Thesis Title

By
XYZ
20-EE-XXX

XYZ
20-EE-XXX

XYZ
20-EE-XXX

XYZ
20-EE-XXX

Final Year Project Report

Department of Electrical Engineering

HITEC University Taxila Cantt, Pakistan

Spring 2024
HITEC University Taxila Cantt, Pakistan

Thesis Title

A Thesis Presented to

HITEC University Taxila Cantt, Pakistan

In partial fulfillment
of the requirement for the degree of

B.S. Electrical Engineering

By
XYZ
20-EE-XXX

XYZ
20-EE-XXX

XYZ
20-EE-XXX

XYZ
20-EE-XXX

Spring 2024
i
 Declaration
We, hereby declare that this project neither as a whole nor as a part there of
has been copied out from any source. It is further declared that we have
developed this project and the accompanied report entirely on the basis of
our personal efforts made under the sincere guidance of our supervisor. No
portion of the work presented in this report has been submitted in the
support of any other degree or qualification of this or any other University
or Institute of learning, if found we shall stand responsible.

Signature:______________

Name: XYZ

Signature:______________

Name: XYZ

Signature:______________

Name: XYZ

Signature:______________

Name: XYZ

HITEC University Taxila Cantt, Pakistan

Spring 2024

ii
 Department of Electrical Engineering
HITEC University Taxila Cantt, Pakistan

The project ______________(Title in Bold)__________________, presented by:

1 XYZ 20-EE-XXX

2 XYZ 20-EE-XXX

3 XYZ 20-EE-XXX

4 XYZ 20-EE-XXX

under the supervision of their project advisor and approved by the project examination
committee, has been accepted by the HITEC University Taxila Cantt, Pakistan, in partial
fulfillment of the requirements for the four year degree of B.S. Electrical Engineering.

(Supervisor Name)
Designation, Project Advisor

(Dr. Muhammad Ali Mughal)

Chairperson-EED
HITEC University Taxila

iii
 Dedication

In this page you can dedicate your project to which you want to dedicate your work.

iv
 Acknowledgement

In this page you are advised to give appreciation to those teachers who have helped you
during your projects, and also the name of those who have guide you through out your
project thesis, evaluation.

v
 Abstract

An abstract is a brief summary of a research article, thesis, review, conference proceeding

or any in-depth analysis of a particular subject or discipline, and is often used to help the
reader quickly ascertain the paper's purpose. When used, an abstract always appears at the
beginning of a manuscript or typescript, acting as the point-of-entry for any given
academic paper or patent application. Abstracting and indexing services for various
academic disciplines are aimed at compiling a body of literature for that particular
subject. The terms précis or synopsis are used in some publications to refer to the same
thing that other publications might call an "abstract". In management reports, an
executive summary usually contains more information (and often more sensitive
information) than the abstract does.

vi
 TABLE OF CONTENTS

1 INTRODUCTION........................................................................................................1

1.1 Some Important Points..........................................................................................1

1.2 Speech Coding.......................................................................................................2

2 LITERATURE SURVEY............................................................................................6

2.1 Detailed Survey.....................................................................................................6

2.1.1 More Detailed Survey....................................................................................6

2.2 Second Part............................................................................................................7

3 REFERENCES...........................................................................................................10

vii
 LIST OF FIGURES
Figure 1-1: A low frequency microstrip filter......................................................................1
Figure 2-1: 3D and polar representation of an antenna radiation pattern............................1

viii
 LIST OF TABLES
Table 1-1: Digital Modulation Schemes..............................................................................1

ix
 LIST OF ABBREVIATIONS
1G First generation
2D Two dimensional
3-D Three dimensional
4G Fourth generation
ABW Absolute bandwidth
API Application programming interface
CAD Computer aided design
DAA Detect and Avoid
dB Decibels
ECC Electronic communications committee
ETRI Electronics and telecommunications research institute
FBW Fractional bandwidth
FCC Federal communications commission
FDTD Finite difference time domain
FEM Finite element method
FIT Finite integration technique
FNBW First null beam width
GA Genetic algorithm
GHz Giga hertz
HPBW Half power beamwidth
IDA Infocomm development authority
ISM Industrial Scientific and Medicine
ITU International telecom union
MATLAB Matrix laboratory
MIC Ministry of Internal Affairs and Communications
mm Millimeter
MoM Method of Moments
PCB Printed circuit board
x
SNR Signal to noise ratio
UWB Ultra-wideband
VSWR Voltage standing wave ratio
Wi-Fi Wireless fidelity
WLAN Wireless local area network
WPAN Wireless personal area network

xi
Chapter One

1 INTRODUCTION

This is chapter one of report. It contains some general format rules, which ought to be
followed. Do not change the format, follow this one as it is.

1.1 Some Important Points

 Do not change headings size or font.
 Do not change the line spacing or font or size for normal text and for whole
document.
 When using bullets, do not add line space
 At start of each chapter you have to write chapter number in English, as written
above and then leave two blank lines of “Times New Roman” size 12. Then write
chapter name with number as shown above in “Times New Roman” size 18.
 Make sure you have selected page size “A4” before printing and also tell the
printing shop not to change the page margins. It is your responsibility to print the
report in proper format, with proper page margins.
 On side of your hard binding you have to write only your FYP title, along with
your group number.
 Make sure the equations are properly numbered. A sample of equation numbering
is given in this document. Equation should be in center of page and equation
number should be at right most location in small parenthesis, e.g. (x.y). Here “x”
is chapter number and “y” is equation number of that specific chapter.
 Similarly make sure that figures and tables are properly numbered and cited (if
you have taken them from any other source). Caption of figure is at end and
written as “Figure x-y”, where “x” is chapter number and “y” is figure number of
that specific chapter. Tables will follow same format, however, caption for table is
on top.
 You are required to write a dedicated chapter titled “IMPACT OF PROJECT ON
ENVIRONMENT AND SOCIETY” after discussing results of your project. In
environment part, you are required to write impact of your project on environment
and sustainability, and in society part you are required to write impact of your
project on our society.
 Make sure to use the proper page number as used in this document.
 Chapter 1 includes heading name “Report Organization”

Although with the emergence of optical fibers bandwidth in wired communications has
become inexpensive, there is a growing need for bandwidth conservation and enhanced

1
privacy in wireless cellular and satellite communications. In particular, cellular
communications have been enjoying a tremendous worldwide growth and there is a great
deal of R&D activity geared towards establishing global portable communications
through wireless personal communication networks (PCNs). On the other hand, there is a
trend toward integrating voice-related applications (e.g., voicemail) on desktop and
portable personal computers - often in the context of multimedia communications. Most
of these applications require that the speech signal is in digital format so that it can be
processed, stored, or transmitted under software control. Speech is generally band limited
to 4 kHz (or 3.2 kHz) and sampled at 8 kHz, although digital speech brings flexibility and
opportunities for encryption, it is also associated (when uncompressed) with a high data
rate and hence high requirements of transmission bandwidth and storage. Speech Coding
or Speech Compression is the field concerned with obtaining compact digital
representations of voice signals for the purpose of efficient transmission or storage.
Speech coding involves sampling and amplitude quantization. While the sampling is
almost invariably done at a rate equal to or greater than twice the bandwidth of analog
speech, there has been a great deal of variability among the proposed methods in the
representation of the sampled waveform. The objective in speech coding is to represent
speech with a minimum number of bits while maintaining its perceptual quality. The
quantization or binary representation can be direct or parametric. Direct quantization
implies binary representation of the speech samples themselves while parametric
quantization involves binary representation of speech model and/or spectral parameters.

The simplest non-parametric coding technique is Pulse Code Modulation (PCM), which is
simply a quantizer of sampled amplitudes. Speech coded at 64 kilobits per second (kbps)
using logarithmic PCM is considered as "non-compressed" and is often used as a
reference for comparisons. The term medium-rate for coding in the range of 8-16 kbps,
low-rate for systems working below 8 kbps and down to 2.4 kbps, and very-low-rate for
coders operating below 2.4 kbps.

1.2 Speech Coding

Speech coding at medium-rates and below is achieved using an analysis-synthesis
process. In the analysis stage, speech is represented by a compact set of parameters,
which are encoded efficiently. In the synthesis stage, these parameters are decoded and

2
used in conjunction with a reconstruction mechanism to form speech. Analysis can be
open-loop or closed-loop.

 Open loop
 Close loop

In closed-loop analysis, the parameters are extracted and encoded by minimizing

explicitly a measure (usually the mean square) of the difference between the original and
the reconstructed speech. Therefore closed-loop analysis incorporates synthesis and hence
this process is also called analysis-by-synthesis.

1 1 (1.1)
sin α ± sin β=2 sin ( α ± β ) cos ( α ∓ β )
2 2

Parametric representations can be speech or non-speech specific. Non-speech specific

coders or waveform coders are concerned with the faithful reconstruction of the time-
domain waveform and generally operate at medium-rates.

Figure 1-1: A low frequency microstrip filter [1]

Speech specific coders or voice coders (vocoders) rely on speech models and are focused
upon producing perceptually intelligible speech without necessarily matching the
waveform. Vocoders are capable of operating at very-low rates but also tend to produce
speech of synthetic quality.

Table 1-1: Digital modulation schemes [2]

Name of modulation
Total no. of symbols No. of bits in each symbol
scheme

BPSK 2 1

QPSK 4 2

3
 OQPSK 4 2

Although this is the generally accepted classification in speech coding, there are coders
that combine features from both categories. For example hybrid coders, which rely on
analysis-by-synthesis linear prediction. Hybrid coders combine the coding efficiency of
vocoders with the high-quality potential of waveform coders by modeling the spectral
properties of speech (much like vocoders) and exploiting the perceptual properties of the
ear, while at the same time providing for waveform matching (much like waveform
coders). Modern hybrid coders can achieve communications quality speech at 8 kbits/s
and below at the expense of increased complexity.

Although this is the generally accepted classification in speech coding, there are coders
that combine features from both categories. For example hybrid coders, which rely on
analysis-by-synthesis linear prediction. Hybrid coders combine the coding efficiency of
vocoders with the high-quality potential of waveform coders by modeling the spectral
properties of speech (much like vocoders) and exploiting the perceptual properties of the
ear, while at the same time providing for waveform matching (much like waveform
coders). Modern hybrid coders can achieve communications quality speech at 8 kbits/s
and below at the expense of increased complexity.

The International Telecommunications Union (ITU) is an international standards

organization chartered by the United Nations to formulate worldwide communications
standards. The members represent nearly every nation in the world, which delegates
typically from the largest telecommunication service providers and equipment
manufacturers in those member countries. All equipment manufactured is according to
these standards and this ensures compatibility of equipment and protocols worldwide. The
most widely adopted ITU standards for speech coding in multimedia applications, are
G.728, G.729 and G.723.1.

The speech compression technology, to be designated as G.723.1, has enabled visual

telephony over the public telephone network, among a variety of other teleconferencing
and multimedia applications. This technology operates at data rates as low as 6.3 and 5.3
kbps producing a substantial improvement in compression ratios over existing ITU

4
standards - while maintaining high speech quality. The high bit rate has a great quality.
The low bit rate gives a good quality and provides system designers with additional
flexibility. The high quality speech is possible because of significant advances in the
digital speech compression introduced by the parties and by advances in digital signal
processing technologies.

The algorithm used for coding of speech at higher rate (6.3 kbps) is Multipulse Maximum
Likelihood Quantization (MP-MLQ) and for lower rate (5.3 kbps) is Algebraic-Code-
Excited Linear Prediction (ACELP). It is possible to switch between the two rates at any
frame boundary.

In this project we have studied and implemented the ITU G.723.1 speech codec in Java,
which provides in more flexible, extensible, robust, secure and platform independent
implementation.

1.3 Report Organization

The report is distributed in the following manner.

Chapter 2: This chapter presents the literature survey, which includes the overview from
different publications on speech compression.

Chapter 3: In this chapter we have examined characteristics of human speech, which will
serve as a foundation for discussing how voice can be analyzed and synthesized. By
discussing different voice-digitization methods, we will also cover different international
methods, laying the foundation for information presented in the chapters followed.

Chapter 4: This chapter presents a block-by-block explanation of the ITU G.723.1 dual
rate speech coder.

Chapter 5: This chapter illustrates the system design aspects of our codec.

Chapter 6: This chapter deals with the implementation aspects and the software
specifications of G.723.1 in Java.

Chapter 7: This chapter illustrates the observation made by executing our codec on
different machines and platforms.

Chapter 8: This chapter extracts the conclusion of the research and offers suggestions for
future attempts in this area.
5
Chapter Two

2 LITERATURE SURVEY

Andreas S. Spanias [5] provides an overview of speech coding methodologies with

emphasis on those algorithms that are part of the recent low-rate standards for cellular
communications. Although the emphasis is on the new low-rate coders, attempts to
provide a comprehensive survey by covering some of the traditional methodologies as
well. Which will not only point out key references but will also provide valuable
background to the beginners.

2.1 Detailed Survey

Richard V. Cox et al. [19] have compared different ITU standards, which are applicable
to low bit-rate multimedia communications. ITU Rec.G.729 8 kb/s CS-ACELP has a 15
ms algorithmic codec delay and provides network-quality speech. It was originally
designed for wireless applications, but is applicable for multimedia communications as
well. Annex A of Rec. G.729 is a reduced complexity version of the CS-ACELP coder. It
was designed explicitly for simultaneous voice and data applications that are prevalent in
low bit-rate multimedia communications. These two coders use the same bit-stream
format and can interoperate. ITU Rec. G.723.1 6.3 and 5.3 kb/s speech coder for
multimedia communications was designed originally for low bit-rate videophones. Its
frame size of 30 ms and one-way algorithmic codec delay of 37.5 ms allow for a further
reduction in bit rate compared to the G.729 coder. In applications where low-delay is
important, the delay of G.723.1 may be too large. However, if the delay is acceptable,
G.723.1 provides a lower complexity alternative to G.729 at the expense of a slight
degradation in quality. The authors describe the attributes of speech coders such as bit
rate, complexity, delay and quality, and discuss the basic concepts of the three ITU coders
by comparing their specific attributes.

6
2.1.1 More Detailed Survey
Kashif Israr Siddiqui et al. [21] gives a brief account of their work i.e. to implement and
optimize a dual-rate speech codec for real-time operation on TriMedia's Very Long
Instruction Word (VLIW) Digital Signal Processor (DSP), Central Processing Unit (CPU)
so that the speech codec can operate under limited processor resources. They
implemented the speech codec which has two-bit rates associated with it, 5.3 and 6.3
kbits/s. This codec was optimized to represent speech with a high quality at the above
rates using a limited amount of complexity.

2.2 Second part

Fu-Kun Chen et al. [24] have proposed condensed stochastic codebook search approaches
that progressively reduce the computation required for the algebraic code excited linear
predictive (ACELP) and multi-pulse maximum likelihood quantization (MP-MLQ)
coders. By reducing the candidates of the codebook before search procedure, the
proposed methods can effectively diminish the computation required for the ITU-T
G.723.1 dual rate speech coder. Their simulation results show that the proposed methods
can save over 50 percent for the stochastic codebook search with perceptually intangible
degradation in speech quality.

Figure 2-2: 3D and polar representation of an antenna radiation pattern [1]

7
J. P. Woodard and L. Hanzo [25], have considered extensions to the Analysis-by-
Synthesis (AbS) loop used in Code Excited Linear Predictive (CELP) speech codecs.
They have examined the methods for updating the short-term synthesis filter once the
excitation parameters have been determined. They show that significant improvements
can be achieved by updating the synthesis filter, similar to those obtained using the well-
known methods of interpolation and bandwidth expansion. However their proposed
method of update avoids the increase in the delay of a codec that is usually associated
with interpolation. Furthermore the traditional sequential method of determining the
adaptive and fixed codebook parameters is examined and compared to an exhaustive
search of both codebooks. Three sub-optimum techniques were proposed for improving
the performance of the codebook search while maintaining a reasonable level of
complexity. The most complex of these increases the codec complexity by only about
40% but provides 80% of the maximum possible 1.1 dB segmental SNR improvement
associated with an exhaustive codebook search.

Benjamin W. Wah et al. [26] discuss a fundamental issue in real-time interactive voice
transmissions over unreliable IP networks due to the loss or late arrival of packets for
playback. This problem is especially serious when transmitting low bit rate-coded speech
with pervasive dependencies introduced. In such a case, the loss or late arrival of a single
packet will lead to the loss of subsequent dependent frames. In their paper, they have
described end-to-end loss-concealment schemes for ensuring high quality in playback.
They propose a novel multiple description-coding methods for concealing packet losses
in transmitting low bit rate-coded speech. Based on high correlations observed in linear
predictor parameters in the form of Line Spectral Paris (LSPs) of adjacent frames, they
generate multiple descriptions in senders by interleaving LSPs, and reconstruct lost LSPs
in receivers by linear interpolations. As excitation codewords have low correlations, they
further enlarge the segment size for excitation generation and replicate excitation
codewords in all descriptions in order to maintain the same transmission bandwidth.

J. P. Woodard et al. [27] have developed a programmable 8-16 kbps low-delay speech
codec, which is compatible with the G.728 16 kbps ITU codec at its top rate and exhibits
similarly attractive trade-offs in terms of speech quality, delay and complexity in the
range of 8-16 kbps.

8
Thomas J. Dillon, Jr. [36] application report describes how the G.723.1 Dual-Rate Speech
Coder has been implemented on the Texas Instruments (TIE) TMS320C62x digital signal
processor (DSP). Beyond the use of the ’C62x intrinsic functions, the application report
includes specific changes required to allow this coder to operate in a real-time system
with other speech coders. Also reported is information on several optimization techniques
used to yield multiple channels running concurrently. Finally, the application report
includes the performance resulting from this implementation of the algorithm.

9
 3 IMPACT OF PROJECT ON ENVIRONMENT AND
SOCIETY

This chapter describes the impact of project on environment and society.

10
 4 REFERENCES

Write the name of only principal author (the first author) and use “et al.” if there are more
than one author. Papers that have not been published cannot be cited in report. Papers that
have been accepted for publication should be cited as “in press” [5]. Capitalize only the
first word in a paper title, except for proper nouns and element symbols.

Use the following references styles for journal paper, conference paper, book,
thesis/report and website.

It is better if you use Endnote X9 for reference management. It will automatically format
all the references.

[1] A. Swaminathan, Y. Mao, and M. Wu, “Robust and secure image hashing,”
Information Forensics and Security, IEEE Transactions on, vol. 1, no. 2, pp. 215-
230, 2006.
[2] S. Roy and Q. Sun, “Robust hash for detecting and localizing image tampering,”
in Image Processing, 2007. ICIP 2007. IEEE International Conference on, vol. 6,
pp. VI-117, IEEE, 2007.
[3] S. Xiang, H. J. Kim, and J. Huang, “Histogram-based image hashing scheme
robust against geometric deformations,” in Proceedings of the 9th workshop on
Multimedia & security, pp. 121-128, ACM, 2007.
[4] M. K. Mihcak and R. Venkatesan, “New iterative geometric methods for robust
perceptual image hashing,” in Security and privacy in digital rights management,
pp. 13-21, Springer, 2001.
[5] J. Fridrich and M. Goljan, “Robust hash functions for digital watermarking," in
Information Technology: Coding and Computing, 2000. Proceedings. Inter-
national Conference on, pp. 178-183, IEEE, (in press)
[6] S. William, “Cryptography and network security: principles and practice," 3rd ed.,
vol. 2, Prentice-Hall, Inc, pp. 23-50, 1999.
[7] V. Monga, “Perceptually based methods for robust image hashing”, PhD. thesis,
in Department of Electrical and Computer Engineering, The University of Texas
at Austin, 2005.
[8] W. Li, “Perceptual Multimedia Hashing,” PhD thesis, in Department of Electrical
Engineering (ESAT), Katholieke University Leuven, Haverlee (Belgium) pp.208,
2012.
[9] N. Zivic, “Robust Image Authentication in the Presence of Noise”. 1 st ed., vol. 1,
Springer, pp. 20-30 2015.

11
[10] “Interfacing of Powerful Micro Gear motor – 90 RPM (6-12V)”, Retrieved from
https://www.riecktron.co.za/en/product/2565, [Last Accessed: July 25, 2017]

12
 APPENDICES
Appendix A
A.1 User manual

13