Abstract

Speech is one of the oldest and most natural means of information exchange between
human. Over the years, Attempts have been made to develop vocally interactive
computers to realise voice/speech synthesis. Obviously such an interface would yield
great benefits. In this case a computer can synthesize text and give out a speech. Text-
To-Speech Synthesis is a Technology that provides a means of converting written text
from a descriptive form to a spoken language that is easily understandable by the end
user (Basically in English Language). It runs on python platform, and the methodology
used was Object Oriented Analysis and Development Methodology; while Expert
System was incorporated for the internal operations of the program. This design will be
geared towards providing a one-way communication interface whereby the computer
communicates with the user by reading out textual document for the purpose of quick
assimilation and reading development.
Introduction

Voice/speech synthesis is a field of computer science that deals with designing

computer systems that synthesize written text. It is a technology that allows a
computer to convert a written text into speech via a microphone or telephone. As an
emerging technology, not all developers are familiar with speech technology. While
the basic functions of both speech synthesis and speech recognition takes only
minutes to understand, there are subtle and powerful capabilities provided by
computerized speech that developers will want to understand and utilize.
Automatic speech synthesis is one of the fastest developing fields in the framework
of speech science and engineering. As the new generation of computing
technology, it comes as the next major innovation in man- machine interaction,
after functionality of Speech recognition (TTS), supporting Interactive Voice
Response (IVR) systems.
The basic idea of text-to-speech (TTS) technology is to convert written input to
spoken output by generating synthetic speech. There are several ways of
performing speech synthesis:
1. Simple voice recording and playing on demand;
2. Splitting of speech into 30-50 phonemes (basic linguistic units) and their re-
assembly in a fluent speech pattern;
3. The use of approximately 400 diaphones (splitting of phrases at the centre of
the phonemes and not at the transition).
The most important qualities of modern speech synthesis systems are its naturalness
and intelligibility. By naturalness we mean how closely the synthesized speech
resembles real human speech. Intelligibility, on the other hand, describes the ease
with which the speech is understood. The maximization of these two criteria is the
main development goal in the TTS field.

Objectives of the Study

The general objective of the project is to develop a Text-to-speech synthesizer for the physically
impaired and the vocally disturbed individuals using English language. The specific objectives are:
1. To enable the deaf and dumb to communicate and contribute to the growth of an
organization through synthesized voice.
2. To enable the blind and elderly people enjoy a User-friendly computer interface.
3. To create modern technology appreciation and awareness by computer operators.
4. To implement an isolated whole word speech synthesizer that is capable of converting
text and responding with speech
5. To validate the automatic speech synthesizer developed during the study.

Scope of the Study

The study is focused on an ideal combination of a human-like behaviour with computer application
to build a one-way interactive medium between the computer and the user. This application was
customized using only one (1) word sentence consisting of the numeric digit 0 to 9 that could be
used in operating a voice operated telephone system.
Human speech is inherently a multi modal process that involves the analysis of the uttered acoustic
signal and includes higher level knowledge sources such as grammar semantics and pragmatics. This
project intends to focus only on the acoustic signal processing without the incorporation of a visual
input.
Significance of the Study
This project has theoretical, practical, and methodological significance:
The speech synthesizer will be very useful to any researcher who may wish to venture into the
“Impact of using Computer speech program for brain enhancement and assimilation process in
human beings”.
This text-to-speech synthesizing system will enable the semi-illiterates assess and read through
electronic documents, thus bridging the digital divide. The technology will also find applicability in
systems such as banking, telecommunications (Automatic system voice output), transport, Internet
portals, accessing PC, emailing, administrative and public services, cultural centres and many
others. The system will be very useful to computer manufacturers and software developers as they
will have a speech synthesis engine in their applications.

Text – To - Speech Synthesis Defined

A speech synthesis system is by definition a system, which produces synthetic speech. It is
implicitly clear, that this involves some sort of input. What is not clear is the type of this input. If the
input is plain text, which does not contain additional phonetic and/or phonological information the
system may be called a text-to-speech (TTS) system. A schematic of the text-to-speech process is
shown in the figure 1 below. As shown, the synthesis starts from text input. Nowadays this may be
plain text or marked-up text e.g. HTML or something similar like JSML (Java Synthesis Mark-up
Language).

Figure 1: Schematic TTS

 Figure 2: A general functional diagram of a TTS System.

Representation and Analysis of Speech Signals

Continuous speech is a set of complicated audio signals which makes producing them artificially
difficult. Speech signals are usually considered as voiced or unvoiced, but in some cases they are
something between these two. Voiced sounds consist of fundamental frequency (F0) and its
harmonic components produced by vocal cords (vocal folds). The vocal tract modifies this excitation
signal causing formant (pole) and sometimes anti-formant (zero) frequencies (Abedjieva et al.,
1993). Each formant frequency has also amplitude and bandwidth and it may be sometimes difficult
to define some of these parameters correctly. The fundamental frequency and formant frequencies
are probably the most important concepts in speech synthesis and also in speech processing. With
purely unvoiced sounds, there is no fundamental frequency in excitation signal and therefore no
harmonic structure either and the excitation can be considered as white noise.
The airflow is forced through a vocal tract constriction which can occur in several places between
glottis and mouth. Some sounds are produced with complete stoppage of airflow followed by a
sudden release, producing an impulsive turbulent excitation often followed by a more protracted
turbulent excitation (Allen et al., 1987). Unvoiced sounds are also usually more silent and less
steady than voiced ones.
Speech signals of the three vowels (/a/ /i/ /u/) are presented in time-frequency domain in Figure 3.
The fundamental frequency is about 100 Hz in all cases and the formant frequencies F1, F2, and F3
with vowel /a/ are approximately 600 Hz, 1000 Hz, and 2500 Hz respectively. With vowel /i/ the
first three formants are 200 Hz, 2300 Hz, and 3000 Hz, and with /u/ 300 Hz, 600 Hz, and 2300 Hz.

Figure 3: The time-frequency domain presentation of vowels /a/, /i/, and /u/.

Applications of Speech Synthesis

The application of synthetic speech is expanding fast whilst the quality of TTS systems is also
increasing steadily. Speech synthesis systems are also becoming more affordable for common
customers, making these systems suitable for everyday use. For example, better availability of TTS
systems may increase employability for people with communication difficulties. Listed below are
some of the applications of TTS system:
1. Applications for the Blind. 2. Applications for the Deafened and Vocally Handicapped
3. Educational Applications. 4. Applications for Telecommunications and Multimedia
5. Other Applications and Future Directions (e.g. Human-Machine Interaction)

Methodology and System Analysis

Analysis and Problems of Existing Systems

Existing systems algorithm is shown below in Figure 4. It shows that the system does not have an
avenue to annotate text to the specification of the user rather it speaks plaintext.

START

STRING VARIABLE DECLARATION

INPUT TEXT

ALLOCATE ENGINE
AND RESOURCES

SPEAK PLAINTEXT

DEALLOCATE ENGINE PRINT STACK

AND RESOURCES TRACE

STOP

Figure 4: Algorithm of already existing systems

Due studies revealed the following inadequacies with already existing systems:
1. Structure analysis: punctuation and formatting do not indicate where paragraphs and other
structures start and end. For example, the final period in “P.D.P.” might be misinterpreted as the
end of a sentence.
2. Text pre-processing: The system only produces the text that is fed into it without any pre-
processing operation occurring.
3. Text-to-phoneme conversion: existing synthesizer system can pronounce tens of thousands or
even hundreds of thousands of words correctly if the word(s) is/are not found in the data
dictionary.

Expectation of the New System

It is expected that the new system will reduce and improve on the problems encountered in the old
system. The system is expected to among other things do the following;
1. The new system has a reasoning process. 2. The new system can do text structuring and annotation.
3. The new system’s speech rate can be adjusted. 4. The Pitch of the voice can be adjusted.
5. You can select between different voices and can even combine or juxtapose them if you want
to create a dialogue between them
6. It has a user friendly interface so that people with less computer knowledge can easily use it
7. It must be compatible with all the vocal engines 8. It complies with SSML specification.

Choice of Methodology for the New System

Two methodologies were chosen for the new system: The first methodology is Object Oriented
Analysis and Development Methodology (OOADM). OOADM was selected because the system has
to be represented to the user in a manner that is user-friendly and understandable by the user. Also
since the project is to emulate human behaviour, Expert system had to be used for mapping of
Knowledge into a Knowledge base with a reasoning procedure. Expert system was used in the
internal operations of the program, following the algorithm of Rule- Based computation. The
technique is derived from general principles described by researchers in knowledge engineering
techniques (Murray et al., 1991; 1996).
The system is based on processes modelled in cognitive phonetics (Hallahan, 1996; Fagyal, 2001)
which accesses several knowledge bases (e.g. Linguistic and phonetic knowledge bases, Knowledge
bases about non- linguistic features, a predictive model of perceptual processes, and knowledge base
about the environment).

Speech Synthesis Module

The TTS system converts an arbitrary ASCII text to speech. The first step involves extracting the
phonetic components of the message, and we obtain a string of symbols representing sound-units
(phonemes or allophones), boundaries between words, phrases and sentences along with a set of
prosody markers (indicating the speed, the intonation etc.). The second step consists of finding the
match between the sequence of symbols and appropriate items stored in the phonetic inventory and
binding them together to form the acoustic signal for the voice output device.

Figure 5: Phases of TTS synthesis process

To compute the output, the system consults

1. A database containing the parameter values for the sounds within the word,
2. A knowledge base enumerating the options for synthesizing the sounds.
Incorporating Expert system in the internal programs will enable the new TTS system exhibit these features:
1. The system performs at a level generally recognized as equivalent to that of a human expert
2. The system is highly domain specific.
3. The system can explain its reasoning process
4. If the information with which it is working is probabilistic or fuzzy, the system can
correctly propagate uncertainties and provide a range of alternative solution with
associated likelihood.

SOURCE INPUT DATA

CONTROL STRUCTURE
OUTPUT
RULE INTERPRETER

KNOWLEDGE WORKING
BASE MEMORY

Figure 6: Data flow diagram of the Speech synthesis system Using Gane and Sarson
Symbol User Interface (Source): This can be Graphical User Interface (GUI), or the Command
Line Interface (CLI).
Knowledge Base (Rule set): FreeTTS module/system/engine. This source of the knowledge
includes domain specific facts and heuristics useful for solving problems in the domain. FreeTTS is
an open source speech synthesis system written entirely in the python programming language. It is
based upon Flite. FreeTTS is an implementation of Sun's Java Speech API. FreeTTS supports end-
of-speech markers.
Control Structures: This rule interpreter inference engine applies to the knowledge base
information for solving the problem.
Short term memory: The working memory registers the current problem status and history of solution to date.

SEMANTIC COMPONENT SYNTACTIC COMPONENT LEXICAL COMPONENT

SPEAKING RATE, USE CONSTITUENT STRUCTURE PHONETIC

OF CONTRASIVE (ESPECIALLY LOCATIONS OF STRING, LEXICAL
STRESS OR EMPHASIS PHRASE AND CLAUSE BOUNDARIES STRESS

PHONOLOGICAL COMPONENT

PHONETIC STRING STRESS

SEGMENTAL DURATION ASPECT

PHONETIC FEATURE
IMPLEMENTATION RULE

TEMPORAL PATTERN OF MOTOR

COMMANDS TO THE ARTICULATORS

ARTICULATORY MODEL

TEMPORAL PATTERN OF VOCAL TRACT

SHAPES, LARINGEAL CONFIGURATIONS,
AND SUBGLOTTAL PRESSURE CHANGES
 SPEECH

Figure 7: High Level Model of the Proposed System

 START

STRING VARIABLE DECLARATION

INPUT STRING

ALLOCATE ENGINE AND RESOURCES

IS INPUT YES
PLAINTEXT

NO SPEAK PLAIN TEXT

IS INPUT NO
ANNOTATED
WITH TAGS

YES
IGNORE THE TAG INFORMATION AND
NO STRUCTURE THE REST OF THE STRING
IS ANNOTATED
VALUE
INPUT JSML

YES
APPLY THE TAG INFORMATION AND SPEAK STRUCTURED TEXT
STRUCTURE THE STRING VALUE

NO PINT STACK TRACE

IS THE
OF THE REASONING
USER DONE
PROCESS

YES

DEALLOCATE ENGINE

STOP

Figure 8: Flowchart representation of the program

Choice of Speech Engine and Programming Language
The speech engine used in this new system was the FreeTTS speech engine. FreeTTS was used
because it is programmed using python (the backbone programming language of this designed TTS
system). It also supports SAPI (Speech Application Programming Interface) which is in
synchronism with the PSAPI (Python Speech Application Programming Interface). PSAPI was also
the standardized interface used in the new system.
FreeTTS includes an engine for the vocal synthesis that supports a certain number of voices (male
and female) at different frequencies. It is recommended to use PSAPI to interface with FreeTTS
because PSAPI interface provides the best methods of controlling and using FreeTTS. FreeTTS
engine enable full control about the speech signal. This new designed system provides the possibility
to choose a voice between three types of voices: an 8 kHz, diphone male English voice named kevin,
a 16 kHz diphone male English voice named kevin16 and a16khz limited domain, male US English
voice named alan. The user could also set the properties of a chosen voice: the speaking rate, the
volume and the pitch.

A determining factor in the choice of programming language is the special connotation (JSML)
given to the program. This is a python specification mark-up language used to annotate spoken
output to the preferred construct of the user. In addition to this, there is the need for a language that
supports third party development of program libraries for use in a particular situation that is not
amongst the specification of the original platform.
Considering these factors, the best choice of programming language was python. Other factors that
made python suitable were its dual nature (i.e. implementing 2 methodologies with one language),
its ability to Implements proper data hiding technique (Encapsulation), its supports for inner abstract
class or object development, and its ability to provide the capability of polymorphism; which is a
key property of the program in question.

Design of the New System

Some of the technologies involved in the design of this system includes the following:
Speech Application Programming Interface (SAPI): SAPI is an interface between applications
and speech technology engines, both text-to-speech and speech recognition (Amundsen 1996). The
interface allows multiple applications to share the available speech resources on a computer without
having to program the speech engine itself. SAPI consists of three interfaces; The voice text
interface which provides methods to start, pause, resume, fast forward, rewind, and stop the TTS
engine during speech. The attribute interface allows access to control the basic behaviour of the
TTS engine. Finally, the dialog interface can be used to set and retrieve information regarding the
TTS engine.
Python Speech API (PSAPI): The Python Speech API defines a standard, easy-to-use, cross-
platform software interface to state-of-the-art speech technology. Two core speech technologies
supported through the Python Speech API are speech recognition and speech synthesis.
Speech recognition provides computers with the ability to listen to spoken language and to
determine what has been said. Speech synthesis provides the reverse process of producing
synthetic speech from text generated by an application, an applet or a user. It is often referred to as
text-to-speech technology.
Functions of the Abstract Classes

1. Menu Bar: This will have the function of selecting through many variables and File chooser system.
2. Monitor: Monitors the reasoning process by specifying the allocation process and de-allocation state
3. Voice System: This shows the different voice option provided by the system
4. Playable session: This maintains the timing of the speech being given out as output, and produces
a speech in synchronism with the rate specified.
5. Playable type: This specifies the type of text to be spoken, whether it is a text file or an annotated JSML file
6. Text-to-Speech activator: This plays the given text and produces an output
7. Player Model: This is a model of all the functioning parts and knowledge base representation in the program
8. Player Panel: This shows the panel and content pane of the basic objects in the program, and
specifies where each object is placed in the system
9. Synthesizer Loader: This loads the Synthesizer engine, allocating and de-allocating resources appropriately

Conclusion and Recommendation

Synthesizing text is a high technology advancement and artificial formation of speech given a text
to be spoken. With Text-to-Speech synthesis, we can actually mediate and fill in the lacuna provided
by not fully exploiting the capabilities of some handicapped individuals. It's never been so easy to
use a text-to-speech program, as just one click and your computer will speak any text aloud in a
clear, natural sounding voice.
Therefore, there is need to use Information Technology to solve the problem for the
Before the use of the new system, proper training should be given to the users. This training can
come in handy with proper tutor on how to handle JSML language and how to use it to annotate text
for the proper output and emphasis.