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Signals and Communication Technology
Keonwook Kim
Conceptual
Digital Signal
Processing
with MATLAB
Signals and Communication Technology
Series Editors
Emre Celebi, Department of Computer Science, University of Central Arkansas,
Conway, AR, USA
Jingdong Chen, Northwestern Polytechnical University, Xi'an, China
E. S. Gopi, Department of Electronics and Communication Engineering, National
Institute of Technology, Tiruchirappalli, Tamil Nadu, India
Amy Neustein, Linguistic Technology Systems, Fort Lee, NJ, USA
H. Vincent Poor, Department of Electrical Engineering, Princeton University,
Princeton, NJ, USA
This series is devoted to fundamentals and applications of modern methods of
signal processing and cutting-edge communication technologies. The main topics
are information and signal theory, acoustical signal processing, image processing
and multimedia systems, mobile and wireless communications, and computer and
communication networks. Volumes in the series address researchers in academia
and industrial R&D departments. The series is application-oriented. The level of
presentation of each individual volume, however, depends on the subject and can
range from practical to scientific.
**Indexing: All books in “Signals and Communication Technology” are indexed
by Scopus and zbMATH**
For general information about this book series, comments or suggestions, please
contact Mary James at mary.james@springer.com or Ramesh Nath Premnath at
ramesh.premnath@springer.com.
123
Keonwook Kim
Division of Electronics
and Electrical Engineering
Dongguk University
Seoul, Korea (Republic of)
This Springer imprint is published by the registered company Springer Nature Singapore Pte Ltd.
The registered company address is: 152 Beach Road, #21-01/04 Gateway East, Singapore 189721,
Singapore
To my parents
To Saea, Saeyun, and my lovely wife
Gumhong Kim
To all of my family
Preface
Digital signal processing (DSP) is an extensive study area to understand the signal
and noise in the digital domain. The advanced digital technology in the 60s and 70s
started to accelerate the theoretical signal processing for the feasible engineering
over the academia. In the 80s, the research in articles were organized and consol-
idated as practical textbooks for the graduate program. There were great textbooks
such as “Discrete-Time Signal Processing” by Oppenheim and Schafer and “Digital
Filter Design” by Parks and Burrus. The decade of the 90s further enjoyed the
processor technologies along with programming environments for DSP. MATLAB
(founded in 1984) has provided a reliable and convenient computer language (also
tools) to realize and verify the mathematical algorithms. Many textbooks appreciate
the benefits of MATLAB, for instance, “Signals and Systems Using MATLAB” by
Chaparro. For DSP, books and tools are existing.
This book addresses DSP in novel approach based on reconfiguration. The
intuitions and theories along with applications are the basic philosophy of the book.
The author tries to avoid the dictionary style (bottom-up structure) and to place
theories within applications. Students often experience a loss of motivation in the
middle of a semester due to the fragmented knowledge of digital filter design. The
author believes that the great researchers in DSP did not establish solid theories
from pure mathematics. The imagination initiates the research, and the equation
finalizes the theory. This book tries to provide the theories with derivations,
illustrations, and/or applications. Once the student has a certain picture of the
procedure and intelligence, the knowledge can be extended and maintained for a
long time. MATLAB also helps to design the top-down structure for the
self-motivated learning process. The book is organized as follows.
Chapter 1
• Basics of digital filters based on the intuition,
• Simple weight and sum of recent inputs for desired output,
• Design the low, high, and band pass filter.
vii
viii Preface
Chapter 2
• Definition of frequency in continuous and discrete time domain,
• Sampling theory,
• Discrete time signal representations.
Chapter 3
• Fundamentals to find the frequency magnitude from time domain,
• Discrete-time Fourier transform (DTFT) for non-periodic signal,
• Discrete Fourier transform (DTF) for periodic signal.
Chapter 4
• Reason to have the linear time invariance property,
• Finite impulse response (FIR) filter,
• Simple FIR filter design from the specification.
Chapter 5
• Extension of DTFT and DFT to Z-transform,
• Z-transform for infinite impulse response (IIR) filter,
• Intuitive IIR filter designs.
Chapter 6
• Understanding the filter specification,
• Advanced FIR filter designs,
• Advanced IIR filter designs.
Chapter 7
• Quantization effect for fixed-point number system,
• Implementation matters for FIR and IIR filters,
• Frequency domain filter realization.
Chapter 8
• Various filter realizations in MATLAB,
• Fixed-point number for MATLAB,
• C code generation for MATLAB.
In addition to above, there are two more sections for MATALB fundamentals
and Symbolic Math Toolbox as an appendix. This book includes comprehensive
information from basic DSP theories to the real-time filter realization of digital
computers and processors. The DSP class from this book is intended to provide
one- or two-semester program at the undergraduate level. The instructor can design
the DSP class for theory or practical intensive program as follows. Also, a
two-semester class can be handled using the complete contents of this book.
Preface ix
xi
xii Contents
Dr. Keonwook Kim received the B.S. degree in Electronics Engineering from
Dongguk University, Seoul, Korea in 1995 and the M.S. and Ph.D. degrees in
Electrical and Computer Engineering from the University of Florida, Gainesville,
United States in 1997 and 2001, respectively. He is presently Professor in the
Division of Electronics and Electrical Engineering at the Dongguk University. Prior
to joining Dongguk University, he worked as Assistant Professor in the Department
of Electrical and Computer Engineering at the Florida State University from 2001
to 2003. His primary research interest is acoustic localization via using the
multi-aural architecture in order to mimic the aural system of animals which include
human.
xiii
Chapter 1
Preliminary Digital Filter Design
The general filter passes various matters to separate out unwanted things. In elec-
trical engineering, the filter is known as the device for minimizing or suppressing
the noise frequencies to obtain the better signals. The digital filter handles the input
and output in discrete time domain as well as the quantized magnitude form. This
chapter introduces the digital filter from scratch. The filter equation is derived from
the general engineering concept. We assume that the input signal to the digital filter
presents the real-time property which represents the past, present, and future tense
in the signal. The conceptual derivation performed in this chapter actually operates
filter function in the real field based on the trial and error approach. In the following
chapters, the abstract design is followed by the further delicate analyses to meet the
accurate filter performance.
The continuous time is t as real number and discrete time is n that is the integer
number. The () handles the continuous arguments and [] deals with discrete argu-
ments. Any functions can be continuous and discrete form with arguments. The
given signal is x[n] as shown in Fig. 1.1.
The discrete signal with integer argument has no tense as past, present, and
future unless we specify the current time. That is the sequence of signal which
shows the relative position of numbers. The higher index number shows later, and
the lower index appears earlier. The real-time signal with n index provides the
present time position n which is the variable integer number. Therefore, the n + 1,
n + 2, and etc. are the future of the signal and n−1, n−2 … are the past of them.
Based on the time schedule, we only have the data up to the n index and n + 1 data
will be gathered next. The magnitude of the data is real number and any numbers
are possible for received value. The frequency components of the signal are not
numerically analyzed yet, but we presume that the signal with rapid fluctuation
© Springer Nature Singapore Pte Ltd. 2021 1
K. Kim, Conceptual Digital Signal Processing with MATLAB,
Signals and Communication Technology,
https://doi.org/10.1007/978-981-15-2584-1_1
2 1 Preliminary Digital Filter Design
1 2 ..
.
0
Fig. 1.1 a Example of discrete time signal with absolute time index number. b The corresponding
discrete time signal with real-time representation with n. The clock shows the current time n
contains the additional high frequency components. In contrast, the signal with
gentle movement includes the dominant low frequency components. The source of
the signal can be anything such as dice toss and heart rate. Conventionally the
discrete time signal is obtained from the continuous information by regular sam-
pling. We assume that the sampling distance for time is even; hence, the actual time
difference between adjacent indexes are always identical. After every fixed time, the
number for the discrete index n will be updated to the next integer. In the digital
signal processing, the x[n] has two meanings those are whole signal sequence
(variable n) and current signal value (given n). The discrete signal can be defined as
the signal sequence as shown below.
p
x½n ¼ cos n ð1:1Þ
2
If the sequence is used to build another discrete time signal, the input sequence is
employed as real-time signal as shown below.
1 1 1
y½n ¼ x½n þ x½n 1 þ x½n 2 ð1:2Þ
3 3 3
The y[n] function uses the current x value as x[n] and two previous values as x[n
−1] and x[n−2].
1.2 Design the Digital Filters 3
The real-time discrete time signal is given, and you are asked to design the filter to
reduce the noise frequencies and emphasize signal information. I believe that this is
the initial point of digital signal processing. Let’s design the filter.
The filter output y[n] is the operated results from the filter input x[n]s, and the
signal is obtained up to the n index value, as shown above Fig. 1.2. Any idea? Note
that the signal is continuously received by the system in every constant interval;
therefore, the system is willing to have the significant amount of data in short time.
It is impossible to process the whole obtained data for filtering; hence, the limited
length of data is considered to provide the filter output. The simple idea is to take
average the recent inputs for low pass filtering (smoothing). The recent N samples
are denoted as below.
1
y½n ¼ fx½n þ x½n 1 þ x½n 2 þ þ x½n N þ 1g ð1:4Þ
N
The filter output with input data is illustrated in Fig. 1.3. The recent three data
samples are considered to create average value for output y[n]. In Fig. 1.3, the
n values (n is the variable and not the current index here) are demonstrated from
Present
Future
Gathered data
n-5 n-4 n-3 n-2 n-1 n n+1 n+2 n+3 n+4 n+5
Fig. 1.2 One example of discrete time signal with real-time representation
4 1 Preliminary Digital Filter Design
seven to ten and the output values are noticeably smoothed comparing to the
income signal. This is the primitive low pass filter (LPF).
Example 1.1
Write the equation for digital average filter with recent three data.
Solution
1
y½n ¼ fx½n þ x½n 1 þ x½n 2g
3
∎
The independent parameter for the LPF is operation (averaging) and length
(three). As shown in Eq. (1.4), the averaging operation can be seen as the weighted
sum for the recent incoming data. Instead of using the averaging computation, the
different weight values can be applied on the data set. First of all, let’s change the
length of the averaging in the LPF as Fig. 1.4.
Example 1.2
Write the equation for digital average filter with recent seven data.
Solution
1
y½n ¼ fx½n þ x½n 1 þ þ x½n 5 þ x½n 6g
7
∎
1.2 Design the Digital Filters 5
For the increased length (seven) of averaging operation, the filter output y
[n] delivers further smoothed outcome than the shorter version (length three).
Therefore, depending on the filtering length, smoothness of the output can be
decided proportionally. In other word, the longer length averaging passes the
narrower range frequencies from zero frequency, and the shorter length averaging
passes the wider range frequencies that demonstrates the high fluctuation in filter
output. The length of the filter is important parameter to determine the frequency
range of the output.
Example 1.3
Write the equation for y[7] digital average filter in Fig. 1.5.
Solution
1
y½7 ¼ fx½7 þ x½6 þ þ x½2 þ x½1g
7
∎
The other parameter to be considered is weight values in the filter. The averaging
operation multiplies the constant 1/N to each data values for limited N length and
accumulates for filter output. The averaging range, latest N data, can be seen as the
window with weight for the given time sequence, and the window slides to the next
for new output. Figure 1.5 illustrates the explained procedures in terms of window
with seven window length. We understand that the filter length controls the output
6 1 Preliminary Digital Filter Design
1
y½n ¼ fx½n þ x½n 1 x½n 2g
3
∎
Unlike averaging filter, the filter output shows the higher variation in magnitude
than the original signal; therefore, the high frequency components are exaggerated via
the filtering process. The designed filter is high pass filter (HPF) with three window
length. The values in the window decide the characteristics of the filter which prefers
to pass the high frequency components; hence, we can think that the window shape
plays an important role. Comparing to the LPF, the HPF also uses the same window
length but the filter output is completely reverse in action. Applying the longer HPF
window derives same variation as LPF? The longer HPF length generates concen-
trated filter output to the high frequencies than the three HPF length situation as
Fig. 1.7.
1.2 Design the Digital Filters 7
Example 1.5
Write the equation for high pass filter in Fig. 1.7.
Solution
1
y½n ¼ fx½n þ x½n 1 þ þ x½n 5 x½n 6g
7
∎
The window length controls the frequency focus of the filter output in inversely
proportional manner. The longer window creates the further concentrated output for
the specific filter type which is determined by the window shape. The relationship
between the output frequencies and window length is derived from the above
simple experiments. How we can decide the filter type from window shape? Let’s
draw the two previously used filters in Fig. 1.8.
Can you figure out the filter type by observing the filter shape? Yes, the filter
output follows the window shape; therefore, smoothness and roughness of the
window provide the filter type that specify the designated frequency you want to
pass. Other than LPF and HPF, let’s perform another example. The window shape
of this filter is certain periodic signal which shows the 6-sample period and
19-sample length. The filter output is similar to the window shape as illustrated in
Fig. 1.9. The longer length expects to emphasize the window shape on the output
based on the intuition of previous lessons.
Fig. 1.8 a Low pass filter weight. b High pass filter weight
1.2 Design the Digital Filters 9
Example 1.6
Write the equation for digital filter in Fig. 1.9.
Solution
X
18
2pk
y½n ¼ x½n k cos
k¼0
6
∎
Up to now, we have designed primitive filters for LPF, HPF, etc. This is the
fundamental of the digital filter theory. The idea and computation initiated the
filtering system and signal processing in digital domain. The design method based
on the trial and error does not lend the solid foundation for building the intended
filter. The shown intuitive methods should be formularized in engineering area for
further analysis and application. This book not only provides the mathematical
representations of the algorithms but also illustrates the physical meanings of the
system extensively in the coming sections and chapters.
1 1 1 1
y½n ¼ x½n þ x½n 1 þ x½n 2 þ þ x½n N þ 1 ð1:5Þ
N N N N
The recent N samples of data sequence is averaged for the low pass filtering in
terms of weighting the 1/N on each sample. Since the weight values in the window
can be various for intended filtering purpose, the window is separated and slid
sequentially in every interval as Fig. 1.10. The signal sequence x[n] starts from the
zero n value and the window will be overlapped at the zero index. As the time goes
by, overlap range will be increased and completed after N−1 sample shift; hence,
we expect to have proper averaged output from the operations.
The window function is specified as below.
1
q½0 ¼ q½1 ¼ q½2 ¼ ¼ q½ðN 1Þ ¼
N
The second and third proper outputs are shown below with sliding input
sequence.
x[n]
... n
0 1 2 ...
q[n]
Sliding
1/N
... n
-(N-1) 0
Sum Sum
Sliding
... ... n
... ... n
Output
Output
y[n] y[n]
... n
... n
Fig. 1.10 The low pass filtering operation. The x[n] is the input sequence and the q[n] is the filter
weight window
1.3 Filter Architecture 11
X
N1
y½n ¼ x½n k q½k ð1:6Þ
k¼0
The n is the current time in the equation and the integer value n starts from zero
to infinite. Based on the recent N input sequence, the filter output is computed by
weight q[] function. This is the nice equation to formulate the intuitive concept of
the primitive filtering. Since the input data sequence initiates from the zero-time
index to positive number, the weight window is also flipped over the vertical axis in
order to start from the zero as below.
Due to the relocation, the tense of the weight window is changed completely as
shown in Fig. 1.11. The original window function q[] represents the q[0] as present
moment and the its tails on the left as past time. Note that the q[0] is always
multiplied with the present x[] value in the filtering computations shown at
Eq. (1.6). The further to the left indicates the earlier times in weight window. In the
flipped window function h[], the further to the right is past and h[0] is present time.
Therefore, the equation with h[] is below.
X
N 1
y½n ¼ x½n k h½k ð1:7Þ
k¼0
The only difference with non-flipped window filter is that there is no minus sign
in the window function. This is the digital filter equation also known as convolution
sum. The equation is the time domain filter with given input signal x[] and pro-
duces the filtered output y[]. For the specific purpose, the shape and length of h[]
have to be determined in analytical manner to improve the signal and reduce the
noise component from incoming discrete data sequence on Fig. 1.12.
The diagram for the filter is given as above. The real-time input sequence x[] is
provided to the filter system represented by h[]. The corresponding filter output is y
[].
filter. The output sequence placed at the filter input provides the output which only
includes the single frequency component. The multiple frequency components are
removed to obtain the signal frequency; therefore, the filter dose not insert any
frequencies to the output.
Example 1.7
Write the equation to add the zero frequency component over the signa x[n].
Solution
Since the constant values indicate the zero frequency, the output y[n] contains
the additional zero frequency component.
∎
Now we know that the designed filter by h[] can compute the output by
Eq. (1.7). When you have the unknown filter, how we can find the h[] and what is
the relationship with y[] from h[]? What signal figures out the h[]? Do not look at
the equation. Observe that we are dealing with discrete time signal. The primitive
and fundamental element of the discrete signal is the function that has single value
in the certain time index. Let’s see the example at Fig. 1.14.
14 1 Preliminary Digital Filter Design
q[n]
2
1
2
n
01
-1
=
q0[n] q1[n] q2[n]
2
1
+ + 2
n n n
012 012 01
-1
The function q[] is the linear combination (simple addition) of q0[], q1[] and
q2[]. The individual q1[] and q2[] can be represented by the shift location and
magnitude multiplication of q0[] as shown below.
Therefore, the manipulations and combinations of q0[] can provide the q[] and
any arbitrary functions. Based on this idea, we can think about the filter output by
analytical combinations. If the filter output of q[] is the linear combination of the
filter outputs from q0[], q1[] and q2[], then the q[] output can be decomposed into
the simpler form. Furthermore, the individual filter output of q1[] and q2[] can be
described by the shift location and magnitude multiplication of q0[] output just as
relationships of q0[], q1[] and q2[], then the filter output will be expressed by
linear combination of representative output like q0[] in this example. The gener-
ating output from individual inputs is illustrated in Fig. 1.15.
Figure 1.15 shows that the filter output can be derived from the linear combi-
nation and time shift of the primitive output. This kind of the system is called as
linear and time invariant system. The linear system denotes the filter outcome by
addition over scaled version of individual element outputs. The time invariant
system preserves the filter output shape for the time shifted input with identical time
relocation. With linear and time invariance condition, the system can be charac-
terized by very simple format that is the output of the single value in single time.
1.4 Digital Filter Definitions and Requirements 15
+ +
q1[n] y1[n]
2 2
n Response
n
012 01234
q2[n]
+ +
y2[n]
2
n Response n
01 01234
-1 -1
q[n] y[n] 3
2 2
1 1 1
2
n Response
01234
n
01
-1 -1
Any other outputs can be produced by the scaled and shifted combinations of the
characterizing output. This is very important property since the characterizing
output represents the complete feature of the system described as h[] in the
equation. How we can find the h[] from the unknown linear and time invariant
system? Let’s perform Fig. 1.10 example in reverse manner as below.
As shown in Fig. 1.16, the impulse signal (one magnitude at time zero), which is
the characterizing input, sequentially scans the h[] of the system and provides the h
[] of the filter. The discovered h[] tells us the property and performance of the filter
in detail. Therefore, the user can find the system feature of the unknown and linear
16 1 Preliminary Digital Filter Design
x[n]
n
0 1 2 ...
h[n]
0 1234 n
1 1 1 1 1
n n n n n
0 1 2 ... 0 1 2 ... 0 1 2 ... 0 1 2 ... 0 1 2 ...
h[-n]
Sliding
& time invariant filter by impulse signal input. Now, we understand that the h[] can
be designed and found for the specific filtering purpose. However, we have to
explore further to study the relationship between the h[] and frequency components
in coming chapters.
If you are going to build the digital filter by using the trial and error method, the
chapter one is enough information for you. The information above shows about
window shape and filter computation intuitively. The low pass filter is realized by
the constant weight window for filter equation and the length is determined by your
trials until you obtain the satisfactory outcome. Other types of the filters can be
implemented by the derived window and numerous executions. The fundamental
concept is delivered but certain analytical approaches are missed in above.
For the engineering aspects, the systematical methodology is required for
approximate closed form solutions in filter design. The frequency is fundamental
elements of the system input/output and the signal is the limited or unlimited
combination of frequency components in general. The definition of the frequency is
introduced in the first part of this book. The digital signal processing manages the
signal in discrete time domain; however, the frequency is well understood in the
continuous time domain. The frequency for the digital signal is explained with the
sampling theory that describes the signal transformation between the continuous
and discrete time domain. The digital signal can be decomposed into the simple
1.5 What We Need for Further (Optional) 17
elements with linear and time shifting combination; therefore, the fundamental and
essential signal elements are described and defined as well.
Typically, the given signals are delivered in time domain and do not represent
the frequency information directly. To understand the signal and analyze the per-
formance, the designer needs to see the spectral distribution in the signal. The
Fourier analysis provides the mathematical tool to transform the information
between the time and frequency. The Fourier analysis computes the magnitude and
phase (or delay) of the frequencies from the periodic and non-periodic signal.
However, the Fourier analysis does not generate all-round solutions to all transform
matters. The users need to note that there are numerous limitations and conditions to
apply the analysis over the signals. The physical meaning and overall constraints of
Fourier analysis are described in this book.
The time domain filter is already introduced in this chapter. The filter equation is
primitive but essential fundamental of the signal frequency managing. Upon the
understanding of the frequency definition and Fourier analysis, the filter equation is
revisited to comprehend further based on the mathematical study. The frequency
response of the filter from the window shape and length is derived to meet the
intended filter requirements. Also, the finite window is extended to the infinite
length by recursive filter architecture and its frequency response is explored as well.
By using the frequency modulation, the low pass filter can be located in any
frequency position for designated filter specification. The simple filters based on the
frequency modulation are provided for practical filter design.
The Fourier analysis can be extended to the Z-transform to understand the signal
and system with additional perspective. The Z-transform converts the subject
between the time domain and the Z domain known as complex number plane. The
rational polynomial of complex number Z represents limited and unlimited length
signal and system with constant coefficients. The solutions to the polynomial
provide the various information such as filter type, stability, and etc. The beauty of
the Z-transform is that the transform presents powerful mathematical tool to handle
and visualize the system in simple manner. Also, the Z-transform can be adopted to
design the any type filters.
The digital filter is implemented by the digital processor (or logic) with analog to
digital converter (ADC) and digital to analog converter (DAC). Since the capability
of the processor and ADC/DAC are limited, the filter algorithm and signal repre-
sentation should be optimized in terms of computation and dimension. Numerous
concerns should be exercised to realize the digital filter in real-time processing. In
the software perspective, the algorithm can be executed quickly via using the
corresponding fast algorithms. In the other hardware viewpoint, the numbers can be
handled in the fixed- or floating-point representation that can change the execution
accuracy and speed. Therefore, the algorithm structure and format significantly
affect the filter performance in various ways.
This book extensively uses the MATLAB to understand the digital signal pro-
cessing theory throughout the chapters. In the last chapter, the comprehensive
Exploring the Variety of Random
Documents with Different Content
nations shall not be barricaded by her or by others. Meanwhile, the
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retired to his winter quarters at Cairo, and the slaves returned to the
Nile. Meanwhile the Ministry, whilst permitting this shameful traffic,
has systematically neglected the gold and silver placers discovered
on the Midian coast, and evidently extending far southwards; in fact,
the old Ophir and Havilah. In Turkish Arabia (the vilayet province of
Yemen, near Sana'a), a new digging has been discovered, and, with
true Oriental exaggeration, has been proclaimed 'one of the richest
in the world.' But in the hands of the Turkish Government even a
diamond-mine, a Golconda, would be a losing affair; it can be
worked with profit only by European heads and hands. Meanwhile
Egypt must recover her prestige by abolishing slavery and by
exploiting her mineral wealth.
"To conclude. Poets are sometimes prophets; and we have a
specimen in the forecast of Camoens, which dates from the year of
grace 1572—
'Those fierce projectiles, of our days the work,
Murderous engines, dire artilleries,
Against Byzantine walls, where dwells the Turk,
Should long ago have belcht their batteries.
Oh, hurl it back, in forest caves to lurk,
Where Caspian crests and steppes of Scythia freeze,
That Turkish ogre-progeny multiplied
By potent Europe's policy and pride.'
"What also wrote Torquato Tasso, only a few years after Camoens?
'For if the Christian Princes ever strive
To win fair Greece out of the tyrant's hands,
And those usurping Ismaelites deprive
Of woeful Thrace, which now captived stands;
You must from realms and sea the Turks forth drive,
As Godfrey chased them from Judah's lands,' etc.
Amen, and so be it!
"R. F. B.
"Trieste."
APPENDIX F.
"On the 30th of June, a massacre of nearly all the Christians took
place at Jeddah on the Red Sea. Amongst the victims were Mr. Page,
the British Consul, and the French Consul and his lady. Altogether
the Arabs succeeded in slaughtering about twenty-five.
"H.M. steamship Cyclops was there at the time, and the captain
landed with a boat's crew, and attempted to bring off some of the
survivors, but he was compelled to retreat, not without having killed
a number of the Arabs. The next day, however, he succeeded in
rescuing the few remaining Christians, and conveyed them to Suez.
"Amongst those who were fortunate enough to escape was the
daughter of the French Consul; and this she succeeded in doing
through the fidelity of a native, after she had killed two men with
her own hands, and been severely wounded in the encounter.
Telegraphic despatches were transmitted to England and France, and
the Cyclops is waiting orders at Suez. As it was apprehended that
the news from Jeddah might excite the Arab population of Suez to
the commission of similar outrages, H.B.M.'s Vice-Consul at that
place applied to the Pasha of Egypt for assistance, which was
immediately afforded by the landing of five hundred Turkish soldiers,
under the orders of the Pasha of Suez."
Second Correspondence.
1.
2.
3.
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