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B. Preetham Kumar
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® soft-
ware 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
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Visit the Taylor & Francis Web site at
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and
In memory of my parents
Contents
Preface.................................................................................................................... xiii
Note to Readers on Structure of Book and Exercises.................................... xvii
Author.................................................................................................................... xix
vii
viii Contents
The motivating factor in the preparation of this book was to develop a practical
and readily understandable laboratory volume in Digital Signal Processing
(DSP). The intended audience is primarily undergraduate and graduate
students taking DSP for the first time as an elective course. The book is very
relevant at the present time, when software and hardware developments in
DSP are very rapid, and it is vital for the students to complement theory with
practical software and hardware applications in their curriculum.
This book essentially evolved from the study material in two courses
taught at the Department of Electrical and Electronic Engineering,
California State University, Sacramento (CSUS). These courses, Introduction
to Digital Signal Processing, and Digital Signal Processing Laboratory, have
been offered at CSUS for the past several years. During these years of DSP
theory and laboratory instruction for senior undergraduate and graduate
students, often with varied subject backgrounds, we gained a great deal of
experience and insight. Students who took these courses gave very useful
feedback, such as their interest for an integrated approach to DSP teaching
that would be comprised of side-by-side training in both theory and practi-
cal software/hardware aspects of DSP. In their opinion, the practical com-
ponent of the DSP course curriculum greatly enhances the understanding
of the basic theory and principles.
The factors above guided us to prepare each chapter of this book to include
the following components: a brief theory to explain the underlying math-
ematics and principles, a problem solving section with a reasonable num-
ber of problems to be worked by the student, a computer laboratory with
programming examples and exercises in MATLAB® and Simulink®,* and
finally, in applicable chapters, a hardware laboratory with exercises using
test and measurement equipment, and the Texas Instruments TMS320C6711
DSP Starter Kit.
In Chapter 1, we go into a brief theory of DSP applications and systems,
with solved and unsolved examples, followed by a computer lab, which
introduces the students to basic programming in MATLAB, and creation of
system models in Simulink. This chapter concludes with a hardware section,
which contains instructions and exercises on usage of basic signal sources,
such as synthesized sweep generators, and measuring equipment, such as
oscilloscopes and spectrum analyzers.
Chapter 2 is a more detailed description of LTI discrete-time signals and
systems, and the mathematical tools used to describe these systems. Basic
concepts such as z-transform, system function, discrete-time convolution,
xiii
xiv Preface
and difference equations are reviewed in the theory section. Practical types
of LTI systems, such as inverse systems and minimum phase systems are also
discussed, with example problems. This is followed by a computer lab,
which has guidance and exercises in the creation and simulation of LTI
system models.
Chapter 3 covers the practical time and frequency analysis of discrete-time
signals, with emphasis on the evolution of the Discrete Fourier Transform
(DFT) and the Fast Fourier Transform (FFT). The software lab includes spec-
tral analysis, using the FFT, of practical periodic and nonperiodic signals
such as noisy signal generators and Amplitude Modulation (AM) systems.
The hardware lab involves actual measurement of harmonic distortion in
signal generators, spectrum of AM signals, and the comparison of measured
results with simulation from the computer lab section.
Chapter 4 is a practical discussion of the Analog-to Digital (A/D) process, with
an initial brief review of sampling, quantization (uniform and nonuniform),
and binary encoding in the Pulse Code Modulation (PCM) process. The soft-
ware lab includes MATLAB/Simulink A/D process simulation of practical audio
signals, and advanced systems such as Differential PCM. The hardware lab gives
guidance of the construction and testing of a FET Sample and Hold circuit.
Chapters 5 and 6 are devoted to design and application of digital filters.
Chapter 5 reviews the basic concepts of digital filters, and analytical design
techniques for Finite Impulse Response (FIR) and Infinite Impulse Response
(IIR) digital filter design. The second edition also includes a section on FIR
and IIR digital filter structures. The computer lab details Computer Aided
Design (CAD) techniques for FIR and IIR digital filters, and has a series of
rigorous exercises in usage of these techniques. Chapter 6 deals with the
application of digital filters to one-dimensional (audio) and two-dimensional
(video) signals. The computer lab has a set of practical exercises in the appli-
cation of one- and two-dimensional digital filters for practical purposes,
such as audio recovery from noise, and image deblurring.
Chapters 7 and 8 are focused on the application of practical DSP applica-
tions through the Digital Signal Processor (DSP) hardware. The hardware
used in this book is the Texas Instruments TMS320C6711 Digital Signal
Processor Starter Kit. Chapter 7 deals in detail with the organization and
usage of the 6711 DSK, with a set of practical introductory exercises, such
as signal generation and filtering. Chapter 8 is more applied and covers the
hardware application and programming of the 6711 DSK for practical filter-
ing applications of noise from audio signals.
There are six appendices. The first four appendices give detailed hardware
description and user instructions of the equipment used in this book. The
four equipment models covered are synthesized sweep generators, spec-
trum analyzers, dynamic signal analyzers, and digitizing oscilloscopes in
Appendices A, B, C, and D, respectively. Appendix E gives detailed schemat-
ics, hardware description, and user instructions on the Texas Instruments
6711 DSK. Finally, Appendix F gives brief descriptions of alternate equipment
Preface xv
This book is organized into eight chapters and six appendices, with each
chapter typically having the following three sections: brief theory, computer
laboratory, and hardware laboratory. All eight chapters have theory and com-
puter laboratory sections; however Chapters 2, 5, and 6 do not have a hard-
ware section. Generally, each chapter includes a brief theory section, followed
by a MATLAB® and Simulink® simulation section and, finally, the hardware
section, which includes experiments on generation and measurement of sig-
nals using signal generators, digital oscilloscopes and spectrum analyzers,
and the Texas Instruments TMS320C6711 Digital Signal Processor Starter Kit.
This three-pronged approach is aimed at taking students from theory to
simulation to experiment in a very effective way. Additionally, instructors
have the option of selecting only the computer laboratory or hardware labo-
ratory or both for their individual classes, based on availability of software
or hardware.
xvii
Author
xix
xx Author
1.2 Applications of DSP
Given an analog signal, as in Figure 1.1, the following applications are possible:
0.5
Amplitude
–0.5
–1
0 0.2 0.4 0.6 0.8 1 1.2 1.4 1.6
Time [sec.]
Sh - e S - ell - s S - ea Sh - ell - s
Figure 1.1
Time waveform of “She sells sea shells.”
1
2 Digital Signal Processing Laboratory
Signal input
A/D
Modulation
Stage
Channel
D/A
Stage Demodulation
Signal output
Figure 1.2
Block diagram of digital communications system.
• Analyze the analog signal for use in a Voice Recognition Telephone System.
Speaker Speaker
recognition name
system
Speech
signal
Figure 1.3
Voice Recognition System.
Introduction to Digital Signal Processing 3
and then analyzed using a FFT. The FFT spectrum of the EKG, for
example, can reveal critical information on the patient, such as excess
of potassium (hyperkalemia) or lack of potassium (hypokalemia).
() (
x n = Aδ n − n0 ) (1.1)
1
Amplitude
–1
–2
0 0.2 0.4 0.6 0.8 1 1.2 1.4 1.6 1.8 2
t, sec.
2
1
Amplitude
–1
–2
0 0.2 0.4 0.6 0.8 1 1.2 1.4 1.6 1.8 2
t, sec.
Figure 1.4
Sampling process with T = 0.1 s.
4 Digital Signal Processing Laboratory
1
A
x(n)
0.5
0
0 1 2 3 4 5 6 7 8 9 10
1
B
x(n)
0.5
0
0 1 2 3 4 5 6 7 8 9 10
1
C
x(n)
0.5
0
0 1 2 3 4 5 6 7 8 9 10
1
n D
x(n)
–1
1 2 3 4 5 6 7 8 9 10 11
n
Figure 1.5
Examples of common discrete-time signals. (a) Unit impulse sequence x(n) = δ(n – 2). (b) Unit
step sequence x(n) = u(n). (c) Exponential signal x(n) = 0.8n u(n). (d) Sinusoidal sequence
x(n) = cos(πn/4).
() (
x n = δ n−2 ) (1.2)
() () ( ) (
x n = δ n +δ n−1 +δ n−2 +δ n−3 ) ( ) (1.3)
() () (
x n = u n −u n− 4 ) (1.4)
Introduction to Digital Signal Processing 5
( )
u n − n0 = 1, n ≥ n0 ;
= 0, n < n0 (1.5)
()
x n = anu n , a < 1 () (1.6)
()
x n = sin an u n ( ) () (1.7)
Example
Represent the following signal vectors in terms of impulse functions. Assume
that the first term in each vector corresponds to the first sample at n = 0.
1. 1 −2 5 3 −12
2. 0 0 5 3 −12
Solution
() ( ) ( )
1. 1 −2 5 3 −12 = δ n − 2δ n − 1 + 5δ n − 2 + 3δ n − 3 − 12δ n − 4 ( ) ( )
in impulse form.
( ) ( ) (
2. 0 0 5 3 −12 = 5δ n − 2 + 3δ n − 3 − 12δ n − 4 in impulse form.)
Example
Represent the following signal vectors in terms of unit step functions. Assume
that the first term in each vector corresponds to the first sample at n = 0.
1. 1 1 1 1 1
2. 0 0 2 2 2
6 Digital Signal Processing Laboratory
Solution
() ( )
1. 1 1 1 1 1 = u n − u n − 5 in unit step form.
( ) ( )
2. 0 0 2 2 2 = 2 u n − 2 − 2 u n − 5 in unit step form.
1.4.1 MATLAB ® Basics
Please try out each of the commands given below and familiarize yourself
with the types of MATLAB commands and formats.
>> filename
Introduction to Digital Signal Processing 7
The program will run, and the results and error messages, if any, will be
displayed on the screen. Plots will appear on a new screen.
Numbers
Generation of numbers
Generate the real numbers z1 = 3, z2 = 4.
>> z1 = 3
>> z2 = 4
>> z1 = 3+j*4
>> z2 = 4+j*5
>> z = 3+j*4
>> zm = abs(z) ; gives the magnitude of z
>> zp = angle(z) ; gives the phase of z in radians
>> z = z1 + z2 ; addition
>> z = z1 – z2 ; subtraction
>> z = x + y ; addition
>> z = x - y ; subtraction
8 Digital Signal Processing Laboratory
>> z = x. * y ; multiplication
>> z = x. / y ; division
S = 1 + 3 + 5. . . . . . . 99.
Vector Approach
>> i = 1 : 2 : 99; ; creates the vector i
>> S = sum (i); ; obtains the sum S
The use of a semicolon after any MATLAB command causes that command
output not to display on the screen. If the semicolon is not used, then each
command will be displayed on the screen, which might be time-consuming
when large loop statements are used.
Generate the discrete-time signal y(n) = n sin(πn/2) in the interval 0 ≤ n ≤ 10.
Vector Approach
>> n = 0 : 10; ; creates the vector n
>> y = n.*sin(pi*n/2); ; obtains the vector y
>> stem(n,y) ; plots the signal y vs. n with
impulses
Solution
1.4.2 Simulink Basics
After logging into MATLAB, you will receive the prompt >>. In order to
open up Simulink, type in the following:
10 Digital Signal Processing Laboratory
>> simulink
Demo Files
Try out the demo files, both in the main library window, and in the Toolboxes
window. There are several illustrative demonstration files in the areas of
signal processing, image processing, and communications.
Introduction to Digital Signal Processing 11
Example
Plot the following continuous-time signals using MATLAB. If you decide
that the signal is periodic, select a range for time t that will cover at least
three time periods.
() ( )
1. x t = 5 cos 2 π 15 t + 0.25π
2. x ( t ) = 5 cos 2 π ( 15 ) t − 0.5π + 5 cos 2 π ( 10 ) t + 0.3π
Solution
% MATLAB program
clear
t=0:0.001:0.5;
x=5*cos(2*pi*15*t+0.25*pi);
subplot(2,1,1);
plot(t,x)
xlabel(‘t,sec’);
ylabel(‘x(t)’);
title(‘Problem a’);
t=0:0.001:0.8;
x=5*cos(2*pi*15*t-0.5*pi)+5*cos(2*pi*10*t+0.3*pi);
subplot(2,1,2);
plot(t,x)
xlabel('t,sec');
ylabel('x(t)');
title('Problem b');
The plots are shown below in Figure 1.6(a) and Figure 1.6(b), respectively.
Example
Plot the following discrete-time signals using MATLAB. If you decide that
the signal is periodic, select a range for time t that will cover at least three
time periods.
() n
1. x n = 0.5 , in the range − 5 ≤ n ≤ 5
2. x ( n) = 5 cos ( ωn + 0.3π ) , ω = 1.5π.
Solution
% MATLAB program
clear
n=-5:5;
x=0.5^abs(n);
subplot(2,1,1);
12 Digital Signal Processing Laboratory
Problem a
5
4
3
2
1
x(t)
0
–1
–2
–3
–4
–5
0 0.05 0.1 0.15 0.2 0.25 0.3 0.35 0.4 0.45 0.5
t, sec
(a)
Problem b
10
2
x(t)
–2
–4
–6
–8
–10
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8
t, sec
(b)
Figure 1.6
Continuous-time signal example (a) Example (a), (b) Example (b).
Introduction to Digital Signal Processing 13
stem(n,x)
xlabel(‘n’);
ylabel(‘x(n)’);
title(‘Problem a’);
clear
n=-10:10;
x=5*cos(1.5*pi*n+0.3*pi);
subplot(2,1,2);
stem(n,x)
xlabel(‘n’);
ylabel(‘x(n)’);
title(‘Problem b’);
The plots are shown in Figure 1.7(a) and Figure 1.7(b), respectively.
1.5.1 Sources
Sources generate signals that vary in shape, amplitude, frequency, and phase.
The source utilized in the laboratory is the Hewlett Packard HP 3324A
Synthesized Sweep generator. Please see Appendix A for the manufacturer’s
specification details and other operating suggestions.
1.5.2 Measurement Devices
Measuring devices are utilized to accurately graph input signals in two
domains: time and frequency. The HP 54510A 100 MHz digitizing oscilloscope
measures the amplitude and frequency of signals as a function of time,
whereas the HP 8590L Signal Analyzer measures the spectrum of the input
signal as a function of frequency. Please see Appendices B and D for manu-
facturer’s sheet and details of these two measuring devices.
The HP 35665 Dynamic Signal Analyzer is a multipurpose device that
can measure signals in both time and frequency domains, in addition to
frequency response of devices. These analyzers are more advanced devices
that can also generate regular signals, as well as random noise. Please see
Appendix C for manufacturer’s details.
14 Digital Signal Processing Laboratory
Problem a
1
0.9
0.8
0.7
0.6
x(n)
0.5
0.4
0.3
0.2
0.1
0
–5 –4 –3 –2 –1 0 1 2 3 4 5
n
(a)
Problem b
5
1
x(n)
–1
–2
–3
–4
–5
–10 –8 –6 –4 –2 0 2 4 6 8 10
n
(b)
Figure 1.7
Discrete-time signal example (a) Example (a), (b) Example (b).
Introduction to Digital Signal Processing 15
1.6 Problem Solving
a. DSP primarily deals with commonly occurring signals such as
speech, music, video, EKG (heart), and EEG (brain). Research the
Internet to get information on the approximate frequency ranges of
the above five commonly occurring signals.
b. Approximately sketch the following discrete-time signals in the
range 0 ≤ n ≤ 10:
i. x(n) = u(n) – u(n – 3)
ii. x(n) = u(3 – n)
iii. x(n) = 0.5n [u(n) – u(n – 5)]
c. Determine if each of the following signals is periodic or nonperiodic.
If a signal is periodic, specify its fundamental period.
i. x(n) = e jπn
ii. x(n) = 0.7nu(n)
iii. x(n) = cos(πn/2)cos(πn/4)
d. A digital audio signal can be represented by a one-dimensional vector
x(n); however, a digital video signal requires a two-dimensional matrix
x(m,n). Describe two reasons why it is more difficult to process and
transmit video signals as compared to audio signals.
1.7 Computer Laboratory
Working with Vectors and Matrices
a. Write a MATLAB program to sketch the following discrete-time
signals in the time range of –10 ≤ n ≤ 10. Please label all the graph
16 Digital Signal Processing Laboratory
Time scope
Analog signal
x(t) = A sin(ωt)
Output y(t)
+ Time scope
Pseudo-random noise
n(t)
Time scope
Figure 1.8
Simulink model of a continuous-time system.
Introduction to Digital Signal Processing 17
x(n) z
(z – 0.3) + y(n)
0.4 z–1
Figure 1.9
Simulink model of a discrete-time system.
1.8 Hardware Laboratory
Basic Experiments Using Sources and Measuring Equipment
a. In this experiment, basic time and frequency measurements will be
performed using the oscilloscope and signal analyzer.
• Connect the equipment together as shown in the schematic in
Figure 1.10. Use BNC cables and a BNC Tee to connect the circuit.
Please ensure that the HP3324A Synthesized sweep generator
POWER button is in the OFF position.
• Set the sweep generator to output a sinusoidal signal, with an
amplitude of 5 volts and frequency f = 2 MHz. Observe the
time-domain signal output on the oscilloscope, and note down
the measured amplitude and frequency of the sinusoidal signal.
• Set the signal analyzer to a START frequency of 1.0 MHz and
a STOP frequency of 3.0 MHz. Observe the frequency-domain
18 Digital Signal Processing Laboratory
HP 54510A
BNC
Oscilloscope
HP 3324A Tee
Synthesized
sweep generator
HP 8590L
RF Spectrum
analyzer
Figure 1.10
Experimental setup for signal time and frequency analysis.
C = 3.4 nF
+ +
– –
Figure 1.11
Low pass filter circuit with 10 kHz cutoff frequency.
HP 35665A Dynamic
signal analyzer
Source Ch 1 Ch 2
Low pass
filter circuit
Input Output
Figure 1.12
Experimental setup for frequency response measurement.
20 Digital Signal Processing Laboratory
Bibliography
1. E. Brigham, Fast Fourier Transform and Its Applications, Pearson Education, 1988.
2. B.P. Lathi, Modern Digital and Analog Communication Systems, 3rd Edition, Oxford
University Press, 1998.
3. Student Edition of MATLAB/Simulink, Mathworks, Natick, Massachusetts.
4. A.V. Oppenheim and R.W. Schafer, with J.R. Buck, Discrete-Time Signal Processing,
2nd Edition, Prentice Hall, New Jersey, 1998.
5. B.P. Kumar, Digital Signal Processing Laboratory, California State University,
Sacramento, 2003.
2
Discrete-Time LTI Signals and Systems
2.1.1 Linear Convolution
Linear convolution is a natural process of LTI systems. It defines the input–output
relation of the system and is defined as:
() () ()
y n = x n ∗h n (2.1)
where the ∗ symbol denotes the convolution process, x(n) is the system input,
y(n) is the system output, and h(n) is the impulse response of the system. The
impulse response is the output of the system, when the input x(n) = δ(n), the
unit impulse. The actual convolution process is defined as:
∞
( ) ∑ x ( k ) h (n − k )
y n = (2.2)
k = −∞
for all values of n. Convolution can be performed using the sliding tape
method, as shown below, or more practically using computer software, as will
be described in Section 2.6.
Example
Determine the linear convolution of the two discrete-time sequences, x(n)
and h(n), given by:
21
22 Digital Signal Processing Laboratory
()
x n = 1 1 1 1
()
h n = 1 2 3
Solution
The linear convolution is given by:
∞
( ) ∑ x ( k ) h (n − k )
y n =
k = −∞
Sliding Tape method: This method [1] can be done by hand calculation,
if the number of points in both the sequences is quite small. The procedure
is as follows:
• Write the sequences x(m), h(m), and h(–m) as shown below. The
sequence h(–m) is obtained by mirroring the sequence h(m) about the
m = 0 axis. Then the dot product of the vectors x(m) and h(–m) gives
the convolution output y(0).
( )
x m = 0 0 0 1 1 1 1
( )
h m = 0 0 0 1 2 3 0
( )
h − m = 0 3 2 1 0 0 0 ; y 0 = 1 ()
The zeros are introduced in x(m) and h(m) sequences to include the
negative time axis, which is required to generate the h(–m) sequence.
• Similarly, the next term in the table below, h(1 – m), is obtained by
shifting h(–m) by one step to the right. The dot product of the vectors
x(m) and h(1 – m) gives the convolution output y(1).
( )
h 1 − m = 0 0 3 2 1 0 0 ; y 1 = 3 ()
• The process is continued until the output y(n) remains at zero.
( )
h 2 − m = 0 0 0 3 2 1 0 ; y 2 = 6 ()
( )
h 3 − m = 0 0 0 0 3 2 1 ; y 3 = 6 ()
Discrete-Time LTI Signals and Systems 23
( )
h 4 − m = 0 0 0 0 0 3 2 ; y 4 = 5 ()
( )
h 5 − m = 0 0 0 0 0 0 3 ; y 5 = 3 ()
Any more shift in the sequence h(–m) will result in a zero output.
Hence, the output vector is:
()
y n = 1 3 6 6 5 3
• Note that the length of the output vector y(n) = [length of x(n)] + [length of
h(n)] – 1 = 4 + 3 – 1 = 6. This is a general law of linear convolution.
() ( ) ( ) () ( ) (
a0 y n + a1 y n − 1 + … + aN y n − N = b0 x n + b1 x n − 1 + … + bM x n − M )
or compactly as follows:
N M
∑ a y (n − k ) = ∑ b x (n − k )
k k (2.3)
k=0 k=0
Example
Consider a discrete-time system with input x(n) and output y(n), which is
described by the following difference equation:
24 Digital Signal Processing Laboratory
() ( ) ()
y n = 2y n − 1 + x n
If the input x(n) = δ(n), determine the output y(n) for the time range 0 ≤ n ≤ 5,
given the initial conditions that y(n) = 0, for n < 0.
Solution
The difference equation can be solved recursively, starting with n = 0;
() ( ) () ()
y 0 = 2 y −1 + x 0 = 2 0 + 1 = 1
() () () ()
y 1 = 2y 0 + x 1 = 2 1 + 0 = 2
() () () ()
y 2 = 2y 1 + x 2 = 2 2 + 0 = 4
() () () ()
y 3 = 2y 2 + x 3 = 2 4 + 0 = 8
() () () ()
y 4 = 2 y 3 + x 4 = 2 8 + 0 = 16
() () () ( )
y 5 = 2 y 4 + x 5 = 2 16 + 0 = 32
∞
( ) ∑ x ( n) z − n
X z = (2.4)
n = −∞
where the variable z is complex, and the function X(z) is defined in the com-
plex plane. A concise list of z-transform properties is given in Table 2.1.
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— Pakkotyöläisen sikiö!…
— No, tule nyt, tule! — toisti poika ja otti askeleen häntä vastaan.
Iljan silmissä näytti kaikki pyörivän ympäri, vaan rinnassaan tunsi
hän väkevän voiman, joka ajoi häntä rohkeasti eteenpäin.
— Pane pois veitsi! — hiljaa toisti isäntä. Ilja, joka näki kaikki kuin
usvan läpi, laski veitsen tiskille, purskahti itkemään ja istuutui taas
lattialle. Hänen päätään huimasi ja särki, korva oli turvoksissa, hänen
rinnallaan tuntui olevan raskas paino, ja se vaikeutti sydämen
tykytystä, vähitellen nousi kurkkuun ja esti hänet puhumasta,
Isännän ääni kuului kuin hyvin kaukaa:
— Mutta sallikaa…
— Hyvä on! Minä lähden… Mutta pitäkää silmällä tuota poikaa, sen
neuvon annan teille… Hän uhkaa veitsellä… he, he! Hänen isänsä on
pakkotyössä… he, he!
— Ulos!
— Murhapoltosta…
— En tiedä… ei haluttanut…
— En…
— Uskon sinua… Entä sitten Karp, tuo sama Karp tuossa, kuinka
on hänen laitansa, — varastaako hän?
— Kuulin…
— Mitä tarkotat?
— Sulje puoti!…
— Seis!
*****
— Kas niin, nyt saamme taas olla yhdessä. Minulla on muuan kirja,
nimeltä "Albigensit"; se on sellainen historia, Siinä kerrotaan eräästä
Simon Montfort'ista… Se on ihmeellinen mies. — Ja Jaakko alkoi
nopeasti, katkonaisesti kertoa kirjan sisältöä. Ilja katsoi häneen ja
ajatteli tyytyväisesti, että hänen toverinsa oli vielä samanlainen kuin
ennen. Jaakon mielestä ei Iljan käytöksessä kauppias Stroganin
luona ollut mitään merkillistä. Kuunneltuaan Iljan kertomusta sanoi
hän yksinkertaisesti:
— Niin pitikin…
Iljan mielestä tuntui Jaakon suhde häntä kohdanneeseen
tapaukseen vähän liian kylmältä. Itse Petruhakin oli, kuultuaan pojan
selonteon tapahtumasta kalakaupassa, hyväksynyt hänen
menettelynsä olipa vielä sanonut:
— Eilen hän minua ylisti. Hyvin teit, sanoi. Mutta sellaisia näkyvät
olevankin kaikki ihmiset: edessä kiittävät, takana moittivat.
— Kunpa tulisit pian suureksi! Silloin läksisin minä… Vaan nyt olet
sinä kuin ankkuri, joka pidätät minua… Sinun tähtesi olen minä tässä
pahennuksen pesässä… ja menehdyn! Tekisin muuten matkan pyhiin
paikkoihin… Sanoisin pyhimyksille: Jumalan miehet? Hyväntekijät ja
puolustajat Jumalan tykönä! Vapahtakaa minut, joka raskautettu
olen! Puhukaa puolestani Jumalan, Isämme tykönä!
Sillä tavoin haaveili hän, kun kauppa kävi hyvin eikä kukaan
loukannut häntä raa'alla käytöksellä. Hän oli näet siitä lähtien, kun
oli ruvennut pitämään itseään riippumattomana miehenä, tullut hyvin
araksi kunniastaan ja loukkaantui pian.
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