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Digital Signal Processing
A Primer With MATLAB®
Digital Signal Processing
A Primer With MATLAB®
Samir I. Abood
MATLAB® is a trademark of The MathWorks, Inc. and is used with permission. The MathWorks does not warrant the accuracy
of the text or exercises in this book. This book’s use or discussion of MATLAB® software or related products does not constitute
endorsement or sponsorship by The MathWorks of a particular pedagogical approach or particular use of the MATLAB® software.
CRC Press
Taylor & Francis Group
6000 Broken Sound Parkway NW, Suite 300
Boca Raton, FL 33487-2742
This book contains information obtained from authentic and highly regarded sources. Reasonable efforts have been made to pub-
lish reliable data and information, but the author and publisher cannot assume responsibility for the validity of all materials or the
consequences of their use. The authors and publishers have attempted to trace the copyright holders of all material reproduced in
this publication and apologize to copyright holders if permission to publish in this form has not been obtained. If any copyright
material has not been acknowledged please write and let us know so we may rectify in any future reprint.
Except as permitted under U.S. Copyright Law, no part of this book may be reprinted, reproduced, transmitted, or utilized in any
form by any electronic, mechanical, or other means, now known or hereafter invented, including photocopying, microfilming, and
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For permission to photocopy or use material electronically from this work, please access www.copyright.com (http://www.copy-
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Trademark Notice: Product or corporate names may be trademarks or registered trademarks, and are used only for identification
and explanation without intent to infringe.
2.4.1.3 Homogeneity.......................................................................26
2.4.1.4 Stability...............................................................................26
2.5 Linear Time-Invariant Causal Systems (LTI) ................................................. 27
2.5.1 Linearity ............................................................................................. 27
2.5.2 Time-Invariance ................................................................................. 31
2.5.3 Causality............................................................................................. 32
2.6 Definitions ....................................................................................................... 36
2.6.1 Continuous-Time System ................................................................... 36
2.6.2 Discrete-Time System ........................................................................ 37
2.6.2.1 Delay Operator.................................................................... 38
2.6.2.2 Convolution Property.......................................................... 38
2.6.2.3 Impulse Function ................................................................ 38
2.6.2.4 Impulse Response ............................................................... 38
2.6.2.5 Frequency Response ........................................................... 38
2.7 System Output ................................................................................................. 39
2.7.1 Causality............................................................................................. 39
2.7.2 Stability .............................................................................................. 39
2.7.3 Invertibility.........................................................................................40
2.7.4 Memory ..............................................................................................40
Problems..................................................................................................................... 42
xiii
xiv Preface
In Chapter 8, the principles of the Fast Fourier Transform, decimation-in-frequency method, and
decimation-in-time method are examined. Chapter 9 discusses the z-transform, region of conver-
gence (roc), properties of the z-transform, and it also discusses the inverse of z-transform. Chapter
10 describes the z-transform applications in DSP for evaluating LTI system responses using z-trans-
form and implementation of the system using z-transform. Chapter 11 introduces pole-zero stability,
difference equations and transfer function, and the stability of DSP systems.
Chapter 12 discusses sampling relating the FT to the DTFT for discrete-time signals, the sam-
pling of continuous-time signals, and instantaneous sampling. The description of digital filters and
filter types and specifications is discussed in Chapter 13. Chapter 14 presents the implementation of
IIR digital filters and their properties and the design of a notch filter by MATLAB.
Chapter 15 deals with the implementation of Finite Impulse Response (FIR), and it is design.
Chapter 16 deals with the digital filter design, the realization of digital filters, and direction-form
I realization.
Earlier experience using the MATLAB program is not needed since the author highly recom-
mends that the reader studies this material in conjunction with the MATLAB Student Version.
Chapter 1 and Appendix C of this text provides a practical introduction to MATLAB.
MATLAB® is a registered trademark of The MathWorks, Inc. For product information, please
contact:
xv
Author
Samir I. Abood received his BSc and MSc from the University of Technology, Baghdad, Iraq, in
1996 and 2001, respectively. From 1997 to 2001, he worked as an engineer at the same university.
From 2001 to 2003, he was an assistant professor at the University of Baghdad and AL-Nahrain
University, and from 2003 to 2016, he was an assistant professor at Middle Technical University
University, Baghdad, Iraq. Presently, he is doing his PhD in the Electrical and Computer Engineering
Department at Prairie View A&M University, Prairie View, Texas. He is the author of 25 papers and
four books. His main research interests are in the area of sustainable power and energy systems,
microgrids, power electronics, and motor drives, the application of digital PID controllers, digital
methods to electrical measurements, digital signal processing, and control systems.
xvii
1 Continuous and
Discrete Signals
Mathematically, signals are represented as a function of one or more independent variables. At
this point, we are focusing the attention on signals that involve a single independent variable.
Conventionally, it will generally refer to the independent variable as time. There are two types of
signals: continuous-time signals and discrete-time signals. In this chapter will focus on the kinds
of signals
This chapter gives you a quick way to become familiar with the MATLAB software by introduc-
ing you the basic features, commands, and functions. You will discover that entering and solving
complex numbers in MATLAB is as easy as entering and solving real numbers, especially with the
help of MATLAB built-in complex functions. Upon completion this chapter, and Appendix A you
should know how to start MATLAB, how to get HELP, how to assign variables in MATLAB and
to perform the typical complex numbers operations (i.e., complex conjugate, addition, subtraction,
multiplication, division, expression simplification) and the conversions of complex numbers in both
rectangular and polar forms with and without using MATLAB built-in functions.
1
2 Digital Signal Processing
L=length(t);
for i=1:L
if t(i)<0
x1(i)=0;
x2(i)=0;
else
x1(i)=1;
x2(i)=t(i);
end;
end;
figure;
plot(t,x1);
xlabel('t');
ylabel('amplitude');
title('unit step');
grid
unit step
0.8
0.6
amplitude
0.4
0.2
x2=sin(2*pi*f2*t);
figure;
subplot(2,1,1);
plot(t,x1,’b’,t,x2,’r’);
xlabel(‘t’);
ylabel(‘amplitude’);
title(‘The signals x1(t) and x2(t)’);
x3=x1+x2;
subplot(2,1,2);
plot(t,x3);
xlabel(‘t’);
ylabel(‘amplitude’);
title(‘The sum of x1(t) and x2(t)’);
Figure 1.2 Shows the results obtained from the sum of the two functions.
0
–1
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
t
0
–2
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
t
0
–1
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
t
0
–1
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
t
Figure 1.3 shows the results obtained from multiplying the two functions.
–1
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
t
–1
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
t
x3=[zeros(1,100),x2(1☹L-100))];
subplot(2,1,2);
plot(t,x3);
title(‘the shifting of x1(t)and x2(t)’);
xlabel(‘t’);
ylabel(‘amplitude’);
Figure 1.4 shows the results obtained from shifting the function.
Discrete-time signals are often derived from sampling a continuous-time signal, such as speech,
with an analog-to-digital (A/D) converter. For example, a continuous-time signal x(n) that is sam-
pled at a rate of fs = l/Ts samples per second produces the sampled signal x(n), which is related to
xa(t) as follows:
x(n)
–2 –1 0 1 2 3 4 5 6 n
The magnitude may be derived from the real and imaginary parts as follows:
Im {z(n)}
arg {z(n)} = tan −1 (1.6)
Re {z(n)}
If z(n) is a complex sequence, the complex conjugate, denoted by z*(n), is formed by changing the
sign on the imaginary part of z(n):
T /2
1
P = lim
T →∞ T ∫ s(t )s (t )dt
−T /2
*
0<P<∞ (1.8)
∞ ∞
∫ s(t ) s (t ) dt = ∫ s(t )
2
E= *
dt 0<E<∞ (1.9)
−∞ −∞
In communication systems, the received waveform is usually categorized into the desired part that
contains the information signal and the undesired part, which is called noise.
(Continued)
10 Digital Signal Processing
(Continued)
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— Mene Hel…!
———
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jo tiedänkin.
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