Artificial Intelligence For Speech Recognition

ARTIFICIAL
INTELLIGENCE FOR
SPEECH
RECOGNITION
IRFANA TABASSUM T
ABSTRACT
Artificial intelligence (AI) for speech recognition involves two basic ideas. First, it involves
studying the thought processes of human beings. Second, it deals with representing those processes
via machines (like computers, robots, etc).AI is behavior of a machine, which, if performed by a
human being, would be called intelligence. It makes machines smarter and more useful, and is less
expensive than natural intelligence. Natural language processing (NLP) refers to artificial
intelligence methods of communicating with a computer in a natural language like English. The
main objective of a NLP program is to understand input and initiate action. The input words are
scanned and matched against internally stored known words. Identification of a keyword causes
some action to be taken. In this way, one can communicate with the computer in one's language
CONTENTS
1. Introduction 1
1.1 Problems 2
1.2 Tools 7
1.3 Applications 11
2. The Technology 12
3. Speech recognition 15
3.1 Speech recognition process 16
3.2 Terms and concepts 17
4. Speaker independency 19
4.1 Speaker Dependence Vs Speaker Independence 19
4.2 System Configuration 21
5. Working of the system 25
5.1 Speaker- dependent word recognizer 26
5.2 What software is available? 28
5.3 What technical issues need to be considered when
purchasing this system? 28
5.4 How does the technology differ from other technologies?29
5.5 What factors need to be considered when using speech
recognition technology? 30
6. The limits of speech recognition 31
7. Application 34
7.1 Health care 34
7.2 Military 35
7.3 Training air traffic controllers 38

7.4 Telephony and other domains 39
7.5 People with disabilities 39
7.6 Hands-free computing 39
8. Conclusion 41
9. Bibliography 42
Artificial intelligence for speech recognition
Chapter 1
INTRODUCTION
The speech recognition process is performed by a software component known as the
speech recognition engine. The primary function of the speech recognition engine is to process
spoken input and translate it into text that an application understands. The application can then
do one of two things: The application can interpret the result of the recognition as a command. In
this case , the application is a command and control application. If an application handles the
recognized text simply as text, then it is considered a dictation application. The user speaks to the
computer through a microphone, which in turn, identifies the meaning of the words and sends it
to NLP device for further processing. Once recognized, the words can be used in a variety of
applications like display, robotics, commands to computers, and dictation. No special commands
or computer language are required. There is no need to enter programs in a special language for
creating software. Voice XML takes speech recognition even further. Instead of talking to your
computer, you're essentially talking to a web site, and you're doing this over the phone. OK , you
say, well, what exactly is speech recognition? Simply put, it is the process of converting spoken
input to text. Speech recognition is thus sometimes referred to as speech-to-text .Speech
recognition allows you to provide input to an application with your voice. Just like clicking with
your mouse, typing on your keyboard, or pressing a key on the phone keypad provides input to
an application; speech recognition allows you to provide input by talking. In the desktop world,
you need a microphone to be able to do this. In the Voice XML world, all you need is a
telephone.
When you dial the telephone number of a big company, you are likely to hear the
sonorous voice of a cultured lady who responds to your call with great courtesy saying “welcome
to company X. Please give me the extension number you want” .You pronounce the extension
number, your name, and the name of the person you want to contact. If the called person accepts
the call, the connection is given quickly. This is artificial intelligence where an automatic call-
handling system is used without employing any telephone operator.
AI is the study of the abilities for computers to perform tasks, which currently are better
done by humans. AI has an interdisciplinary field where computer science intersects with
philosophy, psychology, engineering and other fields. Humans make decisions based upon
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Dept. of E&C H.K.B.K.C.E.
experience and intention. The essence of AI in the integration of computer to mimic this learning
process is known as Artificial Intelligence Integration.
1.1 Problems
The general problem of simulating (or creating) intelligence has been broken down into a
number of specific sub-problems. These consist of particular traits or capabilities that researchers
would like an intelligent system to display. The traits described below have received the most
attention.
1.1.1 Deduction, reasoning, problem solving
Early AI researchers developed algorithms that imitated the step-by-step reasoning that
humans were often assumed to use when they solve puzzles, play board games or make logical
deductions. By the late 1980s and '90s, AI research had also developed highly successful
methods for dealing with uncertain or incomplete information, employing concepts from
probability and economics.
For difficult problems, most of these algorithms can require enormous computational
resources — most experience a "combinatorial explosion": the amount of memory or computer
time required becomes astronomical when the problem goes beyond a certain size. The search
for more efficient problem solving algorithms is a high priority for AI research.
Human beings solve most of their problems using fast, intuitive judgments rather than the
conscious, step-by-step deduction that early AI research was able to model. AI has made some
progress at imitating this kind of "sub-symbolic" problem solving: embodied agent approaches
emphasize the importance of sensorimotor skills to higher reasoning; neural net research
attempts to simulate the structures inside human and animal brains that gives rise to this skill.
1.1.2 Knowledge representation
Knowledge representation and knowledge engineering are central to AI research. Many

of the problems machines are expected to solve will require extensive knowledge about the
world. Among the things that AI needs to represent are: objects, properties, categories and
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relations between objects situations, events, states and time causes and effects knowledge about
knowledge (what we know about what other people know) and many other, less well researched
domains. A complete representation of "what exists" is an ontology (borrowing a word from
traditional philosophy), of which the most general are called upper ontologies.
1.1.2.1 Among the most difficult problems in knowledge representation are:
 Default reasoning and the qualification problem

Many of the things people know take the form of "working assumptions." For example, if
a bird comes up in conversation, people typically picture an animal that is fist sized, sings, and
flies. None of these things are true about all birds. John McCarthy identified this problem in
1969 as the qualification problem: for any commonsense rule that AI researchers care to
represent, there tend to be a huge number of exceptions. Almost nothing is simply true or false in
the way that abstract logic requires. AI research has explored a number of solutions to this
problem.
 The breadth of commonsense knowledge

The number of atomic facts that the average person knows is astronomical. Research
projects that attempt to build a complete knowledge base of commonsense knowledge (e.g., Cyc)
require enormous amounts of laborious ontological engineering — they must be built, by hand,
one complicated concept at a time. A major goal is to have the computer understand enough
concepts to be able to learn by reading from sources like the internet, and thus be able to add to
its own ontology.
 The sub symbolic form of some commonsense knowledge

Much of what people know is not represented as "facts" or "statements" that they could
actually say out loud. For example, a chess master will avoid a particular chess position because
it "feels too exposed" or an art critic can take one look at a statue and instantly realize that it is a
fake. These are intuitions or tendencies that are represented in the brain non-consciously and
sub-symbolically. Knowledge like this informs, supports and provides a context for symbolic,
conscious knowledge. As with the related problem of sub-symbolic reasoning, it is hoped that
situated AI or computational intelligence will provide ways to represent this kind of knowledge.
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1.1.3 Planning
Intelligent agents must be able to set goals and achieve them. They need a way to
visualize the future (they must have a representation of the state of the world and be able to make
predictions about how their actions will change it) and be able to make choices that maximize the
utility (or "value") of the available choices.
In classical planning problems, the agent can assume that it is the only thing acting on the
world and it can be certain what the consequences of its actions may be. However, if this is not
true, it must periodically check if the world matches its predictions and it must change its plan as
this becomes necessary, requiring the agent to reason under uncertainty. Multi-agent planning
uses the cooperation and competition of many agents to achieve a given goal. Emergent behavior
such as this is used by evolutionary algorithms and swarm intelligence.
1.1.4 Learning
Machine learning has been central to AI research from the beginning. Unsupervised
learning is the ability to find patterns in a stream of input. Supervised learning includes both
classification and numerical regression. Classification is used to determine what category
something belongs in, after seeing a number of examples of things from several categories.
Regression takes a set of numerical input/output examples and attempts to discover a continuous
function that would generate the outputs from the inputs. In reinforcement learning the agent is
rewarded for good responses and punished for bad ones. These can be analyzed in terms of
decision theory, using concepts like utility. The mathematical analysis of machine learning
algorithms and their performance is a branch of theoretical computer science known as
computational learning theory
1.1.5 Natural language processing
Figure : 1.1 ASIMO uses sensors and intelligent algorithms to avoid obstacles and navigate stairs.
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Natural language processing gives machines the ability to read and understand the
languages that humans speak. Many researchers hope that a sufficiently powerful natural
language processing system would be able to acquire knowledge on its own, by reading the
existing text available over the internet. Some straightforward applications of natural language
processing include information retrieval (or text mining) and machine translation.
1.1.6 Motion and manipulation
Figure : 1.2
The Care-Providing robot FRIEND uses sensors like cameras and intelligent algorithms
to control the manipulator in order to support disabled and elderly people in their daily life
activities.The field of robotics is closely related to AI. Intelligence is required for robots to be
able to handle such tasks as object manipulation and navigation, with sub-problems of
localization (knowing where you are), mapping (learning what is around you) and motion
planning (figuring out how to get there)
1.1.7 Perception
Machine perception is the ability to use input from sensors (such as cameras,
microphones, sonar and others more exotic) to deduce aspects of the world. Computer vision is
the ability to analyze visual input. A few selected subproblems are speech recognition, facial
recognition and object recognition.
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1.1.8 Social intelligence
Figure : 1.3 Kismet, a robot with rudimentary social skills
Emotion and social skills play two roles for an intelligent agent. First, it must be able to
predict the actions of others, by understanding their motives and emotional states. (This
involves elements of game theory, decision theory, as well as the ability to model human
emotions and the perceptual skills to detect emotions.) Also, for good human-computer
interaction, an intelligent machine also needs to display emotions. At the very least it must
appear polite and sensitive to the humans it interacts with. At best, it should have normal
emotions itself.
1.1.9 Creativity
Figure : 1.4 TOPIO, a robot that can play table tennis, developed by TOSY.
A sub-field of AI addresses creativity both theoretically (from a philosophical and

psychological perspective) and practically (via specific implementations of systems that generate
outputs that can be considered creative). A related area of computational research is Artificial
Intuition and Artificial Imagination.
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1.2 Tools
In the course of 50 years of research, AI has developed a large number of tools to solve
the most difficult problems in computer science. A few of the most general of these methods are
discussed below.
1.2.1 Search and optimization
Many problems in AI can be solved in theory by intelligently searching through many

possible solutions Reasoning can be reduced to performing a search. For example, logical proof
can be viewed as searching for a path that leads from premises to conclusions, where each step is
the application of an inference rule. Planning algorithms search through trees of goals and
subgoals, attempting to find a path to a target goal, a process called means-ends analysis.
Robotics algorithms for moving limbs and grasping objects use local searches in configuration
space. Many learning algorithms use search algorithms based on optimization.
Simple exhaustive searches are rarely sufficient for most real world problems: the search
space (the number of places to search) quickly grows to astronomical numbers. The result is a
search that is too slow or never completes. The solution, for many problems, is to use
"heuristics" or "rules of thumb" that eliminate choices that are unlikely to lead to the goal (called
"pruning the search tree"). Heuristics supply the program with a "best guess" for what path the
solution lies on.
A very different kind of search came to prominence in the 1990s, based on the
mathematical theory of optimization. For many problems, it is possible to begin the search with
some form of a guess and then refine the guess incrementally until no more refinements can be
made. These algorithms can be visualized as blind hill climbing: we begin the search at a random
point on the landscape, and then, by jumps or steps, we keep moving our guess uphill, until we
reach the top. Other optimization algorithms are simulated annealing, beam search and random
optimization.
Evolutionary computation uses a form of optimization search. For example, they may
begin with a population of organisms (the guesses) and then allow them to mutate and
recombine, selecting only the fittest to survive each generation (refining the guesses). Forms of
evolutionary computation include swarm intelligence algorithms (such as ant colony or particle
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swarm optimization) and evolutionary algorithms (such as genetic algorithms and genetic
programming).
1.2.2 Logic
Logic was introduced into AI research by John McCarthy in his 1958 Advice Taker
proposal. Logic is used for knowledge representation and problem solving, but it can be applied
to other problems as well. For example, the satplan algorithm uses logic for planning and
inductive logic programming is a method for learning.
Several different forms of logic are used in AI research. Propositional or sentential logic
is the logic of statements which can be true or false. First-order logic also allows the use of
quantifiers and predicates, and can express facts about objects, their properties, and their
relations with each other. Fuzzy logic, is a version of first-order logic which allows the truth of a
statement to be represented as a value between 0 and 1, rather than simply True (1) or False (0).
Fuzzy systems can be used for uncertain reasoning and have been widely used in modern
industrial and consumer product control systems. Subjective logic models uncertainty in a
different and more explicit manner than fuzzy-logic: a given binomial opinion satisfies belief +
disbelief + uncertainty = 1 within a Beta distribution. By this method, ignorance can be
distinguished from probabilistic statements that an agent makes with high confidence. Default
logics, non-monotonic logics and circumscription are forms of logic designed to help with
default reasoning and the qualification problem. Several extensions of logic have been designed
to handle specific domains of knowledge, such as: description logics situation calculus, event
calculus and fluent calculus (for representing events and time) causal calculus belief calculus;
and modal logics.
1.2.3 Probabilistic methods for uncertain reasoning
Many problems in AI (in reasoning, planning, learning, perception and robotics) require
the agent to operate with incomplete or uncertain information. Starting in the late 80s and early
90s, Judea Pearl and others championed the use of methods drawn from probability theory and
economics to devise a number of powerful tools to solve these problems.
Bayesian networks are a very general tool that can be used for a large number of
problems: reasoning (using the Bayesian inference algorithm), learning (using the expectation-
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maximization algorithm), planning (using decision networks) and perception (using dynamic
Bayesian networks). Probabilistic algorithms can also be used for filtering, prediction, smoothing
and finding explanations for streams of data, helping perception systems to analyze processes
that occur over time (e.g., hidden Markov models or Kalman filters).
A key concept from the science of economics is "utility": a measure of how valuable
something is to an intelligent agent. Precise mathematical tools have been developed that analyze
how an agent can make choices and plan, using decision theory, decision analysis, information
value theory. These tools include models such as Markov decision processes, dynamic
1.2.4 Classifiers and statistical learning methods
The simplest AI applications can be divided into two types: classifiers ("if shiny then
diamond") and controllers ("if shiny then pick up"). Controllers do however also classify
conditions before inferring actions, and therefore classification forms a central part of many AI
systems. Classifiers are functions that use pattern matching to determine a closest match. They
can be tuned according to examples, making them very attractive for use in AI. These examples
are known as observations or patterns. In supervised learning, each pattern belongs to a certain
predefined class. A class can be seen as a decision that has to be made. All the observations
combined with their class labels are known as a data set. When a new observation is received,
that observation is classified based on previous experience.
A classifier can be trained in various ways; there are many statistical and machine
learning approaches. The most widely used classifiers are the neural network, kernel methods
such as the support vector machine, k-nearest neighbor algorithm, Gaussian mixture model,
naive Bayes classifier, and decision tree. The performance of these classifiers have been
compared over a wide range of tasks. Classifier performance depends greatly on the
characteristics of the data to be classified. There is no single classifier that works best on all
given problems; this is also referred to as the "no free lunch" theorem. Determining a suitable
classifier for a given problem is still more an art than science.
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1.2.5 Neural networks
Figure : 1.5
A neural network is an interconnected group of nodes, akin to the vast network of
neurons in the human brain.The study of artificial neural networks began in the decade before the
field AI research was founded, in the work of Walter Pitts and Warren McCullough. Other
important early researchers were Frank Rosenblatt, who invented the perceptron and Paul
Werbos who developed the back propagation algorithm.
The main categories of networks are acyclic or feed forward neural networks (where the
signal passes in only one direction) and recurrent neural networks (which allow feedback).
Among the most popular feed forward networks are perceptrons, multi-layer perceptrons and
radial basis networks. Among recurrent networks, the most famous is the Hopfield net, a form of
attractor network, which was first described by John Hopfield in 1982. Neural networks can be
applied to the problem of intelligent control (for robotics) or learning, using such techniques as
Hebbian learning and competitive learning.
Jeff Hawkins argues that research in neural networks has stalled because it has failed to
model the essential properties of the neocortex, and has suggested a model (Hierarchical
Temporal Memory) that is based on neurological research.
1.2.6 Control theory
Control theory, the grandchild of cybernetics, has many important applications,

especially in robotics.
1.2.7 Languages
AI researchers have developed several specialized languages for AI research, including

Lisp and Prolog.
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1.3 Applications
Artificial intelligence has successfully been used in a wide range of fields including
medical diagnosis, stock trading, robot control, law, scientific discovery, video games, toys, and
Web search engines. Frequently, when a technique reaches mainstream use, it is no longer
considered artificial intelligence, sometimes described as the AI effect .It may also become
integrated into artificial life.
1.3.1Competitions and prizes
There are a number of competitions and prizes to promote research in artificial

intelligence. The main areas promoted are: general machine intelligence, conversational
behavior, data-mining, driverless cars, robot soccer and games.
1.3.2 Platforms
A platform (or "computing platform")is defined by Wikipedia as "some sort of hardware

architecture or software framework (including application frameworks), that allows software to
run." As Rodney Brooks pointed out many years ago, it is not just the artificial intelligence
software that defines the AI features of the platform, but rather the actual platform itself that
affects the AI that results, i.e, we need to be working out AI problems on real world platforms
rather than in isolation.
A wide variety of platforms has allowed different aspects of AI to develop, ranging from
expert systems, albeit PC-based but still an entire real-world system to various robot platforms
such as the widely available Roomba with open interface.
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Chapter 2
THE TECHNOLOGY
A human identity recognition system based on voice analysis could have seamless
applications. The ASR (Automatic Speaker Recognition) is one such system. Automatic Speaker
Recognition is a system that can recognize a person based on his/her voice. This is achieved by
implementing complex signal processing algorithms that run on a digital computer or a
processor.This Application is analogous to the fingerprint recognition system or other biometrics
recognition systems that are based on certain characteristics of a person.
There are several occasions when we want to identify a person from a given group of
people even when the person is not present for physical examination. For example, when a
person converses on a telephone, all we have is the person’s voice for analysis. It then makes
sense to develop a recognition system based on voice.
Speaker recognition has typically been classified as either a verification or identification

task. Speaker verification is usually the simpler of the two since it involves the comparison of
the input signal with a single given stored reference pattern. Therefore, the verification task only
requires a system to verify, if the speaker is the same as the person he/she identifies
himself/herself. Speaker identification is more complex because the test speaker must be
compared against a number of reference speakers to determine if a match can be made. Not only
the input signal is to be examined to see if it came from a speaker, but the identification of the
individual speaker is also necessary.
The identification of speakers remains a difficult task for a number of reasons. First, the
acquiring of a unique speech signal can suffer as a result of the variation of the voice inputs from
a speaker and environmental factors. Both the volume and pace of speech can vary from one test
to another. Also, unless initially constrained, an extensive vocabulary or unstructured grammar
can affect results. Background noise must also be kept to the minimum so that a changing
environment will not divert the speaker’s attention or the final voicing of a word or sentence. As
a result, many restrictions and clarifications have been placed on speaker and speech recognition
systems.
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One such restriction involves using a closed set for speaker recognition. A closed set
implies that only speakers within the original stored set will be asked to be identified. An open
set would allow the extra possibility of a test speaker not coming from the initially trained set of
speakers, thereby requiring the system to recognize the speaker as not belonging to the original
set. An open set system may also have the task to learning a new speaker and placing him or her
within the original set for future reference.
Another common restriction involves using a test dependent speaker recognition system.
This type of system would require the speaker to utter a unique word or phrase to be compared
against the original set of like phrases. Text-independent recognition, which for most cases is
more complex and difficult to perform, identifies the speaker regardless of the text or phrase
spoken.
Once an utterance, or signal, has been recorded, it is usually necessary to process it to get
the voiced signal in a form that makes classification and recognition possible. Various methods
have included the use of power spectrum values, spectrum coefficients, linear predictive coding,
and a nearest neighbour distance algorithm. Tests have also shown that although spectrum
coefficients and linear predictive coding have given better results for conventional template and
statistical classification methods, power spectrum values have performed better when using
neural networks during the final recognition stages.
Various methods have also been used to perform the classification and recognition of the
processed speech signal. Statistical methods utilizing Hidden Markov Models, linear vector
quantifiers, or classical techniques such as template matching have produced encouraging, yet
limited success. Recent deployments using neural networks, while producing varied success
rates, have offered more options regarding the types of inputs sent to the networks, as well as
provided the ability to learn speakers in both an off and online manner. Although back-
propagation networks have traditionally been used, the implementation of more sophisticated
networks, such as an ART 2 network, has been made.
ASR can be broadly classified into four types:
1. Text-independent identification
2. Text-independent verification
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3. Text-dependent identification
4. Text-dependent verification
Speaker identification is a procedure by which a speaker is identified from a group of ‘n’

people. It should be noted that a totally new speaker not belonging to the group could wrongly be
identified as someone from within the group.Speaker verification is a procedure by which a
speaker who claims his/her identity is verified as being correct or not.
A fundamental requirement for any ASR system is gathering reference samples and
finding certain features from the voice that are characteristic to a person. These feature vectors
are then stored. When a new test sample is made available, the references are either searched to
find the closest match (in case of identification), or a threshold of a distance measure is checked
(in case of verification).
The next aspect to the considered is text-dependency. In a text-independent situation, the

reference utterance and the test utterance are not the same. This type of recognition system finds
its applications in criminology. In a text-dependent situation, the reference utterance and the test
utterance are the same, which gives us a higher degree of accuracy. This type of recognition
system has applications where security is a matter of concern, such as access to a building to a
lab, to a computer, etc.
The dominant technology used in ASR is called the Hidden Markov Model, or HMM.
This technology recognizes speech by estimating the likelihood of each phoneme at contiguous,
small regions (frames) of the speech signal. Each word in a vocabulary list is specified in terms
of its component phonemes. A search procedure is used to determine the sequence of phonemes
with the highest likelihood. This search is constrained to only look for phoneme sequences that
correspond to words in the vocabulary list, and the phoneme sequence with the highest total
likelihood is identified with the word that was spoken. In standard HMMs, the likelihoods are
computed using a Gaussian Mixture Model; in the HMM/ANN framework, these values are
computed using an artificial neural network (ANN).
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Chapter 3
SPEECH RECOGNITION
The user speaks to the computer through a microphone, which in turn, identifies the
meaning of the words and sends it to NLP device for further processing. Once recognized, the
words can be used in a variety of applications like display, robotics,
commands to computers, and dictation.
The word recognizer is a speech recognition system that identifies individual words.
Early pioneering systems could recognize only individual alphabets and numbers. Today,
majority of word recognition systems are word recognizers and have more than 95% recognition
accuracy. Such systems are capable of recognizing a small vocabulary of single words or simple
phrases. One must speak the input information in clearly definable single words, with a pause
between words, in order to enter data in a computer. Continuous speech recognizers are far more
difficult to build than word recognizers. You speak complete sentences to the computer. The
input will be recognized and, then processed by NLP. Such recognizers employ sophisticated,
complex techniques to deal with continuous speech, because when one speaks continuously,
most of the words slur together and it is difficult for the system to know where one word ends
and the other
begins. Unlike word recognizers, the information spoken is not recognized instantly by this
system.
What is a speech recognition system?

A speech recognition system is a type of software that allows the user to have their
spoken words converted into written text in a computer application such as a word processor or
spreadsheet. The computer can also be controlled by the use of spoken commands.
Speech recognition software can be installed on a personal computer of appropriate

specification. The user speaks into a microphone (a headphone microphone is usually supplied
with the product). The software generally requires an initial training and enrolment process in
order to teach the software to recognize the voice of the user. A voice profile is then produced
that is unique to that individual. This procedure also helps the user to learn how to ‘speak’ to a
computer.
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3.1 Speech recognition process
DISPLAY
APPLICATION DICTATING
S
SPEECH
RECOGNITIO COMMANDS
USER N DEVICE TO COMPUTER
INPUT TO
OTHER
ROBOTS,
EXPERTS
NLP UNDERSTAND
VIOCE SOUND ING
DIALOGUE WITH USER Figure:3.1
After the training process, the user’s spoken words will produce text; the accuracy of this
will improve with further dictation and conscientious use of the correction procedure. With a
well-trained system, around 95% of the words spoken could be correctly interpreted. The system
can be trained to identify certain words and phrases and examine the user’s standard documents
in order to develop an accurate voice file for the individual.
However, there are many other factors that need to be considered in order to achieve a
high recognition rate. There is no doubt that the software works and can liberate many learners,
but the process can be far more time consuming than first time users may appreciate and the
results can often be poor. This can be very demotivating, and many users give up at this stage.
Quality support from someone who is able to show the user the most effective ways of using the
software is essential.
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When using speech recognition software, the user’s expectations and the advertising on
the box may well be far higher than what will realistically be achieved. ‘You talk and it types’
can be achieved by some people only after a great deal of perseverance and hard work.
3.2 Terms and Concepts
Following are a few of the basic terms and concepts that are fundamental to speech
recognition. It is important to have a good understanding of these concepts when developing
Voice XML applications.
3.2.1 Utterances
When the user says something, this is known as an utterance. An utterance is any stream
of speech between two periods of silence. Utterances are sent to the speech engine to be
processed. Silence, in speech recognition, is almost as important as what is spoken, because
silence delineates the start and end of an utterance. Here's how it works. The speech recognition
engine is "listening" for speech input. When the engine detects audio input - in other words, a
lack of silence -- the beginning of an utterance is signaled.
Similarly, when the engine detects a certain amount of silence following the audio, the
end of the utterance occurs. Utterances are sent to the speech engine to be processed. If the user
doesn’t say anything, the engine returns what is known as a silence timeout - an indication that
there was no speech detected within the expected timeframe, and the application takes an
appropriate action, such as reprompting the user for input. An utterance can be a single word, or
it can contain multiple words (a phrase or a sentence).
3.2.2 Pronunciations
The speech recognition engine uses all sorts of data, statistical models, and algorithms to
convert spoken input into text. One piece of information that the speech recognition engine uses
to process a word is its pronunciation, which represents what the speech engine thinks a word
should sound like. Words can have multiple pronunciations associated with them. For example,
the word “the” has at least two pronunciations in the U.S. English language: “thee” and “thuh.”
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As a Voice XML application developer, you may want to provide multiple pronunciations for
certain words and phrases to allow for variations in the ways your callers may speak them.
3.2.3 Grammars
As a Voice XML application developer, you must specify the words and phrases that
users can say to your application. These words and phrases are defined to the speech recognition
engine and are used in the recognition process. You can specify the valid words and phrases in a
number of different ways, but in Voice XML, you do this by specifying a grammar. A grammar
uses a particular syntax, or set of rules, to define the words and phrases that can be recognized by
the engine. A grammar can be as simple as a list of words, or it can be flexible enough to allow
such variability in what can be said that it approaches natural language capability.
3.2.4 Accuracy
The performance of a speech recognition system is measurable. Perhaps the most widely
used measurement is accuracy. It is typically a quantitative measurement and can be calculated
in several ways. Arguably the most important measurement of accuracy is whether the desired
end result occurred. This measurement is useful in validating application design Another
measurement of recognition accuracy is whether the engine
recognized the utterance exactly as spoken.
Another measurement of recognition accuracy is whether the engine recognized the

utterance exactly as spoken. This measure of recognition accuracy is expressed as a percentage
and represents the number of utterances recognized correctly out of the total number of
utterances spoken. It is a useful measurement when validating grammar design.
Recognition accuracy is an important measure for all speech recognition applications. It

is tied to grammar design and to the acoustic environment of the user. You need to measure the
recognition accuracy for your application, and may want to adjust your application and its
grammars based on the results obtained when you test your application with typical users.
18
Chapter 4
SPEAKER INDEPENDENCY
The speech quality varies from person to person. It is therefore difficult to build an
electronic system that recognizes everyone’s voice. By limiting the system to the voice of a
single person, the system becomes not only simpler but also more reliable. The computer must
be trained to the voice of that particular individual. Such a system is called Speaker-dependent
system.
Speaker-independent system can be used by anybody, and can recognize any voice, even
though the characteristics vary widely from one speaker to another. Most of these systems are
costly and complex. Also, these have very limited vocabularies. It is important to consider the
environment in which the speech recognition system has to work. The grammar used by the
speaker and accepted by the system, noise level, noise type, position of the microphone, and
speed and manner of the user’s speech are some factors that may affect the quality of the speech
recognition.
4.1 Speaker Dependence Vs Speaker Independence

Speaker Dependence describes the degree to which a speech recognition system requires
knowledge of a speaker’s individual voice characteristics to successfully process speech. The
speech recognition engine can “learn” how you speak words and phrases; it can be trained to
your voice.
Speech recognition systems that require a user to train the system to his/her voice are
known as speaker-dependent systems. If you are familiar with desktop dictation systems, most
are speaker dependent. Because they operate on very large vocabularies, dictation systems
perform much better when the speaker has spent the time to train the system to his/her voice.
Speech recognition systems that do not require a user to train the system are known as
speaker-independent systems. Speech recognition in the VoiceXML world must be speaker-
independent. Think of how many users (hundreds, maybe thousands) say be calling into your
web site. You cannot require that each caller train the system to his or her voice. The speech
recognition system in a voice-enabled web application MUST successfully process the speech of
19
many different callers without having to understand the individual voice characteristics of each
caller.
4.1.1 Advantages of speaker independent system
The advantage of a speaker independent system is obvious anyone can use the system
without first training it. However, its drawbacks are not so obvious. One limitation is the work
that goes into creating the vocabulary templates. To create reliable speaker independents
templates, someone must collect and process numerous speech sample. This is a time-consuming
task; creating these templates is not a one-time effort. Speaker-independent templates are
language-dependant, and the templates are sensitive not only to two dissimilar languages but also
to the differences between British and American English. Therefore, as part of your design
activity, you would need to create a set of templates for each language or a major dialect that
your customers use. Speaker independent systems also have a relatively fixed vocabulary
because of the difficulty in creating a new template in the field at the user’s site.
4.1.2 The advantage of a speaker-dependent system
A speaker dependent system requires the user a train the ASR system by providing
examples of his own speech. Training can be tedious process, but the system has the advantage
of using templates that refer only to the specific user and not some vague average voice. The
result is language independence. You can say ja, si, or ya during training, as long as you are
consistent. The drawback is that the speaker-dependent system must do more than simply match
incoming speech to the templates. It must also include resources to create those templates.
4.1.3 Which is better:
For a given amount of processing power, a speaker dependent system tends to provide
more accurate recognition than a speaker independent system. A speaker independent system is
not necessarily better: the difference in performance stems from the speaker independent
template encompassing wide speech variations.
20
4.2 System Configuration

Figures 4.2.1 and 4.2.2 show the identification system and the verification system
configuration, respectively .The first part of the system consists of the data acquisition hardware
that acquires the speech, performs some signal conditioning, digitizes it and gives it to the
computer/processor.
The second part consists of core signal processing and system identification techniques to
extract speaker specific features. These features are stored and are used at a later time for the
actual identification/verification test. At this stage, the system is ready for identification or
verification.
Now, when the test sample is uttered by one of the members of the group, the speech is
digitized and the features are extracted. For identification, distances between this vector and all
the reference vectors are measured and the closest vector is picked up as the correct one. This
vector would correspond to a person whom the system claims as having been identified. For
verification, the person claims his/her identity. The distance between the corresponding reference
vector and the test vector is the computed. If the measured distance is less than a set threshold,
the verification system accepts the speaker; if not, it rejects the speaker.
MEASUREMENT REFERENCE
ADC WITH ALGORITHM OF DISTANCE VECTORS
VOICE SIGNAL TO SELECT
SAMPLE CONDITIONING FEATURES DECISION IDENTITY OF
MAKING PERSON
Figure 4.1: Speaker Identification
21
VOICE
SAMPLE ADC WITH ALGORITHM MEASUREMENT REFERENCE
(PERSON SIGNAL
TO SELECT OF DISTANCE VECTOR OF
CLAIMS CONDITIONIN
HIS/HER G FEATURES THE
IDENTITY SPEAKER
THRESHOLD
COMPARISON PERSON
VERIFIED OR
NOT
Figure 4.2: Speaker Verification
The voice input to the microphone produces an analogue speech signal. An analogue-to-
digital converter (ADC) converts this speech signal into binary words that are compatible with
digital computer. The converted binary version is then stored in the system and compared with
previously stored binary representations of words and phrases. The current input speech is
compared one at a time with the previously stored speech pattern after searching by the
computer. When a match occurs, recognition is achieved. The spoken word is binary form is
written on a video screen or passed along to a natural language understanding processor for
additional analysis.
Since most recognition systems are speaker-dependent, it is necessary to train a system to

recognize the dialect of each new user. During training, the computer displays a word and the
user reads it aloud. The computer digitizes the user’s voice and stores it. The speaker has to read
aloud about 1,000 words. Based on these samples, the computer can predict how the user utters
some words that are likely to be pronounced differently by different people.
The block diagram of a speaker-dependent word recognizer is shown in Fig. 4.2.1 The
user speaks before the microphone, which converts the sound into electrical signal. The electrical
analogue signal from microphone is fed to an amplifier provided with automatic gain control
(AGC) to produce an amplified output signal in a specific optimum voltage range, even when the
input signal varies from feeble to loud.
22
The analogue signal, representing a spoken word, contains many individual frequencies
of various amplitudes and different phases, which when blended together take the shape of a
complex waveform . A set of filters is used to break this complex input signal into its component
parts. Band pass filters (BEP) pass on frequencies only in certain frequency range, rejecting all
other frequencies. Generally, about sixteen filters are used; a simple system may contain a
minimum of three filters. The more the number of filters user, the higher the probability of
accurate recognition.
Presently, switched capacitor digital filters are used because these can be custom-built in
integrated circuit form. These are smaller and cheaper than active filters using operational
amplifiers. The filter output is then fed to the ADC to translate the analogue signal into digital
word. The ADC samples the filter outputs many times a second. Each sample represents a
different amplitude of the signal .
Evenly spaced vertical lines represent the amplitude of the audio filter output at the
instant of sampling. Each value is then converted to a binary number proportional to the
amplitude of the sample. A central processor unit controls the input circuits that are fed by the
ADCs. A large RAM stores all the digital values in a buffer area. This digital information,
representing the spoken word, is now accessed by the CPU to process it further. The normal
speech has a frequency range of 200 Hz to 7 kHz. Recognizing a telephone call is more difficult
as it has bandwidth limitation of 300Hz to 3.3 Hz.
As explained earlier, the spoken words are processed by the filters and ADCs. The binary
representation of each of these words becomes a template or standard, against which the future
words are compared. These templates are stored in the memory. Once the storing process is
completed, the system can go into its active mode and is capable of identifying spoken words. As
each word is spoken, it is converted into binary equivalent and stored in RAM. The computer
then starts searching and compares the binary input pattern with the templates.
It is to be noted that even if the same speaker talks the same text, there are always slight
variations in amplitude or loudness of the signal, pitch, frequency difference, time gap, etc. Due
to this reason, there is never a perfect match between the template and binary input word. The
23
pattern matching process therefore uses statistical techniques and is designed to look for the best
fit.
The values of binary input words are subtracted from the corresponding values, in the
templates. If both the values are same, the difference is zero and there is perfect match. If not,
the subtraction produces some difference or error. The smaller the error, the better the match.
When the best match occurs the word is identified and displayed on the screen or used in some
other manner.
The search process takes a considerable amount of time as the CPU has to make many
comparisons before recognition occurs. This necessitates use of very high-speed processors. A
large RAM is also required as even though a spoken word may last only a few hundred
milliseconds, but the same is translated into many thousands of digital words. It is important to
not e that alignment of words and templates are to be matched correctly in time, before
computing the similarity score. This process, termed as dynamic time warping, recognizes that
different speaker pronounce the same words at different speeds as well as elongate different parts
of the same word. This is important for the speaker-independent recognizers.
Continuous speech recognizers are far more difficult to build than word recognizers. You can
speak complete sentences to the computer. The input will be recognized and, when processed by
NLP, understood. Such recognizers employ sophisticated, complex techniques to deal with
continuous speech, because when one speaks continuously, most of the words slur together and it
is difficult for the system to know where one word ends and the other begins. Unlike word
recognizers, the information spoken is not recognized instantly by this system.
24
Chapter 5
WORKING OF THE SYSTEM

The voice input to the microphone produces an analogue speech signal. An analogue to
digital converter (ADC) converts this speech signal into binary words that are compatible with
digital computer. The converted binary version is then stored in the system and compared with
previously stored binary representation of words and phrases. The current input speech is
compared one at a time with the previously stored speech pattern after searching by the
computer. When a match occurs, recognition is achieved. The spoken word in binary form is
written on a video screen or passed along to a natural language understanding processor for
additional analysis. Since most recognition systems are speaker-dependent, it is necessary to
train a system to recognize the dialect of each new user. During training, the computer displays a
word and user reads it aloud. The computer digitizes the user’s voice and stores it. The speaker
has to read aloud about 1000 words. Based on these samples, the computer can predict how the
user utters some words that are likely to be pronounced differently by different users.
The block diagram of a speaker- dependent word recognizer is shown in figure. The user
speaks before the microphone, which converts the sound into electrical signal. The electrical
analogue signal from the microphone, is fed to an amplifier provided with automatic gain control
(AGC) to produce an amplified output signal in a specific optimum voltage range, even when the
input signal varies from feeble to loud.
The analogue signal, representing a spoken word, contains many individual frequencies
of various amplitudes and different phases, which when blended together take the shape of a
complex wave form as shown in figure. A set of filters is used to break this complex signal into
its component parts. Band pass filters (BFP) pass on frequencies only on certain frequency range,
rejecting all other frequencies. Generally, about 16 filters are used; a simple system may contain
a minimum of three filters. The more number of filters used, the higher the probability of
accurate recognition. Presently, switched capacitor digital filters are used because these can be
custom- built in integrated circuit form. These are smaller and cheaper than active filters using
operational amplifiers. The filter output is then fed to the ADC to translate the analog signal into
digital word. The ADC samples the filter output many times a second. Each sample represents
different amplitude of the signal .A CPU controls the input circuits that are fed by the ADC’s. A
25
large RAM stores all the digital values in a buffer area. This digital information, representing the
spoken word, is now accessed by the CPU to process it further.
5.1 Speaker- dependent word recognizer
I RAM
N
BPF ADC P DIGITISED
U SPEECH
T
C
I
R
C
U
I
T
S
Figure 5.1
The normal speech has a frequency range of 200 Hz to 7KHz. Recognizing a telephone
call is more difficult as it has bandwidth limitations of 300Hz to 3.3KHz.As explained earlier the
spoken words are processed by the filters and ADCs. The binary representation of each of these
word becomes a template or standard against which the future words are compared. These
templates are stored in the memory. Once the storing process is completed, the system can go
into its active mode and is capable of identifying the spoken words. As each word is spoken, it is
converted into binary equivalent and stored in RAM. The computer then starts searching and
compares the binary input pattern with the templates. It is to be noted that even if the same
speaker talks the same text, there are always slight variations in amplitude or loudness of the
signal, pitch, frequency difference, time gap etc. Due to this reason there is never a perfect match
between the template and the binary input word. The pattern matching process therefore uses
statistical techniques and is designed to look for the best fit.
The values of binary input words are subtracted from the corresponding values in the
templates. If both the values are same, the difference is zero and there is perfect match. If not, the
26
subtraction produces some difference or error. The smaller the error, the better the match. When
the best match occurs, the word templates are to be matched correctly in time, before computing
the similarity score. This process, termed as dynamic time warping recognizes that different
speakers pronounce the same word at different is identified and displayed on the screen or used
in some other manner.
The search process takes a considerable amount of time, as the CPU has to make many
comparisons before recognition occurs. This necessitates use of very high-speed processors. A
Large RAM is also required as even though a spoken word may last only a few hundred
milliseconds, but the same is translated into many thousands of digital words. It is important to
note that alignment of words and speeds as well as elongate different parts of the same word.
This is important for the speaker- independent recognizers.
Now that we've discussed some of the basic terms and concepts involved in speech
recognition, let's put them together and take a look at how the speech recognition process works.
As you can probably imagine, the speech recognition engine has a rather complex task to handle,
that of taking raw audio input and translating it to recognized text that an application
understands. The major components discussed are:
• Audio input
• Grammar(s)
• Acoustic Model
• Recognized text
The first thing we want to take a look at is the audio input coming into the recognition
engine. It is important to understand that this audio stream is rarely pristine. It contains not only
the speech data (what was said) but also background noise. This noise can interfere with the
recognition process, and the speech engine must handle (and possibly even adapt to) the
environment within which the audio is spoken. As we've discussed, it is the job of the speech
recognition engine to convert spoken input into text. To do this, it employs all sorts of data,
statistics, and software algorithms. Its first job is to process the incoming audio signal and
convert it into a format best suited for further analysis.
Once the speech data is in the proper format, the engine searches for the best match. It
does this by taking into consideration the words and phrases it knows about (the active
27
grammars), along with its knowledge of the environment in which it is operating for Voice
XML, this is the telephony environment). The knowledge of the environment is provided in the
form of an acoustic model. Once it identifies the the most likely match or what was said, it
returns what it recognized as a text string. Most speech engines try very hard to find a match, and
are usually very "forgiving." But it is important to note that the engine is always returning it's
best guess
for what was said.
5.1.1 Acceptance and Rejection
When the recognition engine processes an utterance, it returns a result. The result can be
either of two states: acceptance or rejection. An accepted utterance is one in which the engine
returns recognized text. Whatever the caller says, the speech recognition engine tries very hard to
match the utterance to a word or phrase in the active grammar. Sometimes the match may be
poor because the caller said something that the application was not expecting, or the caller spoke
indistinctly. In these cases, the speech engine returns the closest match, which might be
incorrect. Some engines also return a confidence score along with the text to indicate the
likelihood that the returned text is correct. Not all utterances that are processed by the speech
engine are accepted. Acceptance or rejection is flagged by the engine with each processed
utterance.
5.2 What software is available?
There are a number of publishers of speech recognition software. New and improved
versions are regularly produced, and older versions are often sold at greatly reduced prices.
Invariably, the newest versions require the most modern computers of well above average
specification. Using the software on a computer with a lower specification means that it will run
very slowly and may well be impossible to use. There are two main types of speech recognition
software: discrete speech and continuous speech.
Discrete speech software is an older technology that requires the user to speak one word
at a time . Dragon Dictate Classic Version 3 is one example of discrete speech software, as it has
fewer features, is simple to train and use and will work on Continuous speech software allows
the user to dictate normally. In fact, it works best when it hears complete sentences, as it
28
interprets with more accuracy when it recognizes the context. The delivery can be varied by
using short phrases and single words, following the natural pattern of speech.
5.3 What technical issues need to be considered when purchasing

this system?
The latest versions of speech recognition software (September 2001) require a Pentium 3
processor and 256 MB of memory. Currently, Dragon Naturally Speaking Version 4 and IBM
Via Voice Millennium edition have been used in school settings. Very good results can be
obtained with these on fast, high-memory machines. When purchasing a machine, it is worth
mentioning to the supplier that it will be required for running speech recognition software.
Whether choosing a desktop or portable computer, it will also require a good quality
duplex (input and output) sound card. Poor sound quality will reduce the recognition accuracy.
The microphones supplied with the software may be perfectly adequate, but better results can
often be obtained by using a noise-cancelling microphone. In addition, mobile voice recorders
allow a number of users to produce dictation that can be downloaded to the main speech
recognition system, but be aware of some of the complexities of their use.
5.4 How does the technology differ from other technologies?

Speech recognition systems produce written text from the user’s dictation, without using,
or with only minimal use of, a traditional keyboard and mouse. This is an obvious benefit to
many people who, for any number of reasons, do not find it easy to use a keyboard, or whose
spelling and literacy skills would benefit from seeing accurate text.
The limitations to this type of software are that:
 It needs to be completely tailored to the user and trained by the user.

 It is often set up on one machine, and so can create difficulties for a user who
works from many locations, for example from school and home.
 It depends on the user having the desire to produce text and be able to invest the
time, training and perseverance necessary to achieve it.
 It is most successful for those competent in the art of dictation.
29
A speech recognition system is a powerful application in that the software’s recognition

of the user’s voice pattern and vocabulary improves with use. A useful tip is to ensure that voice
files can be backed up regularly.
5.5 What factors need to be considered when using speech

recognition technology?
The Becta SEN Speech Recognition Project describes the key factors to success
as ‘The Three Ts’: Time, Technology and Training:
Time
Take time to choose the most appropriate software and hardware and match it to the user.
One option for new users is to start with discrete speech software. The skills learned whilst using
it can be transferred to more sophisticated speech recognition software. If the new user is unable
to make effective use of discrete speech recognition software, then it is unlikely they will
succeed with continuous speech software. Familiarisation with the product and frequent breaks
between talking are also helpful.older computers.
Training
With speech recognition systems, both the software and the user require training.
Patience and practice are required. The user needs to take things slowly, practicing putting their
thoughts into words before attempting to use the system.
Technology
The best results are generally achieved using a high-specification machine. Sound cards
and microphones are a key feature for success, as is access to technical support and advice.
30
Chapter 6
THE LIMITS OF SPEECH RECOGNITION

To improve speech recognition applications, designers must understand acoustic memory
and prosody. Continued research and development should be able to improve certain speech
input, output, and dialogue applications. Speech recognition and generation is sometimes helpful
for environments that are hands-busy, eyes-busy, mobility required, or hostile and shows
promise for telephone-based ser-vices. Dictation input is increasingly accurate, but adoption
outside the disabled-user community has been slow compared to visual interfaces. Obvious
physical problems include fatigue from speaking continuously and the disruption in an office
filled with people speaking.
By understanding the cognitive processes surrounding human “acoustic memory” and
processing, interface designers may be able to integrate speech more effectively and guide users
more successfully. By appreciating the differences between human-human interaction and
human-computer interaction, designers may then be able to choose appropriate applications for
human use of speech with computers. The key distinction may be the rich emotional content
conveyed by prosody, or the pacing, intonation, and amplitude in spoken language. The emotive
aspects of prosody are potent for human interaction but may be disruptive for human-computer
interaction. The syntactic aspects of prosody, such as rising tone for questions, are important for
a system’s recognition and generation of sentences.
Now consider human acoustic memory and processing. Short-term and working Memory
are some-times called acoustic or verbal mems the human brain that transiently holds chunks of
information and solves problems also supports speaking and listening. Therefore, working on
tough problems is best done in quiet environments without speaking or listening to someone.
However, because physical activity is handled in another part of the brain, problem solving is
compatible with routine physical activities like walking and driving. In short, humans speak and
walk easily but find it more difficult to speak and think at the same time .Similarly when
operating a computer, most humans type (or move a mouse) and think but find it more difficult
to speak and think at the same time. Hand-eye coordination is accomplished in different brain
structures, so typing or mouse movement can be performed in parallel with problem solving.
31
Product evaluators of an IBM dictation software the human brain that transiently holds
chunks of information and solves problems also supports speaking and listening. Therefore,
working on tough problems is best done in quiet environments—without speaking or listening to
someone. However, because physical activity is handled in another part of the brain, problem
solving is compatible with routine physical activities like walking and driving. In short, humans
speak and walk easily but find it more difficult to speak and think at the same time .
Similarly when operating a computer, most humans type (or move a mouse) and think but
find it more difficult to speak and think at the same time. Hand-eye coordination is accomplished
in different brain structures, so typing or mouse movement can be performed in parallel with
problem solving. Product evaluators of an IBM dictation software package also noticed this
phenomenon. They wrote that “thought for many people is very closely linked to language. In
keyboarding, users can continue to hone their words while their fingers output an earlier version.
In dictation, users may experience more interference between outputting their initial thought and
elaborating on it.” Developers of commercial speech recognition software packages recognize
this problem and often advise dictation of full paragraphs or documents, followed by a review or
proofreading phase to correct errors. Since speaking consumes precious cognitive resources, it is
difficult to solve problems at the same time. Proficient keyboard users can have higher levels of
parallelism in problem solving while performing data entry. This may explain why after 30 years
of ambitious attempts to provide military pilots with speech recognition in cockpits, aircraft
designers persist in using hand-input devices and visual displays. Complex functionality is built
In to the pilot’s joy-stick, which has up to 17 functions, including pitch-roll- yaw controls, plus a
rich set of buttons and triggers. Similarly automobile controls may have turn signals, wiper
settings, and washer buttons all built onto a single stick, and typical video camera controls may
have dozens of settings that are adjustable through knobs and switches. Rich designs for hand
input can inform users and free their minds for status monitoring and problem solving.
The interfering effects of acoustic processing are a limiting factor for designers of speech
recognition, but the the role of emotive prosody raises further con-cerns. The human voice has
evolved remarkably well to support human-human interaction. We admire and are inspired by
passionate speeches. We are moved by grief-choked eulogies and touched by a child’s calls as
we leave for work. A military commander may bark commands at troops, but there is as much
motivational force in the tone as there is information in the words. Loudly barking commands at
a computer is not likely to force it to shorten its response time or retract a dialogue box.
32
Promoters of “affective” computing, or reorganizing, responding to, and making emotional

displays, may recommend such strategies, though this approach seems misguided.
Many users might want shorter response times without having to work them-selves into a
mood of impatience. Secondly, the logic of computing requires a user response to a dialogue box
independent of the user’s mood. And thirdly, the uncertainty of machine recognition
could undermine the positive effects of user control and interface predictability.
33
Chapter 7
APPLICATION
One of the main benefits of speech recognition system is that it lets user do other works
simultaneously. The user can concentrate on observation and manual operations, and still control
the machinery by voice input commands. Consider a material-handling plant where a number of
conveyors are employed to transport various grades of materials to different destinations.
Nowadays, only one operator is employed to run the plant. He has to keep a watch on various
meters, gauges, indication lights, analyzers, overload devices, etc from the central control panel.
If something wrong happens, he has to run to physically push the ‘stop’ button. How convenient
it would be if a conveyor or a number of conveyors are stopped automatically
by simply saying stop.
Another major application of speech processing is in military operations. Voice control of
weapons is an example. With reliable speech recognition equipment, pilots can give commands
and information to the computers by simply speaking in to their microphones-they don’t have to
use their hands for this purpose .Another good example is a radiologist scanning hundreds of
Xrays, ultra sonograms, CT scans and simultaneously dictating conclusion to a speech
recognition system connected to word processors. The radiologist can focus his attention on the
images rather than writing the text. Voice recognition could also be used on computers for
making airline and hotel reservations. A user requires simply to state his needs, to make
reservation, cancel a reservation, or make enquiries about schedule. sitive effects of user control
and interface predictability.
7.1 Health care
In the health care domain, even in the wake of improving speech recognition
technologies, medical transcriptionists (MTs) have not yet become obsolete. Many experts in the
field anticipate that with increased use of speech recognition technology, the services provided
may be redistributed rather than replaced. Speech recognition is used to enable deaf people to
understand the spoken word via speech to text conversion, which is very helpful.
Speech recognition can be implemented in front-end or back-end of the medical

documentation process.
34
 Front-End SR is where the provider dictates into a speech-recognition engine, the

recognized words are displayed right after they are spoken, and the dictator is
responsible for editing and signing off on the document. It never goes through an
MT/editor.
 Back-End SR or Deferred SR is where the provider dictates into a digital dictation

system, and the voice is routed through a speech-recognition machine and the
recognized draft document is routed along with the original voice file to the
MT/editor, who edits the draft and finalizes the report. Deferred SR is being widely
used in the industry currently.
Many Electronic Medical Records (EMR) applications can be more effective and may be
performed more easily when deployed in conjunction with a speech-recognition engine.
Searches, queries, and form filling may all be faster to perform by voice than by using a
keyboard.
7.2 Military
7.2.1 High-performance fighter aircraft
Substantial efforts have been devoted in the last decade to the test and evaluation of
speech recognition in fighter aircraft. Of particular note are the U.S. program in speech
recognition for the Advanced Fighter Technology Integration (AFTI)/F-16 aircraft (F-16
VISTA), the program in France on installing speech recognition systems on Mirage aircraft, and
programs in the UK dealing with a variety of aircraft platforms. In these programs, speech
recognizers have been operated successfully in fighter aircraft with applications including:
setting radio frequencies, commanding an autopilot system, setting steer-point coordinates and
weapons release parameters, and controlling flight displays. Generally, only very limited,
constrained vocabularies have been used successfully, and a major effort has been devoted to
integration of the speech recognizer with the avionics system.
Some important conclusions from the work were as follows:
1. Speech recognition has definite potential for reducing pilot workload, but this potential
was not realized consistently.
35
2. Achievement of very high recognition accuracy (95% or more) was the most critical
factor for making the speech recognition system useful with lower recognition rates,
pilots would not use the system.
3. More natural vocabulary and grammar, and shorter training times would be useful, but
only if very high recognition rates could be maintained.
Laboratory research in robust speech recognition for military environments has produced
promising results which, if extendable to the cockpit, should improve the utility of speech
recognition in high-performance aircraft.
Working with Swedish pilots flying in the JAS-39 Gripen cockpit, Englund (2004) found
recognition deteriorated with increasing G-loads. It was also concluded that adaptation greatly
improved the results in all cases and introducing models for breathing was shown to improve
recognition scores significantly. Contrary to what might be expected, no effects of the broken
English of the speakers were found. It was evident that spontaneous speech caused problems for
the recognizer, as could be expected. A restricted vocabulary, and above all, a proper syntax,
could thus be expected to improve recognition accuracy substantially.
The Eurofighter Typhoon currently in service with the UK RAF employs a speaker-
dependent system, i.e. it requires each pilot to create a template. The system is not used for any
safety critical or weapon critical tasks, such as weapon release or lowering of the undercarriage,
but is used for a wide range of other cockpit functions. Voice commands are confirmed by visual
and/or aural feedback. The system is seen as a major design feature in the reduction of pilot
workload, and even allows the pilot to assign targets to himself with two simple voice commands
or to any of his wingmen with only five commands.
7.2.2 Helicopters
The problems of achieving high recognition accuracy under stress and noise pertain
strongly to the helicopter environment as well as to the fighter environment. The acoustic noise
problem is actually more severe in the helicopter environment, not only because of the high noise
levels but also because the helicopter pilot generally does not wear a facemask, which would
reduce acoustic noise in the microphone. Substantial test and evaluation programs have been
carried out in the past decade in speech recognition systems applications in helicopters, notably
36
by the U.S. Army Avionics Research and Development Activity (AVRADA) and by the Royal
Aerospace Establishment (RAE) in the UK. Work in France has included speech recognition in
the Puma helicopter. There has also been much useful work in Canada. Results have been
encouraging, and voice applications have included: control of communication radios; setting of
navigation systems; and control of an automated target handover system.
As in fighter applications, the overriding issue for voice in helicopters is the impact on
pilot effectiveness. Encouraging results are reported for the AVRADA tests, although these
represent only a feasibility demonstration in a test environment. Much remains to be done both in
speech recognition and in overall speech recognition technology, in order to consistently achieve
performance improvements in operational settings.
7.2.3 Battle management
Battle Management command centres generally require rapid access to and control of
large, rapidly changing information databases. Commanders and system operators need to query
these databases as conveniently as possible, in an eyes-busy environment where much of the
information is presented in a display format. Human-machine interaction by voice has the
potential to be very useful in these environments. A number of efforts have been undertaken to
interface commercially available isolated-word recognizers into battle management
environments. In one feasibility study speech recognition equipment was tested in conjunction
with an integrated information display for naval battle management applications. Users were
very optimistic about the potential of the system, although capabilities were limited.
Speech understanding programs sponsored by the Defense Advanced Research Projects

Agency (DARPA) in the U.S. has focused on this problem of natural speech interface. Speech
recognition efforts have focused on a database of continuous speech recognition (CSR), large-
vocabulary speech which is designed to be representative of the naval resource management task.
Significant advances in the state-of-the-art in CSR have been achieved, and current efforts are
focused on integrating speech recognition and natural language processing to allow spoken
language interaction with a naval resource management system.
37
7.3 Training air traffic controllers
Training for military (or civilian) air traffic controllers (ATC) represents an excellent
application for speech recognition systems. Many ATC training systems currently require a
person to act as a "pseudo-pilot", engaging in a voice dialog with the trainee controller, which
simulates the dialog which the controller would have to conduct with pilots in a real ATC
situation. Speech recognition and synthesis techniques offer the potential to eliminate the need
for a person to act as pseudo-pilot, thus reducing training and support personnel. Air controller
tasks are also characterized by highly structured speech as the primary output of the controller,
hence reducing the difficulty of the speech recognition task.
The U.S. Naval Training Equipment Center has sponsored a number of developments of
prototype ATC trainers using speech recognition. Generally, the recognition accuracy falls short
of providing graceful interaction between the trainee and the system. However, the prototype
training systems have demonstrated a significant potential for voice interaction in these systems,
and in other training applications. The U.S. Navy has sponsored a large-scale effort in ATC
training systems, where a commercial speech recognition unit was integrated with a complex
training system including displays and scenario creation. Although the recognizer was
constrained in vocabulary, one of the goals of the training programs was to teach the controllers
to speak in a constrained language, using specific vocabulary specifically designed for the ATC
task. Research in France has focused on the application of speech recognition in ATC training
systems, directed at issues both in speech recognition and in application of task-domain grammar
constraints.[4]
The USAF, USMC, US Army, and FAA are currently using ATC simulators with speech
recognition from a number of different vendors, including UFA, Inc, and Adacel Systems Inc
(ASI). This software uses speech recognition and synthetic speech to enable the trainee to control
aircraft and ground vehicles in the simulation without the need for pseudo pilots.
Another approach to ATC simulation with speech recognition has been created by
Supremis. The Supremis system is not constrained by rigid grammars imposed by the underlying
limitations of other recognition strategies.
38
7.4 Telephony and other domains
ASR in the field of telephony is now commonplace and in the field of computer gaming
and simulation is becoming more widespread. Despite the high level of integration with word
processing in general personal computing, however, ASR in the field of document production
has not seen the expected increases in use.
The improvement of mobile processor speeds made feasible the speech-enabled Symbian
and Windows Mobile Smartphones. Speech is used mostly as a part of User Interface, for
creating pre-defined or custom speech commands. Leading software vendors in this field are:
Microsoft Corporation (Microsoft Voice Command), Nuance Communications (Nuance Voice
Control), Vito Technology (VITO Voice2Go), Speereo Software (Speereo Voice Translator) and
SVOX.
7.5 People with disabilities
People with disabilities can benefit from speech recognition programs. Speech
recognition is especially useful for people who have difficulty using their hands, ranging from
mild repetitive stress injuries to involved disabilities that preclude using conventional computer
input devices. In fact, people who used the keyboard a lot and developed RSI became an urgent
early market for speech recognition. Speech recognition is used in deaf telephony, such as
voicemail to text, relay services, and captioned telephone. Individuals with learning disabilities
who have problems with thought-to-paper communication (essentially they think of an idea but it
is processed incorrectly causing it to end up differently on paper) can benefit from the software.
7.6 Hands-free computing
Hands-free computing is a term used to describe a configuration of computers so that

they can be used by persons without the use of the hands interfacing with commonly used human
interface devices such as the mouse and keyboard. Hands-free computing is important because it
is useful to both able and disabled users. Speech recognition systems can be trained to recognize
specific commands and upon confirmation of correctness instructions can be given to systems
without the use of hands. This may be useful while driving or to an inspector or engineer in a
39
factory environment. Likewise disabled persons may find hands-free computing important in
their everyday lives. Just like visually impaired have found computers useful in their lives.
40
8 . CONCLUSION
Speech recognition will revolutionize the way people conduct business over the Web and
will, ultimately, differentiate world-class e-businesses. Voice XML ties speech recognition and
telephony together and provides the technology with which businesses can develop and deploy
voice-enabled Web solutions TODAY! These solutions can greatly expand the accessibility of
Web-based self-service transactions to customers who would otherwise not have access, and, at
the same time, leverage a business’ existing Web investments. Speech recognition and Voice
XML clearly represent the next wave of the Web. It is important to consider the environment in
which the speech system has to work. The grammar used by the speaker and accepted by the
system, noise level, noise type, position of the microphone, and speed and manner of the user’s
speech are some factors that may affect the quality of speech recognition. Since, most
recognition systems are speaker independent, it is necessary to train a system to recognize the
dialect of each user. During training, the computers display a word and the user reads it aloud.
41
9. BIBLOGRAPHY
1. Luger, George; Stubblefield, William (2004). Artificial Intelligence: Structures
and Strategies for Complex Problem Solving (5th ed.). The
Benjamin/Cummings Publishing Company, Inc.. ISBN 0-8053-4780-1.
http://www.cs.unm.edu/~luger/ai-final/tocfull.html.
2. Nilsson, Nils (1998). Artificial Intelligence: A New Synthesis. Morgan Kaufmann
Publishers. ISBN 978-1-55860-467-4.
3. Russell, Stuart J.; Norvig, Peter (2003), Artificial Intelligence: A Modern
Approach (2nd ed.), Upper Saddle River, New Jersey: Prentice Hall, ISBN 0-13-
790395-2, http://aima.cs.berkeley.edu/
4. Poole, David; Mackworth, Alan; Goebel, Randy (1998). Computational
Intelligence: A Logical Approach. New York: Oxford University Press.
http://www.cs.ubc.ca/spider/poole/ci.html.
5. Winston, Patrick Henry (1984). Artificial Intelligence. Reading, Massachusetts:
Addison-Wesley. ISBN 0201082594.
6. www.seminaron.com
7. www.scribd.com
8. www.seminaronly.com
9. http://www.becta.org.uk
10. http://www.edc.org
11. http://www.dyslexic.com
12. http://www.ibm.com
13. http://www.dragonsys.com
14. http://www.out-loud.com
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ARTIFICIAL

7.3 Training air traffic controllers 38

1.1.1 Deduction, reasoning, problem solving

1.1.2 Knowledge representation

Knowledge representation and knowledge engineering are central to AI research. Many

1.1.2.1 Among the most difficult problems in knowledge representation are:

 Default reasoning and the qualification problem

 The breadth of commonsense knowledge

 The sub symbolic form of some commonsense knowledge

1.1.5 Natural language processing

1.1.6 Motion and manipulation

1.1.8 Social intelligence

Figure : 1.3 Kismet, a robot with rudimentary social skills

A sub-field of AI addresses creativity both theoretically (from a philosophical and

1.2.1 Search and optimization

Many problems in AI can be solved in theory by intelligently searching through many

1.2.3 Probabilistic methods for uncertain reasoning

1.2.4 Classifiers and statistical learning methods

1.2.5 Neural networks

1.2.6 Control theory

Control theory, the grandchild of cybernetics, has many important applications,

AI researchers have developed several specialized languages for AI research, including

1.3.1Competitions and prizes

There are a number of competitions and prizes to promote research in artificial

A platform (or "computing platform")is defined by Wikipedia as "some sort of hardware

Speaker recognition has typically been classified as either a verification or identification

ASR can be broadly classified into four types:

Speaker identification is a procedure by which a speaker is identified from a group of ‘n’

The next aspect to the considered is text-dependency. In a text-independent situation, the

What is a speech recognition system?

Speech recognition software can be installed on a personal computer of appropriate

3.1 Speech recognition process

DIALOGUE WITH USER Figure:3.1

3.2 Terms and Concepts

Another measurement of recognition accuracy is whether the engine recognized the

Recognition accuracy is an important measure for all speech recognition applications. It

4.1 Speaker Dependence Vs Speaker Independence

4.1.1 Advantages of speaker independent system

4.1.2 The advantage of a speaker-dependent system

4.1.3 Which is better:

4.2 System Configuration

Figure 4.1: Speaker Identification

Figure 4.2: Speaker Verification

Since most recognition systems are speaker-dependent, it is necessary to train a system to

WORKING OF THE SYSTEM

5.1 Speaker- dependent word recognizer

5.1.1 Acceptance and Rejection

5.2 What software is available?

5.3 What technical issues need to be considered when purchasing

5.4 How does the technology differ from other technologies?

The limitations to this type of software are that:

 It needs to be completely tailored to the user and trained by the user.

A speech recognition system is a powerful application in that the software’s recognition

5.5 What factors need to be considered when using speech

THE LIMITS OF SPEECH RECOGNITION

Promoters of “affective” computing, or reorganizing, responding to, and making emotional

7.1 Health care

Speech recognition can be implemented in front-end or back-end of the medical

 Front-End SR is where the provider dictates into a speech-recognition engine, the

 Back-End SR or Deferred SR is where the provider dictates into a digital dictation

7.2.1 High-performance fighter aircraft

Some important conclusions from the work were as follows:

7.2.3 Battle management