTSFA m-file F1_12
of the real part of g(t) from Eq (1.17) with A = 5e−j π/3 , ω0 =
20π. Clearly, this is a periodic function, and the period easily
can be measured as shown. As found previously, the period is
given by
T0 = 2π/ω0 = 2π/20π = 0.1 sec . (1.18)
1.3 SIGNALS AND SEQUENCES 9

T0 = 0.1 Other useful expressions for the real and imaginary parts of
g(t) are
5
4
3 Exponentially-modulated sinusoidal signals
␾ = −␲/3
2  
ej (ω0 t+ϕ) + e−j (ω0 t+ϕ)
1 |A|e cos(ω0 t + ϕ) = |A|e
αt αt
2
g(t)

0 T0 = 0.1
−1
 
−2 ej (ω0 t+ϕ) − e−j (ω0 t+ϕ)
|A|e sin(ω0 t + ϕ) = |A|e
αt αt
.
−3 2j
−4
(1.21)
−5
−0.1 −0.05 0 0.05 0.1 0.15 0.2
t, sec The expressions in Eq. (1.21) frequently occur in linear,
continuous-time systems and are known as exponentially
Figure 1-12: A continuous-time sinusoidal signal. modulated sinusoidal signals. These expressions are obtained
by applying Euler’s identity to expand the definitions of the
cosine and sine functions in terms of exponentials. Several
special cases easily can be deduced by substituting α = 0,
We can also evaluate the phase ϕ as shown in Figure 1-12, and ω0 = 0, or ϕ = 0 as appropriate in these general expressions.
we find that it is indeed −π/3 by determining the value of the Figure 1-13 illustrates two possibilities with positive and
cosine function at t = 0. negative values of α. The dotted curves represent the
At this point it is appropriate to summarize the units of exponential terms that multiply the sinusoid; these terms are
the various quantities in Eq. (1.17). We will use the referred to as the envelope. Notice that exponentially modulated
meter-kilogram-second (MKS) system. The function g(t) has sinusoidal signals are periodic only if α = 0.
whatever units are appropriate for the physical quantity being It is always good practice when relying on computer
represented, such as volts, amperes, watts, joules, meters per programs to ask if the results are reasonable. Figure 1-13
second, and so forth. Time, t, is measured in seconds, and the was generated by running DTSFA file F1_13, and we observe
units of ω0 are radians/second. that the exponential envelope (the term eαt ) decreases as time
Another variation on the complex exponential theme we have increases when the exponent is negative and the envelope
been pursuing is letting increases with increasing time when the exponent is positive.
We can also check a couple of points. For example, at t = 0,
2e−0.6(0) = 2, 2e0.3(0) = 2, and cos(0 + 1.047) = 0.5. This
A = |A|ej ϕ and β = α + j ω0 (1.19)
indicates that the values of g1 (t) and g2 (t) should be 1.0 at
t = 0, as shown on the plots. Other points may be verified in a
in Eq. (1.10) with α and ω0 as real numbers. In this case, both similar fashion.
A and β are complex quantities and

g(t) = Aeβt = |A|ej ϕ e(α+j ω0 )t

= |A|eαt cos(ω0 t + ϕ) + j |A|eαt sin(ω0 t + ϕ)
= |A|eαt cos(2πf0 t + ϕ) + j |A|eαt sin(2πf0 t + ϕ)
= |A|eαt cos(2πt/T0 + ϕ) + j |A|eαt sin(2πt/T0 + ϕ),
(1.20)
where the relationships among ω0 , f0 , and T0 are displayed
explicitly.
Piecewise-Linear and Periodic Functions
Equations readily can be obtained for signals that are composed
of piecewise-linear segments. Example 1-1 illustrates a
procedure for accomplishing this. In addition, a process for
writing an analytical expression for a periodic signal is also
demonstrated. (Sinusoidal signals are periodic, but they are
not the only periodic signals that occur.)
10 CHAPTER 1 SIGNALS, SEQUENCES, AND SYSTEMS

10 g1(t)
8
A
6 2e−0.6t
4
2 t
0 2
g1(t)

0
−2 (a) A sawtooth signal
g1(t) = 2e−0.6tcos(πt + 1.047)
−4
−2e−0.6t g1p(t)
−6
−8 A
−10
−2 0 2 4
Time, t t
−4 −2 0 2 4 6
(a) Exponential decreasing
with t increasing (b) A periodic sawtooth signal

10 g2(t)
8
B
6
4
2 2e0.3t
t
g2(t)

0 0 1 4
g2(t) = 2e0.3tcos(πt + 1.047)
−2 (c) A triangular pulse
−4 −2e0.3t
−6 Figure 1-14: Signals for Example 1-1.
−8
−10 result, using the subscript p to denote periodic, is
−2 0 2 4
Time, t ∞
 A
g1p (t) = (t − 2m)[u(t − 2m) − u(t − 2m − 2)].
(b) Exponential increasing
m=−∞
2
with t increasing
where m is an integer.
Figure 1-13: Exponentially modulated sinusoidal signals.
(c) We can represent the rising pulse edge by
Bt[u(t) − u(t − 1)] and the falling edge by
Example 1-1: Describing Continuous-Time Signals −[B/3][t − 4][u(t − 1) − u(t − 4)]; adding these
gives
Use the functions defined previously to determine analytical
expressions for the signals shown in Figure 1-14. B
g2 (t) = Bt[u(t)−u(t −1)]− [t −4][u(t −1)−u(t −4)]
3
Solution:
which can be simplified as
(a) The expression for the signal in the interval 0 ≤ t < 2 is 4B B
(A/2)t, so we need only “turn on” this signal at t = 0 and g2 (t) = Btu(t) − [t − 1]u(t − 1) + [t − 4]u(t − 4).
3 3
“turn it off” again at t = 2, giving

A
g1 (t) = t [u(t) − u(t − 2)] .
2
1.3.2 Sequences (Discrete-Time Functions)
(b) Here we have a repeating, or periodic signal with a period
of 2. We need only to replicate the triangular pulse of We now consider sequences that occur and are processed in
part (a) an infinite number of times with each replica discrete-time systems. As with signals found in continuous-
displaced by two time units from the previous one. The time systems, our focus will be on how to describe sequences
1.3 SIGNALS AND SEQUENCES 11

using analytical expressions (equations) and connecting these Arbitrary Sequences

expressions with their graphical representations.
An arbitrary sequence can be described by a summation of
weighted unit impulse sequences; that is,
Unit Impulse Sequence

The unit impulse sequence is defined as

Description of any sequence
∞

Unit impulse sequence g[n] = g[n]δ[n − m]. (1.23)
 m=−∞
1, n−m=0
δ[n − m] = (1.22)
0, n − m  = 0,
For example, the system input x[n] in Figure 1-4(a) on page 3
is described by
where n and m are integers. The notation [n − m] is used to
indicate that we are dealing with a function of a discrete-time x[n] = 7.5δ[n]−3.5δ[n−1]+2.5δ[n−2]+6.0δ[n−3]. (1.24)
integer variable n − m. Notice that, unlike a continuous-time
unit impulse function, there is no approximation or limiting
argument involved for the discrete-time impulse sequence. We Unit Step Sequence
can represent an impulse sequence of arbitrary amplitude A by
simply multiplying a unit impulse or a shifted unit impulse (as Figure 1-16(a) shows a unit step sequence defined as
appropriate) by A to obtain the sequence A[n] or Aδ[n − n0 ].
This makes it possible to represent an arbitrary sequence as the
weighted sum of shifted unit impulse sequences. Figure 1-15 Unit step sequence
shows the unit impulse sequence δ[n] and the shifted unit 
impulse sequence δ[n − 3] plotted as functions of n. 1, 0 ≤ n
u[n] = (1.25)
When δ[n] is used as the input to a discrete-time system, the 0, n < 0.
resulting output is referred to as the unit impulse response. As
we shall see subsequently, if the response of a discrete-time
system to a unit impulse input is known, the system’s response
Notice that the unit step sequence u[n] is related to the unit
to an arbitrary input can be determined.
impulse sequence δ[n] by

␦[n] 
n
u[n] = δ[m]. (1.26)
1 m=−∞

... ... As with the unit impulse sequence, we can shift and scale the
n step sequence to obtain
0

(a) Unit impulse sequence

General step sequence
␦[n + 3]

B, n0 ≤ n
x[n] = B · u[n − n0 ] = (1.27)
1 0, n < n0 ,

n
0 1 2 3
as shown in Figure 1-16(b) for B = −5 and n0 = −3. It is
(b) Shifted unit impulse sequence also possible to generalize the definition of the step sequence
by considering the sequence given by B · u[−n − n0 ]. This
Figure 1-15: A unit impulse sequence and a shifted unit impulse sequence will be zero for [−n − n0 ] < 0 or −n0 < n and will
sequence. be equal to B for n ≤ −n0 .
12 CHAPTER 1 SIGNALS, SEQUENCES, AND SYSTEMS

2
20

1.5 10
g[n] = 2[n − 10]
1 0

−10
u[n]

g[n]
0.5
−20
0
−30
−0.5
−40
−1 −10 −5 0 5 10 15 20
−10 −5 0 5 10 15 20
Sample number, n
Sample number, n
(a) Unit step sequence Figure 1-17: Shifted ramp sequence.
2
Exponential Sequences
1
x[n] = −5u[n + 3] A sequence characterized by
0
−1
Exponential sequence
−2
x[n]

g[n] = A(a)n (1.29)

−3
−4
is called an exponential sequence. A and a are constants that
−5
may be real or complex. Let’s consider some possibilities for
−6 A and a.
−7
−10 −5 0 5 10 15 20 Real Exponential Sequences:
Sample number, n We begin by assuming that A and a are real constants. Fig-
(b) Shifted step sequence B = −5, n0 = −3 ure 1-18 shows the two exponential sequences ga [n] = 4(0.9)n
and gb [n] = 4(−0.9)n . As n → ∞, these sequences (both of
Figure 1-16: Step sequences. which have |a| < 1) approach zero as n → ∞; whereas for
n → −∞, the sequences approach ±∞. For gc [n] and gd [n]
in Figure 1-18 a = ±1.1, and these sequences with |a| > 1
Ramp Sequences approach ±∞ as n → ∞ and approach 0 as n → −∞. By
multiplying (point by point) the sequences ga [n] and gd [n] by
A shifted ramp sequence with slope B and defined as unit step sequences u[n], we form the composite sequences
ge [n] and gf [n], which are zero for n < 0.

Ramp sequence Complex Exponential Sequences:

g[n] = B · (n − n0 )
is shown in Figure 1-17 for B = 2 and n0 = 10. For a unit
ramp sequence that is zero for n ≤ 0, we make B = 1, n0 = 0,
and multiply by u[n].
Next, having pursued a similar line of reasoning for continuous-
(1.28)
time complex exponential functions, we move on to the general
situation in which both A and a in g[n] = A(a)n are known
complex constants, specifically A = |A|ej ϕ and a = rej ω̂0 ,
where r, ϕ, and ω̂0 are real constants. In this case,
g[n] = A(a)n = |A|ej ϕ (rej ω̂0 )n = |A|r n ej (nω̂0 +ϕ) . (1.30)
1.3 SIGNALS AND SEQUENCES 13

10 10
Focusing on the units of the quantities appearing in this
expression, we observe that the units of g[n] and hence those
of Ar n are of whatever physical quantity g[n] represents (e.g.,
5 5
volts, meters, meters/second, and so forth). The argument that
appears in the cosine and sine functions is in radians, and the
ga[n]

gb[n]
0 0 phase shift, ϕ, is also in radians. The integer n has units of
samples, so the discrete-time angular frequency ω̂0 is expressed
−5 −5 in radians per sample (rad/smp).4
Equation (1.31) is a general expression that allows us to
−10 −10 investigate several special cases. Defining g1 [n] as the sequence
that results by letting r = 1 and ϕ = 0 in g[n] and taking the
−10 −5 0 5 10 −10 −5 0 5 10 real part, we have
Sample number, n Sample number, n
(a) ga[n] = 4(0.9)n (b) gb[n] = 4(−0.9)n g1 [n] = |A| cos(nω̂0 ), (1.32)

which is a cosine sequence with magnitude |A| and angular

10 10 frequency ω̂0 . Let’s consider Example 1-2 that illustrates a
surprising characteristic of sinusoidal sequences.
5 5
Example 1-2: Characteristic of Cosine Sequences
gd[n]
gc[n]

0 0

−5 −5 (a) Write an m-file to calculate and plot the cosine sequence

ga [n] = A cos(nω̂0 ) with A = 5, ω̂0 = π/8, and −18 ≤
−10 −10 n ≤ 18. What is the period of this sequence?

−10 −5 0 5 10 −10 −5 0 5 10 (b) Repeat part (a) with A = 5, ω̂0 = 0.8, and −18 ≤ n ≤ 18.
Sample number, n Sample number, n What is the period of this sequence?
(c) gc[n] = 4(1.1)n (d) gd[n] = 4(−1.1)n
(c) On the plot of part (b), superimpose a plot of gc (t) =
5 cos(0.8t) for −18 ≤ t ≤ 18. Compare the plots of parts
(b) and (c).
10 10

Solution:
5 5
(a) Figure 1-19(a) is a plot of the sequence ga [n]. This plot
ge[n]

gf[n]

0 0 was obtained from the segment of DTSFA file F1_19 that

follows.
−5 −5
m-file
−10 −10 % F1-19: Some cosine sequences
set(gcf,’DefaultLineLineWidth’,2);
−10 −5 0 5 10 −10 −5 0 5 10 % set default line width to 2 units
Sample number, n Sample number, n
(e) ge[n] = ga[n] • u[n] (f) gf[n] = gd[n] • u[n] ns=-18; % starting sample number
nf=18; % final sample number
Figure 1-18: Real exponential sequences. n=ns:1:nf; % sample values
wzhat=pi/8; % angular frequency in radians
As previously, we can use the Euler identity to write gan=5*cos(n*wzhat); % set cosine values
stem(n,gan); % obtain a lollipop plot of a sequence
g[n] = |A|r n {Re[ej (nω̂0 +ϕ) ] + j I m[ej (nω̂0 +ϕ) ]} 4An alternative also used is to define n as an integer without units, in which
(1.31)
= |A|r n cos(nω̂0 + ϕ) + j |A|r n sin(nω̂0 + ϕ). case the units of ω̂0 are radians.
14 CHAPTER 1 SIGNALS, SEQUENCES, AND SYSTEMS

6 axis([-19 19 -6 6]); % set axis limits

xlabel (’sample number, n’); % x-axis label
4 ylabel (’g_a[n]’); % y-axis label
%title( ’(a): the cosine sequence g_a[n]
2 =5cos(n\pi/8’); % title (suppressed)
grid; % place grid on plot
ga[n]

0
By inspecting the plot, we observe that the sequence is
−2 periodic and begins to repeat itself every 16 samples.

−4 Comment:
The graphics function stem is a built-in m-function of
−6 LabVIEW and MATLAB® , software applications that
−15 −10 −5 0 5 10 15
Sample number, n provide many such functions. When referring to these
(a) Cosine sequence ga[n] = 5 cos(nπ/8) m-functions we use boldface type, and the most
significant ones are listed at the end of each chapter.
6
(b) In Figure 1-19(b), the sequence gb [n] = 5 cos(0.8n) is
4 plotted for −18 ≤ n ≤ 18. Inspecting this graph, we
observe that the sequence is not periodic, at least over
2 the interval shown. In fact, this sequence is not periodic,
period!
gb[n]

0
(c) To get an idea of what’s going on here, Figure 1-19(c)
−2 again shows the sequence gb [n], and it also shows the
continuous-time signal gc (t) = 5 cos(0.8t) for −18 ≤ t ≤
18. But t is a continuous-time variable; it is not restricted
−4
to integer values as n is.5 We observe that gc (t) is periodic
with a period of approximately eight units. (Without
−6 knowing the time scale on the plot, we cannot state the
−15 −10 −5 0 5 10 15
Sample number, n period in seconds.)
(b) Cosine sequence gb[n] = 5 cos(0.8n)
6

4
Example 1-2 illustrates that a cosine sequence isn’t always
periodic so let’s consider the conditions that are required for
periodicity. Similar to the continuous-time situation, we define
2
gb[n] and gc(t)

a periodic sequence as one whose values repeat themselves after

an interval of N samples. That is a sequence g[n] is periodic if
0 there exists an integer N for which

−2 g[n + N ] = g[n] (1.33)

−4 for all integers n. N is the smallest integer for which Eq. (1.33)
holds. For a cosine sequence, this requires that
−6
−15 −10 −5 0 5 10 15 cos(nω̂0 ) = cos([n + N ]ω̂0 ). (1.34)
Sample number, n
(c) Sequence gb[n] = 5 cos(nπ/8) 5Actually, since these plots were obtained by using a PC, all of the variables

and the signal gc(t) = 5 cos(0.8t) are discrete, but in this case, the value 0.005 was used for the time increment
to obtain the plot of gc (t). Thus, this plot is approximately a continuous-time
version of the cosine signal gc (t).
Figure 1-19: Some cosine sequences.
1.3 SIGNALS AND SEQUENCES 15

For the equality to be true, N ω̂0 must be an integer multiple of Therefore, if 0 ≤ ω̂0 < 2π , there will always be an infinite
2π number of sinusoidal sequences with frequencies outside of
N ω̂0 = 2πm (1.35) this range that have identical sample values. Conversely,
if ω̂0 lies outside of the range 0 ≤ ω̂0 < 2π , there will
(i.e., where m is an integer). This in turn requires that
always be an integer  and a frequency ω̂0 + 2π for which
ω̂0 = 2π m/N, (1.36) 0 ≤ ω̂0 + 2π < 2π . Figure 1-20(a) shows samples of
three continuous-time sinusoidal signals. These sinusoidal
which tells us that ω̂0 must be a rational multiple6 of 2π if g[n] sequences of samples have discrete-time angular frequencies
is to be a periodic sequence. If this is the case then the period of ω̂ = ω̂0 = 3π/4 rad/s, ω̂0 + 2π = 11π/4 rad/s, and
N is given by ω̂0 + 6π = 27π/4 rad/s. As shown, these three sets of sample
values are identical. Figure 1-20(b) shows the first three
samples from Figure 1-20(a) along with three continuous-time
Period of a periodic sinusoidal sequence sinusoidal signals (represented by the dashed curves) that have
the same sample values.
N = 2πm/ω̂0 . (1.37)

6
Comparing Eq. (1.37) with Eq. (1.15)
4
T0 = 2π/ω0 , (1.38)
g1[n], g2[n], and g3[n] 2
we see that they have a similar structure; however, the period of
the cosine signal T0 in the continuous-time case is a real number,
0
whereas N in the discrete-time case must be an integer.
Looking back at Example 1-2 part (a), we have ω̂0 = 2π/16,
which satisfies Eq. (1.36) for m = 1 and N = 16. Clearly, −2
this relationship is also satisfied for m = 2, and N = 32, etc.,
but it is the smallest value of N that we seek. In part (b), −4
however, ω̂0 = 0.8, and there are no integer values of m for
which N = 2π m/0.8 = 2.5πm will be an integer, because π −6
−2 0 2 4 6 8 10 n
is not a rational number.
So a cosine sequence, even one resulting from sampling (a) Three sinusoidal sequences having the same values
a continuous-time sinusoid that is guaranteed to be periodic,
g1[n] and g1(t), g2[n] and g2(t), and g3[n] and g3(t)

may not be periodic. Fortunately, this does not cause practical 6

limitations in processing these sequences, periodic or not.
Another characteristic of discrete-time sinusoids that does 4
not occur for continuous-time sinusoids is that there are an
infinite number of angular frequencies ω̂0 , ω̂1 , ω̂2 , . . ., for which
2
cos(nω̂0 ) = cos(nω̂1 ) = cos(nω̂2 ) = . . . . (1.39)
0
In particular, if we select ω̂1 = ω̂0 + 2π, ω̂2 = ω̂0 + 4π ,
. . ., ω̂ = ω̂0 + 2π, where  is an integer, then, using the −2
trigonometric identity for cos(α ± γ ), we obtain
−4
cos(n[ω̂0 +2π]) = cos(nω̂0 ) cos(n2π)−sin(nω̂0 ) sin(n2π ).
(1.40)
Since n and  are integers, cos(n2π) = 1 and sin(n2π ) = 0. −6
−2 −1 0 n
So for all integer n, we have
(b) Three continuous-time sinusoids with the same sample values
cos(n[ω̂0 + 2π ]) = cos(nω̂0 ). (1.41)
6A rational multiple is the ratio of two integers; the denominator integer Figure 1-20: Sinusoidal sequences of different frequencies but with
cannot be zero. identical sample values.
16 CHAPTER 1 SIGNALS, SEQUENCES, AND SYSTEMS

Although it may be unsettling to find that many continuous- g1[n] = 2(1.1)n cos(nπ/8 + π/3) g2[n] = 2(−1.1)n cos(nπ/8 + π/3)
time sinusoidal signals have the same sample values, we shall
see in Section 1.5 on the sampling theorem how this issue is 25 25
20 2  (1.1)n 20 2  (1.1)n
addressed.
15 15
Finally, let us consider the general exponential sequence
10 10
5 5
g[n] = |A|r n {Re[ej (nω̂0 +ϕ) ] + j Im[ej (nω̂0 +ϕ) ]}

g1[n]

g2[n]
(1.42) 0 0
= |A|r n cos(nω̂0 + ϕ) + j |A|r n sin(nω̂0 + ϕ). −5 −5
−10 −10
As previously assumed, we have −15 −15
−20 −2  (1.1)n −20 −2  (1.1)n
A = |A|ej ϕ and a = rej ω̂0 , (1.43) −25 −25
−20 0 20 n −20 0 20 n
where |A|, ϕ, r, and ω̂0 are known, real constants. Again
focusing on the real part
g3[n] = 2(0.95)n cos(nπ/8 + π/3) g4[n] = 2(−0.95)n cos(nπ/8 + π/3)
Exponentially modulated sinusoidal sequence 10 10
8 8
2  (0.95)n 2  (0.95)n
|A|r n cos(nω̂0 + ϕ), (1.44) 6 6
4 4
2 2
we have what is called an exponentially modulated sinusoidal
g3[n]

g4[n]
0 0
sequence. The values of |A| cos(nω̂0 + ϕ) vary between ±|A|.
-2 -2
The factor r n , however, can cause the sequence to increase
-4 -4
without bound or decay to zero as n → ∞, depending whether
-6 -6
1 < |r| or |r| < 1, respectively.7 We also need to consider the −2  (0.95)n −2  (0.95)n
possibility that r may be negative, in which case the sign of -8 -8
the sequence will generally alternate from sample to sample. -10 -10
−20 0 20 n −20 0 20 n
Figure 1-21 shows plots with A = 2.0, ω̂0 = π/8, ϕ = π/3,
r = ±0.95, and r = ±1.1. The dashed curves, collectively Figure 1-21: Exponentially modulated sinusoidal sequences.
referred to as the envelope, are the exponentials ±|Ar n | that
modulate the sinusoidal sequence.
Another relationship that occurs frequently in the analysis of analytical relationships for sequences specified by graphical
linear, discrete-time systems is obtained by expanding cosine representations.
and sine sequences in terms of exponentials as
Example 1-3: Writing Analytical Expressions That
Exponentially modulated sinusoidal sequences Describe Sequences
 
j (nω̂0 +ϕ) + e−j (nω̂0 +ϕ) Use the sequences defined previously to describe analytically
n e
|A|r cos(nω̂0 + ϕ) = |A|r
n
the sequences shown in Figure 1-22.
2
  Solution:
j (nω̂0 +ϕ) − e−j (nω̂0 +ϕ)
n e
|A|r sin(nω̂0 + ϕ) = |A|r
n
2j (a) The pulse sequence can be described by g1 [n] =
u[n] − u[n − 3]. The first step sequence turns on the pulse
(1.45)
at n = 0 and the second step turns it off at n = 3.

(b) The periodic sequence g2 [n] can be described by first

We now look at Examples 1-3 and 1-4 of how to use the determining the sequence for one period, for example, for
sequences described previously as building blocks to write n = 0 to n = 5 (the period is N = 6). This already has
7 We have already considered the case where r = 1, which results in a been done in part (a), so all we need to do is create replicas
sustained oscillation. of u[n]−u[n−3] with each displaced by mN units, where
1.3 SIGNALS AND SEQUENCES 17

g1[n] can be described as (−n + a), and the value of a is

1 easily found by observing that, at n = 7, (−n + a)
has the value 0, which indicates that a = 7. So this
piece is described by (−n + 7)(u[n − 5] − u[n − 7]).
n
−2 −1 0 1 2 3 4 (v) For 7 ≤ n, the sequence is zero.
(a) A pulse sequence
Putting all of these pieces together, we have
g2[n]

1
g3 [n] = (n + 3)(u[n + 2] − u[n]) + 3(u[n] − u[n − 5])

... ... +(−n + 7)(u[n − 5] − u[n − 7]),

n
−6 −5 −4 −3 −2 −1 0 1 2 3 4 5 6 7 8
which can be simplified as
(b) A periodic pulse sequence
g3 [n] = (n + 3)u[n + 2] − nu[n] − (n − 4)u[n − 5]
g3[n]
+(n − 7)u[n − 7].
3
2 This also can be written as
2
1 1
g3 [n] = (n + 3)u[n + 3] − nu[n] − (n − 4)u[n − 4]
−3 −2 −1 0 1 2 3 4 5 6 7 n
+(n − 7)u[n − 7].
(c) A piecewise pulse sequence

This latter form illustrates another approach (explored in

Figure 1-22: Sequence for Example 1-3.
Problem 1.17) for writing expressions describing piecewise-
m is a positive or negative integer and N = 6. The result linear sequences.
is
∞
 In Example 1-4 we consider a left-sided step sequence and
g2 [n] = g1 [n − 6m] develop an m-file to plot the sequence. We also introduce
m=−∞ the use of stepfun, a built-in m-function that simplifies the
∞
 process.
= {u[n − 6m] − u[n − 6m − 3]},
m=−∞

Example 1-4: A Left-Sided Step Sequence

where g1 [n] = u[n] − u[n − 3].
(c) We can determine the overall sequence by partitioning the A left-sided unit step sequence g[n] is shown in Figure 1-23.
time axis into intervals where the sequence is linear: n ≤ This plot was generated by the segment of DTSFA file
−3; −2 ≤ n ≤ −1 (or −2 ≤ n ≤ 0); 0 ≤ n ≤ 4; 5 ≤ n ≤ F1_23_24 shown.
6; and 7 ≤ n.
(a) Write an analytical expression for g[n] using a left-sided
(i) For n ≤ −3, the sequence is identically zero. unit step sequence.
(ii) When −2 ≤ n ≤ −1, the sequence values are given
by (n + 3), so we need to turn on this sequence at (b) Write an analytical expression for g[n] using a right-sided
n = −2 and turn it off again at n = 0. This results unit step sequence.
in the subsequence (n + 3)(u[n + 2] − u[n]).
(c) Use the m-function stepfun to write an m-file to generate
(iii) In the interval 0 ≤ n ≤ 4, the sequence has the and plot the sequence
constant value 3 and can be represented by

3(u[n] − u[n − 5]). 2, n ≤ 3
(iv) For 5 ≤ n ≤ 6, the envelope of the sample values is g[n] =
0, 3 < n.
a straight line with a slope of −1. This envelope thus
18 CHAPTER 1 SIGNALS, SEQUENCES, AND SYSTEMS

C 3
…
2.5

2
… g[n] = 2u[−n + 3]
g[n] 1.5 = 2 − 2u[n − 4]

g[n]
1

0.5
…
0
…
−0.5

−1
n0 n −15 −10 −5 0 5 10 15
Sample number, n
Figure 1-23: A left-sided step sequence.
Figure 1-24: A left-sided step sequence using stepfun.
m-file
% F1-23: A left-sided step sequence (b) Using a right-sided step sequence, we can write
set(gcf,’DefaultLineLineWidth’,2);
% set default line width to 2 units g[n] = C − Cu[n − n0 − 1] = C(1 − u[n − n0 − 1]).
n=-15:1:15; % n=[-15 -14 ... 9 10]
gn=zeros(size(n)); % f=[0(at n=-15) 0...0 0(at n=10)] (c) The m-file segment that follows uses the m-function
gn(1:13)=2*ones(size(n(1:13))); stepfun and gives the plot in Figure 1-24. The plotting
% Makes first 13 entries of f=2.0, i.e., statements are omitted. Comparing the m-file segments
% gn=[2 2 ... 2 2 2 2 2 2(at n=-3) 0 0...0] used to plot Figures 1-23 and 1-24, we observe the
stem(n,gn); % plot advantages of using stepfun rather than setting the
axis off; individual elements of arrays.

%title(’Fig. 1-23: A Left-Sided Step Sequence’); m-file

% plot title(suppressed) nzp = 4; % turn-on value for right-sided seq.
gnp = 2*stepfun(n,nzp); % form right-sided seq. on at n=3
% text for plot gn = 2 - gnp; % flip the sequence to form left-sided seq.
text(-2,2,’C’); % plotting statements
text(-17.7,1.2,’g[n]’);
text(7,-0.1,’n’); Comment:
text(-3.2,-0.1,’n_0’); In most m-files from here on, plotting statements will be omitted
text(-17.7,1.9,’\ldots’,’fontsize’,24); as in part (c), but they can be found in the DTSFA files that are
% dots to show continuation available (see Preface).
text(15.4,0.4,’\ldots’,’fontsize’,24);
% dots to show continuation
pause; % pause execution to view the plots in window Figure 1 Next in Example 1-5, we consider in more generality how
to determine analytical expressions that characterize pulse
Solution: sequences.

(a) From the definition of the unit step sequence, u(−n − n0 )

Example 1-5: Writing Analytical Expressions for Pulse
is 1 whenever −n − n0 ≥ 0, and it is 0 otherwise. Since
−n − n0 ≥ 0 when n ≤ −n0 , we have Sequences

C, n ≤ −n0
g[n] = Cu[−n − n0 ] = (a) Write an analytical expression for the pulse sequence ga [n]
0, −n0 < n. shown in Figure 1-25.
1.4 CONVERSION OF CONTINUOUS-TIME SIGNALS AND DISCRETE-TIME SEQUENCES 19

2 1.4 Conversion of Continuous-Time

Signals and Discrete-Time Sequences
1.5

1 In Sections 1.1 and 1.2, it was indicated that discrete-time

processing has many advantages. Even when the input is a
ga[n]

0.5 continuous-time signal, it is often beneficial to convert it to

discrete-time form. Similarly, after processing a discrete-time
0 sequence, it is often desirable to convert the output sequence
back to a continuous-time signal. Conversion of a continuous-
time signal to a discrete-time format is accomplished by a
−0.5
sampling operation. Conversion from a discrete-time sequence
to a continuous-time signal is called reconstruction. In this
−1 section, we will consider how these operations are carried out.
−10 −8 −6 −4 −2 0 2 4 6 8 10
Sample number, n We begin by looking at the analog-to-digital conversion
process in a bit more detail. It is conceptually useful to be aware
Figure 1-25: A pulse sequence. of the operations that take place in an analog-to-digital converter
(ADC), even though we obtain an ADC as an electronic package
that accepts analog voltages (or currents) as inputs and produces
(b) Write an analytical expression for the pulse sequence gb [n] coded sequences as outputs. The conversion begins with a
that is 3.0 for 9 ≤ n ≤ 15 and is 0 otherwise. sample-and-hold operation that yields a staircase analog signal
xs (t). Each step in this staircase is represented by a sample
(c) Write an analytical expression for a sequence gc [n] that
value
is 2.0 for −3 ≤ n ≤ 3 and for 12 ≤ n ≤ 18 and is 0
otherwise.
Sampling
Solution:
xs (t)|t=nT = xs (nT ) n = . . . , −2, −1, 0, 1, 2, 3, . . .
(a) The pulse is to “turn on” at n = −3 and should “turn off’
(1.46)
at n = 4; thus, we have ga [n] = u[n + 3] − u[n − 4].

(b) This pulse is to “turn on” at n = 9 and should “turn off” at

n = 16, so the result is gb [n] = 3 · (u[n − 9] − u[n − 16]).
Notice also that gb [n] can be expressed as a scaled and
shifted version of ga [n]. That is,
gb [n] = 3ga [n − 12]
= 3(u[n − 12 + 3] − u[n − 12 − 4])
= 3(u[n − 9] − u[n − 16]).
(c) Here we can represent gc [n] as the sum of a scaled version
of ga [n] and a scaled and shifted version of ga [n]:
gc [n] = 2ga [n] + 2ga [n − 15]
= 2(u[n + 3] − u[n − 4]) + 2(u[n − 12] − u[n − 19]).
Comment:
The approach illustrated here of generating a sequence as a
summation of shifted and scaled sequences frequently can be
used to advantage.
where T represents the interval between samples obtained by
the device represented by the switch in Figure 1-26. The
sampling interval T and the sampling frequency fs are related
by
T = 1/fs . (1.47)
For simplicity, we represent xs (nT ) as simply xs [n] with the
sampling interval T understood and the square brackets indicate
that n is a discrete variable with integer values. The samples of
the continuous-time signal are then quantized (i.e., represented
by a sequence that has a discrete set of values). Typically, each
of these values will be represented by a coded binary word. For
example, each output from the ADC may be represented as a
2’s complement binary number.
Let’s look at the progression of an analog signal through an
analog-to-digital converter. Consider the analog signal xa (t)
shown in Figure 1-27. This signal is sampled at uniform
intervals of 0.04 s so T = 0.04 seconds/sample (s/smp). The
reciprocal of T is the sampling frequency, which is denoted by
fs and measured in samples/second (smp/s). In Figure 1-27, for
20 CHAPTER 1 SIGNALS, SEQUENCES, AND SYSTEMS

Analog-to-Digital Converter (ADC)

Sample and hold S/H

Analog Sampled-and-held Quantized Coded

signal T signal signal sequence

Hold Quantizer Encoder

xa(t) xs(t) xq(t) xc(t)

Figure 1-26: Analog-to-digital conversion.

6 be described by the relationship

xa(t), Input to sampler
5 xa(nT), Sample values ⎧
⎪
⎪3D, for 5D/2 ≤ xs (t),
4 xs(t), Output of hold ⎪
⎨mD, for D(m − 1/2) ≤ xs (t) < D(m + 1/2),
xq (t) =
3 ⎪
⎪ m = −3, −2, . . . , 1, 2
⎪
⎩
2 −4D, for xs (t) < −7D/2
(1.48)
1
where D is the quantization increment measured, for example,
0 in volts.
This three-bit quantizer is an example of a rounding
−1
quantizer, also referred to as a 21 -bit offset quantizer. Another
−2 type, considered in Problem 1.26, is a truncation quantizer.
In Figure 1-28, the ideal quantizer input–output characteristic
−3
0 0.08 0.16 0.24 0.32 0.4 0.48 is shown by the diagonal, dashed straight line passing
Time, seconds
Offset
Figure 1-27: Illustrating the sample-and-hold process. Level Binary
7 111
3D

2D 6 110
Quantizer output, xq(t)

D 5 101

example, we have fs = 1/T = 25 smp/s; the sample values are 0

4 100
shown as circles, and the held samples, which are denoted by
−D 3 011
xs (t), appear as the piecewise-constant function. The signal
xs (t) is sometimes referred to as a sampled-analog signal, 2 010
−2D
because it is defined for all times in the interval 0 ≤ t < 0.46 s
and is able to take on any amplitude values, not just a finite set −3D 1 001
of values. As shown in Figure 1-26, the next step is quantization 0 000
−4D
of the signal xs (t).
The input–output characteristic of a three-bit (binary digit) −9D/2 −5D/2 −D/2 D/2 5D/2
quantizer is shown in Figure 1-28. With three bits, the quantizer Quantizer input, xs(t)
is capable of representing 23 = 8 different output levels. These
eight levels have been numbered as 0 to 7 and are shown at the Figure 1-28: Input-output characteristic of a three-bit rounding
right side of the diagram. The input–output characteristic can quantizer.
1.4 CONVERSION OF CONTINUOUS-TIME SIGNALS AND DISCRETE-TIME SEQUENCES 21

through the origin. A quantizer having this straight-line 4

characteristic would be perfectly accurate, but to obtain this
ideal characteristic, we would need an infinite number of
bits, which is obviously impractical. However, we can lessen
quantization errors by increasing the number of quantizer bits,
Level
3 7
2 6
1 5

xs(t), xq(t) in volts

and this is one of the criteria used when selecting an ADC.
By increasing the number of bits, the quantizer “staircase” 0 4
function more closely approximates the ideal characteristic. −1 3
Another aspect of the quantizer to be noted is its saturation
−2 2
characteristic. For example, any input that exceeds 5D/2 will Output of hold, xs(t)
be quantized as 3D; similarly, any negative input less than −3 xs(nT) 1
−7D/2 will be quantized as −4D. Output of quantizer, xq(t)
−4 0
Assume that the increment D in Figure 1-28 is one volt xq(nT)
and that the units of xs (t) in Figure 1-27 are volts. In this −5
0 0.08 0.16 0.24 0.32 0.40 0.48
case, applying xs (t) to the quantizer yields the output shown
Time, seconds
for xq (t) shown in Figure 1-29(a). Notice that the sampled-
and-held signal xs (nT ) has a variety of values as shown in (a) The original analog signal, sampled-and-held
version, and quantized signal
Table 1-1, whereas the quantized signal xq (nT ) has values
that are integer multiples of one volt. Saturation occurs for 4
the sample occurring at t = 0.2 s, where xs (0.2) = 3.35 V is
3
quantized to an output voltage of 3.0 V. We also observe that the
sample and its quantized version in the interval 0.36 ≤ t < 0.40 2
are indistinguishable from one another on the plot. The
1
xq(nT) in volts

row labeled “Level” in Table 1-1 gives the values of xq (nT )

corresponding to the scale at the right side of Figure 1-28. 0
Converting these levels to binary values yields the row −1
labeled “Offset Binary” in Table 1-1. The encoder converts the
offset binary values to the 2’s complement representation shown −2
in the last row of Table 1-1. This encoding typically would be −3
made if the digital processing system for which xq (nT ) is the
input is using fixed-point binary arithmetic. Alternatively, the −4
Output sequence of ADC, xq[n]
output values could be represented as floating-point numbers −5
if the arithmetic is based on this format. Notice that the 2’s 0 1 2 3 4 5 6 7 8 9 10 11 n, samples
0.0 0.2 0.4 t, seconds
complement representation can be obtained from the offset
binary values by reversing the most significant bit value and (b) Representation of an output sequence from an ADC
retaining the remaining bits.
In addition to quantization errors introduced by the number of Figure 1-29: Signals at various stages of the analog-to-digital
conversion process.
bits available in the analog-to-digital converter, there are other
sources of error to be considered.8 In addition to the number
of bits that characterize an ADC, there are other specifications the discrete-time processing system, so we find it convenient to
that influence the selection of a particular model. Among these represent these sample values as shown in Figure 1-29(b). From
are the maximum (and minimum) speed of conversion, power here on, we will use a representation resembling Figure 1-29(b)
requirements, and various noise factors. to show sequences. Generally, we will not include the variable
Figure 1-29(a) shows the input analog signal to an ADC, T , and we will denote the sequence values xq (nT ) as simply
the sampled-and-held values, and their quantized representation xq [n] with the implicit presence of the sampling period T being
using a three-bit quantizer. The output of the ADC provides the understood.
input to a discrete-time system that latches the values received Often, a discrete-time system is used to process an analog
into a register. This process means that it is only the values of signal that has been converted to digital form by an ADC, and
the coded signal xc (t) at the sampling instants that are input to the processed sequence is then converted back to analog form,
8A detailed discussion of these issues is given in Gray, Nicholas, “The as shown in the system diagram of Figure 1-5. This would be the
ABCs of ADCs,” National Semiconductor Corporation, available at the website case, for example, if the input were an audio signal that is to be
www.national.com/appinfo/adc/files/ABCs_of_ADCs.pdf. filtered and is then used to drive speakers or headphones. This
22 CHAPTER 1 SIGNALS, SEQUENCES, AND SYSTEMS

Table 1-1: Sampled and Quantized Signals and Representations

n 0 1 2 3 4 5 6 7 8 9 10 11
xs (nT ). 1.45 −0.93 1.45 −0.61 −0.78 3.35 0.26 −1.73 1.61 0.02 0.27 2.76
xq (nT ) 1 −1 1 −1 −1 3 0 −2 2 0 0 3

Level 5 3 5 3 3 7 4 2 6 4 4 7
Offset Binary 101 011 101 011 011 111 100 010 110 000 000 111
2’s Complement 001 111 001 111 111 011 000 110 010 100 100 011

conversion from discrete-time (or digital) form to an analog
signal is accomplished by a digital-to-analog converter (DAC).
The input to a DAC is a sequence, and the output is a continuous-
time signal. One type of DAC converts each input sample value
to a pulse whose amplitude equals the value of the sample value.
The result of this process is shown in Figure 1-30. The sequence
values that appear at the output registers of the discrete-time
processing system are shown as blue circles. This sequence is
quantized to the same levels as shown earlier with the three-bit
ADC and also may be represented in a 2’s complement binary
representation, for example. The particular DAC represented
here provides a piecewise-constant output signal. This signal
is similar to what occurs in the sample-and-hold portion of an
ADC. This piecewise-constant output is provided by a zero-
order hold DAC; there also are other possibilities.9 The output
signal from this DAC is smoothed by a lowpass analog filter.
Figure 1-31 gives a representation of the process that begins
with an incoming analog signal that is converted to discrete-time
form, processed by a discrete-time system, and reconstructed to
analog form. The sequences shown in Figure 1-31 are quantized
of course. In the remainder of this text, however, we will ignore
quantization effects. This corresponds to the assumption that
we have ideal conversion of analog signals to discrete-time form
(as indicated by the dashed straight line in Figure 1-28) and ideal
reconstruction using a DAC as well. End-of-chapter references
provide additional details on quantization effects in discrete-
time systems.

1.5 The Sampling Theorem

Output signal from DAC, yd(t)
Input sequence to DAC, yc[n]
6
The sampling theorem, whose development is traced primarily
to C. E. Shannon, H. Nyquist, J. M. Whittaker, and D. Gabor,
is the fundamental result linking continuous-time and discrete-
time systems. A statement of the sampling theorem is given
here.

4 Sampling theorem

2 If an analog signal has no frequency components at

yc[n] and yc(t)

frequencies greater than fmax :

0
1. the signal can be uniquely represented by equally
−2 spaced samples if the sampling frequency fs is
greater than 2 · fmax , and
−4
2. the original analog signal can be reconstructed
−6 exactly from its samples.

0 0.08 0.16 0.24 0.32 0.40 time, s The minimum acceptable sampling frequency, 2 · fmax , is
0 1 2 3 4 5 6 7 8 9 10 n known as the Nyquist rate.

Figure 1-30: A DAC input sequence and stepwise constant output 9 See, for example, the discussion in Section 4-4 of Reference [1] in the
signal. bibliography at the end of this chapter.
1.5 THE SAMPLING THEOREM 23

Analog Sequence Processed Processed

signal Analog-to- Digital sequence Digital-to analog
Digial computer Analog signal
x(t) Converter x[n] or y[n] Converter y(t)
(ADC) DSP chip set (DAC)

x(t) x[n] y[n] y(t)

t n n t

Figure 1-31: A discrete-time processing system with analog input and output signals.

Although not a part of the sampling theorem as originally
stated, it turns out that reconstruction of the original analog
signal can be carried out by passing the samples through an
ideal lowpass analog filter having an appropriate bandwidth.
The importance of the sampling theorem is that we can
sample a frequency-band-limited, continuous-time waveform;
use a discrete-time system to process the samples; and then
convert the processed samples back to an analog signal without
losing information! Thus, the sampling theorem is what makes
it possible to obtain the benefits of discrete-time processing as
discussed in Sections 1.1 and 1.2 and shown in Figure 1-31.
Our goal in this section is demonstrate why the sampling
theorem works and what happens if its conditions are violated.
We will concentrate on the sampling of sinusoidal signals, but
the results can be generalized to other signals as well.
Comment:
Deriving the sampling theorem requires background in Fourier
transform theory. If you have previously studied continuous-
time Fourier transforms, however, Appendix C gives an
abbreviated development of the sampling theorem.
We begin by observing what happens when a continuous-
time sinusoidal signal is sampled. For example, consider
x(t) = A cos(2πf0 t + ϕ) = A cos(ω0 t + ϕ), (1.49)
where A is the amplitude, f0 is the frequency in hertz (or cycles
per second), t is time in seconds, ω0 is the angular frequency
in radians/second, and ϕ is the phase shift in radians. If this
signal is the input to an ideal analog-to-digital converter with a
sampling interval of T seconds/sample (a sampling frequency
fs = 1/T smp/s), the sampled output sequence from an ideal
ADC is
x(t)|t=nT = A cos(ω0 t + ϕ)|t=nT = x(nT )
(1.50)
= A cos(ω0 nT + ϕ),
which, by defining ω̂0 = ω0 T can be written as the (discrete-
time) sinusoidal sequence
x[n] = A cos(nω̂0 + ϕ). (1.51)
We have dropped the sampling period T from x(nT ) knowing
that this dependence is implicit. We define ω̂0 as the digital
radian frequency of the discrete-time sinusoid, and ω̂0 , ω0 , T ,
and fs have the following important relationships:
ω̂0 = ω0 T = ω0 /fs = 2πf0 T = 2πf0 /fs . (1.52)
Again, we note that the units of fs are samples/second, and
ω̂0 has units of radians/sample. Clearly, ω0 = 0 corresponds
to ω̂0 = 0. If ω0 ≈ 2πfs /2, which is the highest frequency
that can be represented by the specified sampling frequency,
this corresponds to the digital frequency ω̂0 ≈ π . Thus, the
useful range of digital frequencies for a discrete-time system
processing analog signals is
0 ≤ ω̂ < π, (1.53)
corresponding to analog frequencies in the range
0 ≤ f ≤ fmax < fs /2. (1.54)
It is useful at this point to consider some alternative
representations of sinusoidal sequences. Consider
x[n] = cos(nω0 T ) = cos(nω̂0 )
1 1 (1.55)
= ej nω̂0 + e−j nω̂0 ,
2 2
which is a cosine sequence with angular frequency ω̂0 . One
way to represent this sequence of two exponentials is as a
pair of rotating vectors, as shown in Figure 1-32 where we
have assumed that ω̂0 = π/4. Notice that one vector rotates
counterclockwise and the other clockwise. This representation
24 CHAPTER 1 SIGNALS, SEQUENCES, AND SYSTEMS

Im theorem. The different markers on the spectral lines at ±ω̂0 are

used to distinguish positive and negative frequency components
Radius = 1/2 n = 2,10,… associated with a sinusoidal sequence.
n = 3,11,… n = 1, 9,…
For Figures 1-32 and 1-33, we started with the sinusoidal
sequence of Eq. (1.55). Let us now consider what happens
␻
^0
when we obtain this sequence by sampling a continuous-
n = 4,12,… Re time sinusoid. Assume that we sample x(t) = 4 cos(2πf0 t) =
n = 0, 8,… 4 cos(2π · 50t), which is a 50-Hz sinusoidal signal. The
sampling theorem tells us that, to uniquely represent this
n = 5,13,… sinusoid by its samples, we must select a sampling frequency
n = 7,15,…
of greater than 100 smp/s. Assume that we select a sampling
n = 6,14,… frequency of fs = 200 Hz. This corresponds to a sampling
interval of

Im T = 1/fs = 1/200 = 0.005 s/smp, (1.56)

Radius = 1/2 and the sampled signal is given by the sequence

n = 6,14,…
n = 5,13,… n = 7,15,… x(t)t=nT = x[n]
(1.57)
= 4 cos(2π · 50 · n · 0.005) = 4 cos(nπ/2)
n = 4,12,… Re from which we see that ω̂0 = π/2. We already know from
n = 0, 8,… Section 1.3.2 [see Eq. (1.41)] that these same sample values
could have come from any one of an infinite number of
sinusoidal sequences given by
n = 3,11,… −␻
^0 n = 1, 9,…
n = 2,10,… r [n] = 4 cos(n[ω̂0 + 2π ]) (1.58)
where  is a positive or negative integer. Writing this expression
Figure 1-32: Rotating vector representation of a sinusoidal sequence in exponential form gives
with ω̂0 = π/4.
r [n] = 4 cos(n[ω̂0 + 2π ])
looks very similar to the one for a continuous-time sinusoid  
shown in Figure 1-11. An important difference, however, is that 1 j (n[ω̂0 +2π]) 1 −j (n[ω̂0 +2π])
=4 e + e (1.59)
the variable t in Figure 1-11 is a continuously varying quantity, 2 2
whereas n in Figure 1-32 can assume only integer values (e.g.,  = 0, ±1, ±2, . . . .
n = 0, 1, 2, 3, . . .).
This indicates that the sample values of x[n] could have come
An alternative representation of Eq. (1.55) is given in
from any of the family of sequences r [n]. If we represent some
Figure 1-33, where we show the frequency content of a
of these frequencies on a spectrum diagram with ω̂0 = π/2,
sinusoidal sequence. The two lines, one at ω̂ = ω̂0 and the
the result is shown in Figure 1-34. The lines at ±π/2 are
other at ω̂ = −ω̂0 , indicate that the sequence is made up of
those of the samples of the 50-Hz continuous-time sinusoidal
two frequency components with each having a magnitude of 21 .
signal. The other frequency lines are from aliases,10 which are
This plot is known as the magnitude spectrum of the sinusoidal
continuous-time sinusoids of other frequencies having the same
sequence x[n]. The type of plot as shown in Figure 1-33 will be
sample values as those of the 50-Hz sinusoid. Notice that the
very useful to us in considering the implications of the sampling
spectral lines at ω̂ = . . . , −9π/2, −5π/2, 3π/2, 7π/2, . . . are
Magnitude
displaced from the spectral line at ω̂ = −π/2 by 2π  radians
for  = . . . , −2, −1, 1, 2, . . .. The diamond-shaped end
1/2 1/2 of these lines indicates this relationship. Similarly, the
spectral lines at ω̂ = . . . , 9π/2, 5π/2, −3π/2, −7π/2, . . . are
displaced from the spectral line at ω̂ = π/2 by 2π  radians for
0  = . . . , 2, 1, −1, −2, . . ., as indicated by the lollipop-shaped
−␻ˆ 0 ˆ0
␻ ˆ , rad
␻
line ends. Also shown are analog frequencies corresponding to
ω̂ = ±π (i.e., ±fs /2) and ω̂ = ±ω̂0 (i.e., ±f0 ).
Figure 1-33: Magnitude spectrum of a sinusoidal sequence. 10 alias: an assumed or other name; also known as (aka).
1.5 THE SAMPLING THEOREM 25

Magnitude
x(t) x[n] xR(t)
C/D D/C
2

Figure 1-35: Ideal conversion of analog signal to discrete-time

−π 0 π sequence and back again.
−9π −7π −5π −3π −π π 3π 5π 7π 9π ␻ˆ , rad
2 2 2 2 2 2 2 2 2 2
We know that fs = 200 smp/s and solving for f0 gives f0 =
2 −2 1 −1 0 0 −1 1 −2 2  50 Hz. So the signal obtained by reconstructing this an analog
signal from its samples is
fs fs
− −f0 0 f0
2 2 xR (t) = 4 cos(2π · 50t). (1.65)

Figure 1-34: Frequency spectrum lines of a sequence and aliases.

An important issue then is how to reconstruct the continuous-
time sinusoidal signal from samples that could have been
obtained from many continuous-time sinusoids (an infinite
number, actually). The capability to reconstruct the original
continuous-time sinusoid is based on the condition specified
in Eq. (1.53), where we observed that the useful range of
frequencies for a discrete-time system is
0 ≤ ω̂ < π.11
So, for example, in Figure 1-34, we select only those spectrum
lines in the interval 0 ≤ ω̂ < π and pair them with the
corresponding negative frequencies, so we have those spectral
components enclosed by the dashed lines. From this selection
process, we obtain
xR [n] = 2ej nπ/2 + 2e−j nπ/2 = 4 cos(nπ/2),
where the subscript R indicates that this sequence is
reconstructed (or recovered) from the sequences whose spectral
lines are represented in Figure 1-34. To convert this sequence
back to the original analog signal that was sampled, we recall
that this signal was given by
x(t) = A cos(2πf0 t)
with t = nT = n/fs . To accomplish the reconstruction we
need to determine A and the analog frequency f0 .
Substituting t = n/fs in the expression for x(t) gives
x[n] = A cos(2πnf0 /fs )
and comparing the equations for xR [n] and x[n], we see that
A = 4 and
nπ/2 = 2πnf0 /fs . (1.64)
11 We actually can consider frequency components in the interval 0 ≤ ω̂ <
2π , as in Figure 1-34, but the frequency components in the range π ≤ ω̂ < 2π
are the same as the components in the range −π ≤ ω̂ < 0. This symmetry
exists for all positive and negative values of ω̂. See Figure 1-34. More on this
later.
Comment:
This may seem like an obvious result—and it is—but what
we are looking for is a process that will accept a sequence’s
sample values (given knowledge of the sampling frequency)
and reconstruct an analog signal. What we have done here can
be represented by the diagram of Figure 1-35. The blocks shown
in this diagram represent ideal continuous-time-to-discrete-
time conversion (C/D) and ideal discrete-time-to-continuous-
time conversion (D/C). Since these processes are assumed to
(1.60) be ideal, the reconstructed signal should be identical to the
input signal [i.e., xR (t) = x(t)]. Throughout the remainder of
this text, we assume the configuration shown in Figure 1-36
with ideal conversions to enable us to focus on the theory
and applications of discrete-time systems. If we were to look
inside the ideal conversion boxes, we would see something
like what is shown in Figure 1-37. The anti-aliasing ideal
(1.61) lowpass analog filter12 removes any input signal components
that are at frequencies above the maximum frequency assumed
for application of the sampling theorem. This ensures that
any unwanted signal components do not corrupt the ADC
conversion process. Similarly, the ideal lowpass analog
reconstruction filter shown in the D/C block removes high-
frequency components that accompany the output of a zero-
hold DAC, which is a device commonly used to convert digital
(1.62)
signals to a piecewise-constant output signal. It turns out that
oversampling, which is using a sampling frequency more than
the minimum required, eases the requirements on the lowpass
analog reconstruction filter so that something approaching an
ideal filter characteristic is not required. For example, analog
(1.63) audio signals that can be heard by the human ear generally do
not exceed 20 kHz, but the sampling frequency used for CD
recording is 44.1 k samples/second (smp/s).
Finally, let’s consider Example 1-6 where an input sinusoidal
signal is sampled at a frequency below the minimum rate
required by the sampling theorem.
12As we shall see subsequently, an ideal filter removes all frequency
components completely except those in the passband, and these passband
frequencies are all transmitted undistorted to the output of the filter.
26 CHAPTER 1 SIGNALS, SEQUENCES, AND SYSTEMS

x(t) x[n] Discrete- y[n] y(t)

C/D time D/C
system

Figure 1-36: Discrete-time system with ideal continuous-time/discrete-time conversions.

C/D From our previous investigation, we know that in addition

to the spectral lines at ±4π/3 there will also be aliases
Anti-aliasing (lines) at
x(t) ^
x(t) Ideal x[n]
filter
analog-to-digital
(ideal lowpass
converter ±(4π/3 +  · 2π ),  = ±1, ±2, . . . .
analog filter)
Evaluating this expression for a few values of  yields
spectral lines at
D/C

(π/3)×(. . . , −14, −10, −8, −4, −2, 2, 4, 8, 10, 14, . . .)

Reconstruction
y[n] Ideal ^
y(t) y(t)
filter
digital-to-analog as shown in Figure 1-38. Notice that the spectral lines
(ideal dowpass
converter
analog filter) at ω̂ = ±4π/3 lie outside of the interval −π ≤ ω̂ < π,
so the reconstruction selects the lines at ω̂ = ±2π/3 as
representing the sequence of samples from the analog
signal. Also notice that the lollipop- and diamond-shaped
Figure 1-37: Looking inside the ideal conversion processes.
tops are used to indicate from which spectral line at
ω̂ = ±4π/3 each of the alias lines came from.
Example 1-6: Consequences of Sampling Below the
Nyquist Rate (b) We know that the analog signal is to be reconstructed from
the sequence
A sinusoidal signal x(t) = cos(2π · 50t) is sampled at a rate
1 j 2πn/3 1 −j 2πn/3
of 75 smp/s. xR [n] = e + e = cos(2π n/3).
2 2
(a) Draw a spectrum diagram showing the frequencies of
The reconstructed analog signal has the form
sequences having the samples obtained by this process,
including the aliases.
xR (t) = A cos(2πf1 t),
(b) Determine the continuous-time signal obtained by
reconstructing the analog signal from the spectral lines where A and f1 are to be determined. The samples that
in the interval −π ≤ ω̂ < π and compare the results with correspond to xR (t) are given by
the original continuous-time sinusoid.
xR [n] = A cos(2π nf1 /fs )
A
Solution: = [(ej 2πnf1 /fs ) + (e−j 2πnf1 /fs )].
2
(a) The sampled sequence is given by x[n] =
cos(2π n · 50/75) = cos(4πn/3). Thus, for the Comparing the two representations of xR [n], we see that
exponential form of the cosine sequence, we have A = 1 and, with fs = 75 smp/s,

1 j n(4π/3) 1 −j n(4π/3) 2π n/3 = 2π nf1 /fs ⇒ f1 = fs /3 = 25 Hz.

x[n] = e + e .
2 2
1.6 A ROAD MAP 27

Magnitude

1/2

0 π
−14 −10 −8 −4 −2 −2 4 8 10 14 ␻
ˆ  3
−3 3

−3 1 −2 0 −1 −1 0 −2 1 −3 

fs fs
− 0 f
2 2

Figure 1-38: Spectral lines for 50 Hz sinusoid sampled at 75 smp/s.

So, even though the original analog signal was a 50 Hz

Table 1-2: Domains for Discrete-Time and Continuous-Time Systems
sinusoid, after sampling and reconstructing, it appears
as if we had a 25-Hz analog sinusoid. This erroneous Discrete-Time Systems Continuous-Time Systems
reconstruction is because we violated the requirement
of the sampling theorem that tells us that the sampling Time, n, samples (smp) Time, t, seconds (s)
frequency fs should have been more than 100 smp/s.
Additional aspects of the sampling theorem are the subject Frequency, ω̂, Frequency, f , Hertz (Hz) or
of the Exploration Problems at the end of the chapter. radians/sample (rad/smp) ω, radians/second (rad/sec)

Comment: Transform, z Transform, s

To help understand the concepts in this text and explore
extensions, several visualizations are provided. The first of
these, Sampler, is a LabVIEW Virtual Instrument (VI for short)
Table 1-3: Models for Discrete-Time and Continuous-Time Systems
on the DTSFA website (see Preface). This use of Sampler is
featured in Exploration Problems 1.37 through 1.41. Discrete-Time Systems Continuous-Time Systems
We again emphasize that, although we have used sinusoidal
sequences to illustrate important aspects of the sampling Difference equation, DE Differential equation, DE
theorem, it is important to keep in mind that this theorem applies
to all bandwidth-limited signals, not just sinusoids.
From this point onward, we will focus attention on the models Transfer function, H (z) Transfer function, H (s)
and solution methods for discrete-time systems. We begin by
taking a view from 30,000 feet of where we are going. Frequency response, Frequency response,
H (ej ω̂ ) H (j ω)
1.6 A Road Map
The study of signals and linear systems involves domains, State difference equation State differential equation
models, and methods. The rest of this book will focus on
learning to use the various models and techniques to analyze Unit impulse response, h[n] Unit impulse response, h(t)
discrete-time linear systems in the time [n], frequency (ω̂),
and transform (z) domains. Table 1-2 presents the domains Signal flowgraph or Signal flowgraph or
of interest. We shall see that system characteristics that are block diagram block diagram
hard to observe in one domain may be easily visible in another.
In addition, depending on the information available, it may be
more direct and efficient to perform system analysis in one Corresponding to the domains in Table 1-2 are the system
domain rather than another. models listed in Table 1-3.
28 CHAPTER 1 SIGNALS, SEQUENCES, AND SYSTEMS

State
equation Signal
flowgraph/
block diagram

Difference
equation
Impulse
response

Transfer
function

Frequency
response

Figure 1-39: Finding the frequency response from other system models.

It is often important to be able to convert a model from one

form to another. This concept is illustrated in Figure 1-39,
where it is shown that a conversion may be done directly from
Table 1-4: Operations for Linear Systems
one form to another (for example, from a difference equation
describing a discrete-time system to its frequency response) Discrete-Time Systems Continuous-Time Systems
or indirectly, as shown in the conversion from a difference
equation to its frequency response via an intermediate transfer
z transform Laplace (s) transform
function. In Figure 1-39, we also see that the path to a frequency
response from other models also can be either direct or indirect.
The model and domain used for a particular task depend on Convolution sum (time) Convolution integral (time)
various factors, including what is known about the system; what
information is desired; and what solution aids (for example, a Correlation sum (time) Correlation integral (time)
computer with LabVIEW software) are available.
Along with the models and domains are various operations Discrete-time Fourier, Continuous-time Fourier,
(given in Table 1-4) employed in the analysis and design of transform (DTFT), transform (CTFT),
linear systems. Notice that several of these operations may be Discrete Fourier Continuous-time Fourier
carried out in more than one domain—as the (very) old saying transform (DFT), series (CTFS)
goes, “there’s more than one way to....” Discrete Fourier
series (DFS)

Convolution Convolution
Final Comment: (by z transforms) (by Laplace transforms)
Tables 1-2 through 1-4 and Figure 1-39 are offered in the spirit
of telling you, the reader, where were going. Depending on the Correlation Correlation
state of your current knowledge, the information in these tables (by z transforms) (by Laplace transforms)
and the figure may mean a lot or very little. We recommend
that you periodically return to this section to gain perspective
as you work your way through the following chapters.
1.6 A ROAD MAP 29

Definitions, Techniques, and Connections

Continuous-Time Signals Discrete-Time Sequences


Unit δ (σ ) = lim g(, σ ) Unit 1, n − m = 0
impulse
→0 δ[n − m] =
impulse 0, n − m = 0
signal sequence
 
B, t0 ≤ t B, n0 ≤ n
Step signal B · u(t − t0 ) = Step B · u[n − n0 ] =
0, t < t0 0, n < n0
sequence

Exponential g(t) = Aeβt Exponential g[n] = A (a)n

signal sequence

Exponentially g(t) = |A|eαt cos(ω0 t + ϕ) Exponentially g[n] = |A| (r)n cos(nω̂0 + ϕ)

modulated modulated
sinusoidal sinusoidal
signal sequence
∞ ∞

Representation g(t) = −∞ g(τ )δ(t − τ )dτ
Representation g[n] = g(m)δ(n − m)
of any
of any m=−∞
signal
sequence

Sampling Theorem

If an analog signal has no frequency components at

frequencies greater than fmax :
1. the signal can be uniquely represented by equally
spaced samples if the sampling frequency fs is greater
than 2 · fmax , and
2. the original analog signal can be reconstructed exactly
from its samples

The minimum acceptable sampling frequency, 2 · fmax , is

known as the Nyquist rate.
30 CHAPTER 1 SIGNALS, SEQUENCES, AND SYSTEMS

m-functions used
Function Purpose and Use

plot Provides continuous-time or discrete-time plots.

stem Plots discrete-time functions (sequences).

stepfun Generates a shifted step function and/or a step sequence.

a(t) = Aeptu(t − t0) b(t) = [K − Aept]u(t)

Problems 8 3

Reinforcement Problems 7 2.5

6 2
1.1. Sketching step and pulse continuous-time signals.
5
Sketch the following signals. 1.5
4
1

a(t)

b(t)
(a) u(t − 4) 3
0.5
(b) −2u(t − 1) 2
0
(c) 1.3u(t + 6) 1
(d) u(t − 2) − u(t − 5) 0 −0.5
−1 −1
1.2. Sketching continuous-time signals. −1 0 1 2 3 4 −1 0 1 2 3 4
t in seconds t in seconds
Sketch the following signals.
(a) −2u(t + 1) + 3u(t) − u(t − 2) 2
(b) 2.5t[u(t) − u(t − 2)]
1.5
(c) −2.5t[u(t + 2) − u(t)] c(t) = Kept cos(␻t + ␾)u(t)
1
(d) −(t+4)u(t+4)+(t+2)u(t+2)+(t−2)u(t−2)−(t−4)u(t−4)
1.3. Signal synthesis. 0.5
c(t)

Write analytical expressions with numerical values of the 0

unknown constants for the signals shown in Figure 1-40.
−0.5
1.4. Periods of periodic signals.
−1
Find the period of each of the periodic signals shown in
Figure 1-41. −1.5

1.5. Signal synthesis. −2

0 0.2 0.4 0.6 0.8 1 1.2
Write analytical expressions for the signals shown in t in seconds
Figure 1-42.
Figure 1-40: Signals to be described analytically.
1.6. Describing periodic signals.
Write analytical expressions for the periodic signals shown in
Figure 1-43. 1.8. Modifying m-files.
1.7. Modifying m-files. Modify DTSFA file F1_12 so that the sinusoid has a period of
Use the m-function stepfun to modify DTSFA file F1_10 so T = 0.05 s, the phase shift is ϕ = 0, and use the m-function
that the exponentials are zero for t < 0. Adjust the axes as stepfun to make the signal zero for t < 0. Adjust the axes as
necessary to provide useful plots. necessary to provide useful plots.
PROBLEMS 31

8 8 g1(t)
7 7
6
… … 6
… … 2
5 5
4 4 t
0 2 4 6
3 3 (a)
2 2 g2(t)
1 1 5
0 0
−1
0 5 10 15 −1 0 5 10 15 20
t, seconds t, seconds t
(a) (b) 0 4
(b)
g3(t)
5 5
4… …
4
3 2
… … 2
3 t
1 0 4 5 10 11
2 0 (c)
−1 g4(t)
1
−2 t
0 −3 0 4 6 10
−4
−1 −5
−5 0 5 10 −10 0 10 20 30 40
t, seconds t, seconds −10
(c) (d) (d)

Figure 1-41: Periodic signals. Figure 1-42: Signals to be described analytically.

1.9. Modifying m-files.

(c) d3 (t) = 2e−1.8t cos(10t − π/4)u(t)
Modify DTSFA file F1_13 so that the sinusoids of both parts
(d) d4 (t) = e1.01t cos(8t + π/5)u(t)
have a period of T = 0.8 s, the phase shift, exponentials and
sinusoidal amplitudes are unchanged, and use the m-function 1.13. Sketching step and pulse sequences.
stepfun to make the signals zero for t < 0. Adjust the axes as
necessary to provide useful plots. Sketch the following sequences.

1.10. Generation of signal plots using m-files. (a) 3u[n − 4]

Write and execute m-files to generate and plot the signals of (b) −2u[n − 1]
Problem 1.1. (c) 1.3u[n + 6]
1.11. Generation of signal plots using m-files. (d) u[n − 2] − u[n − 5]
Write and execute m-files to generate and plot the signals of
1.14. Sketching sequences.
Problem 1.2.
Sketch each sequence.
1.12. Generation of signal plots using m-files.
Write and execute m-files to generate and plot the following (a) 4u[n − 3] − 2(n − 6)u[n − 6] + 2(n − 8)u[n − 8]
signals. (b) 4(u[n − 3] − u[n − 6]) − 2(n − 6)(u[n − 6] − u[n − 8])
(a) d1 (t) = 3e−2t u(t) (c) 2(n + 5)u[n + 5] − 3nu[n] + (n − 10)u[n − 10]
(b) d2 (t) = 2[1 − e−1.8t ]u(t) (d) 2(n + 5)(u[n + 5] − u[n − 10]) − 3n(u[n] − u[n − 10])
32 CHAPTER 1 SIGNALS, SEQUENCES, AND SYSTEMS

g1p(t) 6
4
2

x1p[n]
1 0
−2
... ...
t −4
−4 −2 0 2 4 6 8 10 −6 −4 −2 0 2 4 6 8
Samples, n
(a) (a)
g2p(t) 2
2 1

x2p[n]
0
−1
... ... −2
t −6 −4 −2 0 2 4 6 8
−4 −2 0 2 4 6 8 10 Samples, n
(b)
(b)
g3p(t) 4
2
1

x3p[n]
0
t −2
−2 0 2 4 6 8 −4
−6 −4 −2 0 2 4 6 8 10
−1 Samples, n
(c) (c)
3
g4p(t) 2
1
x4p[n]

1 0
−1
... ... −2
t
−5 −3 −1 01 3 5 7 9 11 13 −3
−4 −2 0 2 4 6
(d) Samples, n
(d)
Figure 1-43: Signals to be described analytically.
Figure 1-44: Periodic sequences.
1.15. Periods of periodic sequences.
Write analytical expressions for the sequences shown in
Determine the period N of each of the periodic sequences shown Figure 1-47.
in Figure 1-44.
1.19. Describing periodic sequences.
1.16. Periods of periodic sequences. Write analytical expressions to describe the periodic sequences
Determine the period N of each of the periodic sequences shown fp (n) and gp (n) shown in Figure 1-48.
in Figure 1-45.
1.20. Modifying m-files.
1.17. Sequence synthesis. Modify DTSFA file F1_16 by using the m-function stepfun to
generate the two step sequences.
(a) Describe the sequence shown in Figure 1-46 as the sum
of subsequences consisting of either of the following. 1.21. Modifying m-files.
(i) Ramp sequences "turned on" at n = 0, 2, 6, and 8, Modify DTSFA file F1_21 by using the m-function stepfun to
respectively. generate pulsed, exponentially modulated sinusoidal sequences
(ii) Two triangular sequences and a rectangular pulse whose values are zero except in the interval 0 ≤ n ≤ 20. Adjust
sequence. the axes as necessary to provide useful plots.

(b) Show that the two forms obtained in part (a) are equivalent. 1.22. m-file generation of sequences.
Write and run m-files to generate and plot the sequences of
1.18. Sequence synthesis. Problem 1.13.
PROBLEMS 33

5 e[n]
4 3
3
x1p[n]

2
1
0 ...
−1
−4 −2 0 2 4 6 8 −2 −1 0 1 2 3 4 5 6 7 8 n
Samples, n
(a)
4 f [n]
3
2
x2p[n]

0
−2 ...
−5 0 5 10 −8 −7 −6 −5 −4 −3 −2 −1 0 n
Samples, n
(b) g[n]
2
1 8
x3p[n]

0
−1
−2
−4 −2 0 2 4 6 8 . 2n 4 4 8 . 2−n
Samples, n
(c)
6 2 2
4 1 1
... ...
x4p[n]

2 −4 −3 −2 −1 0 1 2 3 4 n
0
−4 −2 0 2 4 6 8 Figure 1-47: More analytical description of sequences.
Samples, n
(d)
fp[n]
Figure 1-45: More periodic sequences.
2 2 2 2
g[n] 1 1 1 1 1 1 1
... ...

−2 0 2 5 n
2
1
... gp[n]
0 1 2 6 8 n
3

Figure 1-46: Sequence to be described analytically.

... ...
−12 0 3 7 12 24 n
1.23. m-file generation of pulse sequences.
Write and run m-files to generate and plot the sequences of Figure 1-48: Analytical description of periodic sequences.
Problem 1.14.

1.24. Analog-to-digital conversion.

(a) Using the rounding ADC input–output characteristic
in Figure 1-28 as a model, sketch the input–output
characteristic of a four-bit rounding quantizer whose input
values are as in Figure 1-28 and whose output values are
in the range −4D to 3.5D, where D/2 is the quantization
increment measured in volts, for example.
(b) On the sketch of part (a), show the input–output
characteristic of an ideal four-bit quantizer and the levels
from 0 to 15.
34 CHAPTER 1 SIGNALS, SEQUENCES, AND SYSTEMS

1.27. Sampling frequency.

5
4 (a) The sampling frequency for an unknown analog signal
3 is fs = 1000 smp/s. What is the highest frequency
2 component that can be present in the analog signal if it
1 can be reconstructed exactly from its samples?
xs(t)

0 (b) Repeat part (a) for fs = 250 smp/s.

−1
(c) Repeat part (a) for fs = 25k smp/s.
−2
−3 (d) Repeat part (a) for fs = 2M smp/s.
−4
1.28. Sampling frequency.
−5
The following signals are to be sampled at frequencies that
−1 0 1 2 3 4 5 guarantee satisfying the requirements of the sampling theorem.
t, seconds  10−3 In each case, determine the minimum acceptable sampling
frequency.
Figure 1-49: Input to a quantizer.
(a) 5 cos(2π · 150t + π/3)
(c) If quantization error is defined as quantizer input minus (b) 2 sin(500π t − π/4)
quantizer output, sketch the quantization error as a (c) 3 cos(20π t) · cos(500π t)
function of quantizer input voltage.
(d) 7 sin(500π t)[1 + 2 cos(30π t)]
1.25. Analog-to-digital conversion. 1.29. Aliasing.
Consider the three-bit rounding quantizer of Figure 1-28 with A sinusoidal sequence is given by x[n] = 5 cos(0.3π n + ϕ).
D = 1 V.
(a) Find equations for two other sequences having the same
(a) The input to the quantizer is the signal shown in sample values as x[n].
Figure 1-49. Sketch the output xq (t).
(b) An analog signal is given by y(t) = 5 cos(2πf0 t + ϕ)
(b) Sketch the quantization error e(t) = xs (t) − xq (t) for the where f0 = 300 Hz. Find three sampling frequencies that
input of part (a). yield sequences having the same sample values as x[n].
(c) Comment on the errors you obtained in part (b).
1.30. Sampling and aliasing.

1.26. Analog-to-digital conversion.
This problem uses the notation of Figure 1-28 and Eq. (1.48).
A three-bit quantizer of the truncation type is described by
⎧ Explain.
⎪
⎨3D, 3D < xs (t)
xq (t) = mD, mD ≤ xs (t) < (m+1), m = −3, −2, . . . , 2
⎪
⎩
−4D, xs (t) ≤ −3D.
(a) Sketch the input–output characteristic of this quantizer.
(b) Defining the quantization error as e(t) = xs (t) − xq (t),
sketch the error as a function of the input xs (t).
(c) Repeat part (b) using the three-bit rounding quantizer of
Figure 1-28.
(d) Which of the quantizers, the rounding quantizer of
Figure 1-28 or the truncating quantizer of part (a) would
you prefer to use? Explain.
The sinusoidal signal x(t) = 4 sin(2πf0 t) with f0 = 100 Hz is
sampled with fs = 400 smp/s to obtain the sequence x[n].
(a) Are the conditions of the sampling theorem satisfied?
(b) Find three sequences all having the same sample values
as x[n].
(c) Sketch the magnitude frequency spectrum, including
aliases, of the sequence x[n].
1.31. Frequency spectra.
Let fsmin denote the minimum sampling frequency to satisfy the
sampling theorem. For each of the signals of Problem 1.28, use
a sampling frequency fs = 2 × fsmin .
(a) Determine the resulting sequences
(b) Sketch the frequency spectra magnitudes in the interval
−π ≤ ω̂ < π .
PROBLEMS
1.32. Frequency spectra and sampling frequency.
Let fsmin denote the minimum sampling frequency to satisfy the
sampling theorem.
(a) For the signal of part (a) of Problem 1.28, sketch
the frequency spectrum magnitudes for the sequences
obtained when
(i) fs = 2fsmin
(ii) fs = 3fsmin
(iii) fs = 4fsmin
(b) On the sketches from part (a), show the aliases closest to
the range −π ≤ ω̂ < π .
(c) What do you notice about the aliases shown in part (b)?
1.33. Frequency spectra and sampling frequency.
Repeat Problem 1.32 for the signal of Problem 1.28, part (b).
Exploration Problems
1.34. Frequency spectra and sampling frequency.
Repeat Problem 1.32 for the signal of Problem 1.28, part (c).
1.35. Frequency spectra and sampling frequency.
Repeat Problem 1.32 for the signal of Problem 1.28, part (d).
1.36. Aliases in discrete-time and continuous-time domains
In Eq. (1.41), we found that the sinusoidal sequences having
the same values are related to one another by
cos(n[ω̂0 + 2π]) = cos(nω̂0 )
where n and  are integers and  = ±1, ±2, ±3, . . .. We also
know from Eq. (1.52) that
ω̂0 = ω0 T = ω0 /fs = 2πf0 T = 2πf0 /fs .
We now ask ourselves what analog sinusoidal frequencies when
sampled with a sampling frequency fs will give the samples
cos(n[ω̂0 + 2π ])? These frequencies are the aliases of f0 .
Clearly, there will be several such sinusoids, so we denote their
analog frequencies by f , where each frequency corresponds to
a value of the integer .
(a) Show that the relationship between f0 and f is f =
±(f0 + fs ),  = ±1, ±2, ±3, . . ..
(b) Write the expressions for f in terms of f0 and fs for
 = ±1, ±2, and ±3.
35
(c) On a magnitude spectrum sketch, show the frequency
components that are aliases of one another assuming that
fs = 200 smp/s and f0 = 50 Hz.
(d) Compare the results of part (c) with those of Figure 1-34.
Visualization Problems
In each of the following four problems, we assume the
configuration in Figure 1-36. That is, we assume ideal
continuous-time to discrete-time conversion followed by
ideal discrete-time to continuous-time conversion. It is
recommended that all four problems be solved.
1.37. Reconstruction of analog signals.
Equation (1.41), cos(n[ω̂0 + 2π ]) = cos(nω̂0 ) with n and 
integers, demonstrates that there are many sequences having the
same sample values as the sequence cos(nω̂0 ). In Problem 1.36,
it was shown that the sinusoidal analog frequencies f that yield
the sample values cos(n[ω̂0 +2π ]) are given by f = f0 +fs ,
 = ±1, ±2, ±3, . . .. The quantity f0 is the analog frequency
of a sinusoidal signal that when sampled with sampling
frequency fs gives the sequence cos(nω̂0 ). We also know that
the reconstruction process described in Section 1.5 selects the
digital frequencies in the interval 0 ≤ ω̂0 < π and that these
frequencies correspond to the analog frequencies in the range
0 ≤ f0 < fs /2. Thus, for reconstruction purposes, we are
interested in the values of f for  having the negative values,
because clearly f1 = f0 + fs , f2 = f0 + 2fs , . . . all lie outside
the reconstruction interval corresponding to 0 ≤ f0 < fs /2.
(a) Evaluate |f0 + fs | for  = 0, −1, −2, −3. The sampling
frequency is fs = 30 smp/s and the analog sinusoidal
input signal to be sampled has frequency f0 that varies
from 0 Hz to 90 Hz in steps of 3 Hz. Use the values
obtained to complete Table 1-5.
Table 1-5:
f0 , Hz |f0 | |f0 − fs | |f0 − 2fs | |f0 − 3fs |
0
3
..
.
90
It is recommended that you write a spreadsheet or an
m-file to generate the entries for the table.
36 CHAPTER 1 SIGNALS, SEQUENCES, AND SYSTEMS

(a) fs = 100 smp/s

Table 1-6:
(b) fs = 80 smp/s
f0 |f0 − fs | |f0 − 2fs | |f0 − 3fs | (c) fs = 25 smp/s
(d) fs = 15 smp/s
0. · fs
(e) fs = 10 smp/s
0.1 · fs ( f ) fs = 7 smp/s.

.. Use the Sampler VI to determine the reconstructed output

. frequencies in each case.

30 · fs 1.40. Alias exploration.

(b) Explain why you were asked to use absolute values for
the four rightmost columns.
(c) From the table obtained in part (a), determine the
reconstructed (or apparent) analog frequencies that would
be determined for each value of f0 .
(d) Use the Sampler VI to verify the results of part (c) for the
range 0.0 ≤ f0 ≤ 25 Hz.
(e) Plot the reconstructed frequency values from part (c) as a
function of the input frequency f0 .
( f ) Repeat parts (a), (c), and (e) using literal quantities, (i.e.,
using f0 for the analog sinusoidal input frequency and fs
for the sampling frequency). Complete Table 1-6.
Hint: All of the table entries will be expressed in terms of
fs .
1.38. More reconstruction of analog signals.
An analog sinusoidal input signal of frequency 20 Hz is sampled
at the following sampling frequencies:
(a) fs = 100 smp/s
(b) fs = 60 smp/s
(c) fs = 42 smp/s
(d) fs = 30 smp/s
(e) fs = 15 smp/s.
Use the Sampler VI to determine the frequencies of the
reconstructed output signal and plot it as a function of fs /f0 .
1.39. Still more reconstruction of analog signals.
The analog signal
x(t) = cos(2πf1 t) + 0.5 cos(2πf2 t) + 0.2 cos(2πf3 t),
where f1 = 4 Hz, f2 = 7 Hz, and f3 = 12 Hz is sampled at the
following frequencies:
The input signal to an ideal continuous-time-to-discrete-time
conversion device is
x(t) = sin(20π t + π/12),
and y(t) is the reconstructed output of an ideal discrete-time-
to-continuous-time conversion device.
(a) Predict and tabulate the frequencies that will appear in
y(t) as the sampling frequency, fs , is decreased from its
starting value of 30.0 smp/s in steps of 1.0 smp/s to a final
value of 1.0 smp/s.
(b) Run the Sampler VI using the given input signal and the
sampling specifications from part (a).
(c) Compare your results from parts (a) and (b) and resolve
any differences you find.
1.41. Design your own.
In learning new problem-solving techniques, it is often helpful
to formulate a hypothesis, design an experiment to test the
hypothesis, conduct the experiment, evaluate the experimental
results, and determine if the results support the hypothesis.13 If
the results do not support the hypothesis, an iterative approach is
used in which the hypothesis and/or the experiment is modified
and the sequence of steps is repeated. This is an example of
the scientific method. The kind of "experiments" in this context
exemplify what is called a thought experiment (in the German
language gedankenexperiment), because we are not actually in
a laboratory with equipment and measuring devices but instead
use numerical techniques with the aid of VIs to investigate.
Develop a thought experiment that can be investigated using
the Sampler VI. Decide on a question you want to explore,
formulate a hypothesis, design and conduct the experiment with
the VI, and state your conclusions. If necessary, modify your
hypothesis and the experiment and repeat the process.
13 This approach is also useful when learning to use a new software statement
or package.
BIBLIOGRAPHY 37

Bibliography [5] Peled, Abraham, Bede Liu, Digital Signal Processing,

[1] McClellan, James H., Ronald W. Schafer, and Mark A.
Yoder, Signal Processing First, 2nd ed., Pearson Prentice
Hall, Upper Saddle River, N.J., 2003. Chapter 4 is devoted
to sampling and aliasing and gives a treatment based on
first principles that provides insight on these topics.
[2] Porat, Boaz, A Course in Digital Signal Processing, John
Wiley & Sons, Inc., New York, 1997. Chapter 1 provides
a good discussion of the advantages and disadvantages
of discrete-time systems along with an overview of
application areas.
[3] Gabel, Robert A. and Richard A. Roberts, Signals and
Linear Systems, 3rd ed., John Wiley and Sons, New York,
1987. Chapter 1 includes a discussion of classification of
systems, linearity, and models of discrete-time systems.
Theory, Design and Implementation, John Wiley & Sons,
New York, 1976. Chapter 1 includes a discussion of the
basic principles and characteristics of analog-to-digital
(A/D) and digital-to-analog (D/A) conversions. Chapter 4
is devoted to the hardware implementation of digital signal
processors. The first two sections of this chapter review the
basics of binary arithmetic and digital hardware.
[6] Soliman, Samir S., and Mandyam D. Srinath, Continuous
and Discrete Signals and Systems, Prentice-Hall, Inc.,
Englewood Cliffs, N.J., 1990. Discrete-time sequences are
described in Chapter 6.

[4] Oppenheim, Alan V. and Ronald W. Schafer, Discrete-

Time Signal Processing, 3rd ed., Prentice Hall, Inc.,
Englewood Cliffs, N.J., 2010. This book is a new version
of the landmark work Digital Signal Processing that
was published by the same authors in 1975. Although
Discrete-Time Signal Processing is used primarily
in senior/graduate-level courses, the discussion on
quantization illuminates some practical aspects of digital
processing of signals that are of interest when studying
linear systems at an introductory level.