This paper presents a design of voice enabled examination system which can be used by the visually challenged students.
The system uses Text-to-Speech (TTS) and Speech-to-Text (STT) technology. The text-to-speech and speech-to-text web based
academic testing software would provide an interaction for blind students to enhance their educational experiences by providing them
with a tool to give the exams. This system will aid the differently-abled to appear for online tests and enable them to come at par with
the other students. This system can also be used by students with learning disabilities or by people who wish to take the examination in
a combined auditory and visual way.
Industrial Applications of Automatic Speech Recognition SystemsIJERA Editor
Current trends in developing technologies form important bridges to the future, fortified by the early and
productive use of technology for enriching the human life. Speech signal processing, which includes automatic
speech recognition, synthetic speech, and natural language processing, is beginning to have a significant impact
on business, industry and ease of operation of personal computers. Apart from this, it facilitates the deeper
understanding of complex mechanism of functioning of human brain. Advances in speech recognition
technology, over the past five decades, have enabled a wide range of industrial applications. Yet today's
applications provide a small preview of a rich future for speech and voice interface technology that will
eventually replace keyboards with microphones for designing human machine interface for providing easy
access to increasingly intelligent machines. It also shows how the capabilities of speech recognition systems in
industrial applications are evolving over time to usher in the next generation of voice-enabled services. This
paper aims to present an effective survey of the speech recognition technology described in the available
literature and integrate the insights gained during the process of study of individual research and developments.
The current applications of speech recognition for real world and industry have also been outlined with special
reference to applications in the areas of medical, industrial robotics, forensic, defence and aviation
Application for Iraqi sign language translation on Android system IJECEIAES
Deaf people suffer from difficulty in social communication, especially those who have been denied the blessing of hearing before the acquisition of spoken language and before learning to read and write. For the purpose of employing mobile devices for the benefit of these people, their teachers and everyone who has contact with them, this research aims to design an application for social communication and learning by translating Iraqi sign language into text in Arabic and vice versa. Iraqi sign language has been chosen because of a lack of applications for this field. The current research, to the best of our knowledge, is the first of its kind in Iraq. The application is open source; words that are not found in the application database can be processed by translating them into letters alphabetically. The importance of the application lies in the fact that it is a means of communication and e-learning through Iraqi sign language, reading and writing in Arabic. Similarly, it is regarded as a means of social communication between deaf people and those with normal hearing. This application is designed by using JAVA language and it was tested on several deaf students at Al-Amal Institute for Special Needs Care in Mosul, Iraq. It was well comprehended and accepted.
The role of speech technology in biometrics, forensics and man-machine interfaceIJECEIAES
Day by day Optimism is growing that in the near future our society will witness the Man-Machine Interface (MMI) using voice technology. Computer manufacturers are building voice recognition sub-systems in their new product lines. Although, speech technology based MMI technique is widely used before, needs to gather and apply the deep knowledge of spoken language and performance during the electronic machine-based interaction. Biometric recognition refers to a system that is able to identify individuals based on their own behavior and biological characteristics. Fingerprint success in forensic science and law enforcement applications with growing concerns relating to border control, banking access fraud, machine access control and IT security, there has been great interest in the use of fingerprints and other biological symptoms for the automatic recognition. It is not surprising to see that the application of biometric systems is playing an important role in all areas of our society. Biometric applications include access to smartphone security, mobile payment, the international border, national citizen register and reserve facilities. The use of MMI by speech technology, which includes automated speech/speaker recognition and natural language processing, has the significant impact on all existing businesses based on personal computer applications. With the help of powerful and affordable microprocessors and artificial intelligence algorithms, the human being can talk to the machine to drive and control all computer-based applications. Today's applications show a small preview of a rich future for MMI based on voice technology, which will ultimately replace the keyboard and mouse with the microphone for easy access and make the machine more intelligent.
International Journal of Engineering Research and Applications (IJERA) is an open access online peer reviewed international journal that publishes research and review articles in the fields of Computer Science, Neural Networks, Electrical Engineering, Software Engineering, Information Technology, Mechanical Engineering, Chemical Engineering, Plastic Engineering, Food Technology, Textile Engineering, Nano Technology & science, Power Electronics, Electronics & Communication Engineering, Computational mathematics, Image processing, Civil Engineering, Structural Engineering, Environmental Engineering, VLSI Testing & Low Power VLSI Design etc.
Speech Automated Examination for Visually Impaired Studentsvivatechijri
We that know education is not a luxury but a necessity in today's life. It's a human right and every individual including blind and visually impaired people should also take it. It is very difficult for them to manage their education and learning process. According to recent study it has been found that over 200+ million blind people and visually impaired people are there and the number is growing. so the user have develop a learning application in which a blind or visually impaired student can study on their own without including third party in it. The student will be able to separately operate the application. Here they will be able to study various subjects of their choice, they will be able to take their own notes, and also take tests on the subject. The prime goal of this application is to solve the problem of the blind students regarding their education or studying pattern. Here they will be able to move the cursor anywhere on the desktop and the text will be dictated in the human synthesized AI voice using TTS conversion. They can easily make notes by simply dictating their points to the system, and the system will be convert into text format and also we can listened to it in an audio format. Once the learning part is done, and the student have to revise the particular topic then he/she can give test on that topic by using hand gestures. So here the system will help the visually impaired/blind student to learn and operate the system independently and also they will be able to use it as per their suitable time preference. So, the system is developed to resolve the problem of students being dependent on the third party in their studies and to increase their interest in studies. Hence, not only blind people but also other handicapped people will be able to use this application.
Industrial Applications of Automatic Speech Recognition SystemsIJERA Editor
Current trends in developing technologies form important bridges to the future, fortified by the early and
productive use of technology for enriching the human life. Speech signal processing, which includes automatic
speech recognition, synthetic speech, and natural language processing, is beginning to have a significant impact
on business, industry and ease of operation of personal computers. Apart from this, it facilitates the deeper
understanding of complex mechanism of functioning of human brain. Advances in speech recognition
technology, over the past five decades, have enabled a wide range of industrial applications. Yet today's
applications provide a small preview of a rich future for speech and voice interface technology that will
eventually replace keyboards with microphones for designing human machine interface for providing easy
access to increasingly intelligent machines. It also shows how the capabilities of speech recognition systems in
industrial applications are evolving over time to usher in the next generation of voice-enabled services. This
paper aims to present an effective survey of the speech recognition technology described in the available
literature and integrate the insights gained during the process of study of individual research and developments.
The current applications of speech recognition for real world and industry have also been outlined with special
reference to applications in the areas of medical, industrial robotics, forensic, defence and aviation
Application for Iraqi sign language translation on Android system IJECEIAES
Deaf people suffer from difficulty in social communication, especially those who have been denied the blessing of hearing before the acquisition of spoken language and before learning to read and write. For the purpose of employing mobile devices for the benefit of these people, their teachers and everyone who has contact with them, this research aims to design an application for social communication and learning by translating Iraqi sign language into text in Arabic and vice versa. Iraqi sign language has been chosen because of a lack of applications for this field. The current research, to the best of our knowledge, is the first of its kind in Iraq. The application is open source; words that are not found in the application database can be processed by translating them into letters alphabetically. The importance of the application lies in the fact that it is a means of communication and e-learning through Iraqi sign language, reading and writing in Arabic. Similarly, it is regarded as a means of social communication between deaf people and those with normal hearing. This application is designed by using JAVA language and it was tested on several deaf students at Al-Amal Institute for Special Needs Care in Mosul, Iraq. It was well comprehended and accepted.
The role of speech technology in biometrics, forensics and man-machine interfaceIJECEIAES
Day by day Optimism is growing that in the near future our society will witness the Man-Machine Interface (MMI) using voice technology. Computer manufacturers are building voice recognition sub-systems in their new product lines. Although, speech technology based MMI technique is widely used before, needs to gather and apply the deep knowledge of spoken language and performance during the electronic machine-based interaction. Biometric recognition refers to a system that is able to identify individuals based on their own behavior and biological characteristics. Fingerprint success in forensic science and law enforcement applications with growing concerns relating to border control, banking access fraud, machine access control and IT security, there has been great interest in the use of fingerprints and other biological symptoms for the automatic recognition. It is not surprising to see that the application of biometric systems is playing an important role in all areas of our society. Biometric applications include access to smartphone security, mobile payment, the international border, national citizen register and reserve facilities. The use of MMI by speech technology, which includes automated speech/speaker recognition and natural language processing, has the significant impact on all existing businesses based on personal computer applications. With the help of powerful and affordable microprocessors and artificial intelligence algorithms, the human being can talk to the machine to drive and control all computer-based applications. Today's applications show a small preview of a rich future for MMI based on voice technology, which will ultimately replace the keyboard and mouse with the microphone for easy access and make the machine more intelligent.
International Journal of Engineering Research and Applications (IJERA) is an open access online peer reviewed international journal that publishes research and review articles in the fields of Computer Science, Neural Networks, Electrical Engineering, Software Engineering, Information Technology, Mechanical Engineering, Chemical Engineering, Plastic Engineering, Food Technology, Textile Engineering, Nano Technology & science, Power Electronics, Electronics & Communication Engineering, Computational mathematics, Image processing, Civil Engineering, Structural Engineering, Environmental Engineering, VLSI Testing & Low Power VLSI Design etc.
Speech Automated Examination for Visually Impaired Studentsvivatechijri
We that know education is not a luxury but a necessity in today's life. It's a human right and every individual including blind and visually impaired people should also take it. It is very difficult for them to manage their education and learning process. According to recent study it has been found that over 200+ million blind people and visually impaired people are there and the number is growing. so the user have develop a learning application in which a blind or visually impaired student can study on their own without including third party in it. The student will be able to separately operate the application. Here they will be able to study various subjects of their choice, they will be able to take their own notes, and also take tests on the subject. The prime goal of this application is to solve the problem of the blind students regarding their education or studying pattern. Here they will be able to move the cursor anywhere on the desktop and the text will be dictated in the human synthesized AI voice using TTS conversion. They can easily make notes by simply dictating their points to the system, and the system will be convert into text format and also we can listened to it in an audio format. Once the learning part is done, and the student have to revise the particular topic then he/she can give test on that topic by using hand gestures. So here the system will help the visually impaired/blind student to learn and operate the system independently and also they will be able to use it as per their suitable time preference. So, the system is developed to resolve the problem of students being dependent on the third party in their studies and to increase their interest in studies. Hence, not only blind people but also other handicapped people will be able to use this application.
Forensic and Automatic Speaker Recognition System IJECEIAES
Current Automatic Speaker Recognition (ASR) System has emerged as an important medium of confirmation of identity in many businesses, ecommerce applications, forensics and law enforcement as well. Specialists trained in criminological recognition can play out this undertaking far superior by looking at an arrangement of acoustic, prosodic, and semantic attributes which has been referred to as structured listening. An algorithmbased system has been developed in the recognition of forensic speakers by physics scientists and forensic linguists to reduce the probability of a contextual bias or pre-centric understanding of a reference model with the validity of an unknown audio s ample and any suspicious individual. Many researchers are continuing to develop automatic algorithms in signal processing and machine learning so that improving performance can effectively introduce the speaker’s identity, where the automatic system performs equally with the human audience. In this paper, I examine the literature about the identification of speakers by machines and humans, emphasizing the key technical speaker pattern emerging for the automatic technology in the last decade. I focus on many aspects of automatic speaker recognition (ASR) systems, including speaker-specific features, speaker models, standard assessment data sets, and performance metrics.
American Journal of Multidisciplinary Research and Development is indexed, refereed and peer-reviewed journal, which is designed to publish research articles.
Speech recognition is an advanced technology that uses desired equipment and a service which can be controlled through voice without touching the screen of the smart phone. In current century, there are many researches with the help of speech recognition on mobile devices. In this system, mobile phone users can command with their voice to easily make phone call. Google's cloud speech API is used to recognize the incoming user voice. The speech API recognizes over 120 languages but it cannot correctly provide Myanmar Language still now. The system will classify the Myanmar proper name recognized by the Googles speech API to get the correct name with the help of Naïve Bayesian Classifier. The contact name classified by Naive Bayes can only meet user's desired one just written in English script and it cannot provide the name written in Myanmar script. This system uses hybrid transliteration approach to solve the contact name recorded by Myanmar script. Therefore the system can make phone call to the contact name typed with not only English script but also Myanmar script. The system applies Jaro Winker distance measure to outperform the accuracy of system output. Success rate is used to measure the performance of each process contained in the system. This system is implemented with Android programming language. Aye Thida | Yee Wai Khaing "Voice Command Mobile Phone Dialer" Published in International Journal of Trend in Scientific Research and Development (ijtsrd), ISSN: 2456-6470, Volume-3 | Issue-5 , August 2019, URL: https://www.ijtsrd.com/papers/ijtsrd26814.pdfPaper URL: https://www.ijtsrd.com/computer-science/artificial-intelligence/26814/voice-command-mobile-phone-dialer/aye-thida
Forensic and Automatic Speaker Recognition System IJECEIAES
Current Automatic Speaker Recognition (ASR) System has emerged as an important medium of confirmation of identity in many businesses, ecommerce applications, forensics and law enforcement as well. Specialists trained in criminological recognition can play out this undertaking far superior by looking at an arrangement of acoustic, prosodic, and semantic attributes which has been referred to as structured listening. An algorithmbased system has been developed in the recognition of forensic speakers by physics scientists and forensic linguists to reduce the probability of a contextual bias or pre-centric understanding of a reference model with the validity of an unknown audio s ample and any suspicious individual. Many researchers are continuing to develop automatic algorithms in signal processing and machine learning so that improving performance can effectively introduce the speaker’s identity, where the automatic system performs equally with the human audience. In this paper, I examine the literature about the identification of speakers by machines and humans, emphasizing the key technical speaker pattern emerging for the automatic technology in the last decade. I focus on many aspects of automatic speaker recognition (ASR) systems, including speaker-specific features, speaker models, standard assessment data sets, and performance metrics.
American Journal of Multidisciplinary Research and Development is indexed, refereed and peer-reviewed journal, which is designed to publish research articles.
Speech recognition is an advanced technology that uses desired equipment and a service which can be controlled through voice without touching the screen of the smart phone. In current century, there are many researches with the help of speech recognition on mobile devices. In this system, mobile phone users can command with their voice to easily make phone call. Google's cloud speech API is used to recognize the incoming user voice. The speech API recognizes over 120 languages but it cannot correctly provide Myanmar Language still now. The system will classify the Myanmar proper name recognized by the Googles speech API to get the correct name with the help of Naïve Bayesian Classifier. The contact name classified by Naive Bayes can only meet user's desired one just written in English script and it cannot provide the name written in Myanmar script. This system uses hybrid transliteration approach to solve the contact name recorded by Myanmar script. Therefore the system can make phone call to the contact name typed with not only English script but also Myanmar script. The system applies Jaro Winker distance measure to outperform the accuracy of system output. Success rate is used to measure the performance of each process contained in the system. This system is implemented with Android programming language. Aye Thida | Yee Wai Khaing "Voice Command Mobile Phone Dialer" Published in International Journal of Trend in Scientific Research and Development (ijtsrd), ISSN: 2456-6470, Volume-3 | Issue-5 , August 2019, URL: https://www.ijtsrd.com/papers/ijtsrd26814.pdfPaper URL: https://www.ijtsrd.com/computer-science/artificial-intelligence/26814/voice-command-mobile-phone-dialer/aye-thida
This is what we will do when we attend the workshop around 10 days in department of electronic and telecommunication engineering, King Mongkut's University of Technology Thonburi. This PBL is associated between KMUTT and SIT(Shibaura Institute of Technology)
This is a Powerpoint presentation about the use of assistive technology(AT) devices and services in the field of education. It defines assistive technology, describes the laws regarding AT in education, and provides examples of AT used in classrooms today for a variety of different disabilities.
Seminar report on Raspberry Pi, submitted in SEMINAR subject of GTU Gujarat Technological University by Nipun Parikh from Bhagwan Mahavir College of Engineering & Technology
A group of motor impairment syndromes resulting from disorders of early brain development and often associated with epilepsy and abnormalities of speech, vision and intellect
Read2 me a cloud based reading aid for the visually impairedLeMeniz Infotech
Read2Me A Cloud based Reading Aid for the Visually Impaired
Web : http://www.lemenizinfotech.com
Web : http://ieeemaster.com
Web : http://www.lemenizinfotech.com/android-ieee-projects-2016-2017-2/
Web : http://ieeemaster.com/android-ieee-projects-2016-2017/
Address: 36, 100 Feet Road(Near Indira Gandhi Statue), Natesan Nagar, Pondicherry-605 005
Contact numbers: +91 95663 55386, 99625 88976 (0413) 420 5444
Mail : projects@lemenizinfotech.com
Mobile : 9566355386 / 9962588976
Developing a hands-free interface to operate a Computer using voice commandMohammad Liton Hossain
The main focus of this study is to help a handicap person to operate a computer by voice command. It can be used to operate the entire computer functions on the user’s voice commands. It makes use of the Speech Recognition technology that allows the computer system to identify and recognize words spoken by a human using a microphone. This Software will be able to recognize spoken words and enable user to interact with the computer. This interaction includes user giving commands to his computer which will then respond by performing several tasks, actions or operations depending on the commands they gave. For Example: Opening /closing a file in computer, YouTube automation using voice command, Google search using voice command, make a note using voice command, calculation by calculator using voice command etc.
Abstract
In this paper we propose a new product inwhich speech is used to interact with computers. Speech is a man’s most powerful form
of communication. The user will be able to give various voice commands to the system, which the system will recognize and
execute tasks based on the input command. This system will provide another form of input (apart from mouse and keyboard) for
daily users. It will also be of great assistance to physically challenged users. The user will be able to perform all the operations
using his voice as input which he is able to perform normally using mouse and keyboard. The system however requires a little bit
training from the user, so that the system will understand the user better. The need of training is due to the fact that every person
has different voice. Also the voice of women is totally distinct from men. More training will result in faster and accurate response.
Extensive experiments are conducted in order to check the accuracy and efficiency of the system, and the results show that our
system is reliable and achieves better accuracy and efficiency than previous systems.
KeyWords: Speech Technology, Voice Response System, Voice User Interface, Voice Recognition.
Text Mining in Digital Libraries using OKAPI BM25 ModelEditor IJCATR
The emergence of the internet has made vast amounts of information available and easily accessible online. As a result, most libraries have digitized their content in order to remain relevant to their users and to keep pace with the advancement of the internet. However, these digital libraries have been criticized for using inefficient information retrieval models that do not perform relevance ranking to the retrieved results. This paper proposed the use of OKAPI BM25 model in text mining so as means of improving relevance ranking of digital libraries. Okapi BM25 model was selected because it is a probability-based relevance ranking algorithm. A case study research was conducted and the model design was based on information retrieval processes. The performance of Boolean, vector space, and Okapi BM25 models was compared for data retrieval. Relevant ranked documents were retrieved and displayed at the OPAC framework search page. The results revealed that Okapi BM 25 outperformed Boolean model and Vector Space model. Therefore, this paper proposes the use of Okapi BM25 model to reward terms according to their relative frequencies in a document so as to improve the performance of text mining in digital libraries.
Green Computing, eco trends, climate change, e-waste and eco-friendlyEditor IJCATR
This study focused on the practice of using computing resources more efficiently while maintaining or increasing overall performance. Sustainable IT services require the integration of green computing practices such as power management, virtualization, improving cooling technology, recycling, electronic waste disposal, and optimization of the IT infrastructure to meet sustainability requirements. Studies have shown that costs of power utilized by IT departments can approach 50% of the overall energy costs for an organization. While there is an expectation that green IT should lower costs and the firm’s impact on the environment, there has been far less attention directed at understanding the strategic benefits of sustainable IT services in terms of the creation of customer value, business value and societal value. This paper provides a review of the literature on sustainable IT, key areas of focus, and identifies a core set of principles to guide sustainable IT service design.
Policies for Green Computing and E-Waste in NigeriaEditor IJCATR
Computers today are an integral part of individuals’ lives all around the world, but unfortunately these devices are toxic to the environment given the materials used, their limited battery life and technological obsolescence. Individuals are concerned about the hazardous materials ever present in computers, even if the importance of various attributes differs, and that a more environment -friendly attitude can be obtained through exposure to educational materials. In this paper, we aim to delineate the problem of e-waste in Nigeria and highlight a series of measures and the advantage they herald for our country and propose a series of action steps to develop in these areas further. It is possible for Nigeria to have an immediate economic stimulus and job creation while moving quickly to abide by the requirements of climate change legislation and energy efficiency directives. The costs of implementing energy efficiency and renewable energy measures are minimal as they are not cash expenditures but rather investments paid back by future, continuous energy savings.
Performance Evaluation of VANETs for Evaluating Node Stability in Dynamic Sce...Editor IJCATR
Vehicular ad hoc networks (VANETs) are a favorable area of exploration which empowers the interconnection amid the movable vehicles and between transportable units (vehicles) and road side units (RSU). In Vehicular Ad Hoc Networks (VANETs), mobile vehicles can be organized into assemblage to promote interconnection links. The assemblage arrangement according to dimensions and geographical extend has serious influence on attribute of interaction .Vehicular ad hoc networks (VANETs) are subclass of mobile Ad-hoc network involving more complex mobility patterns. Because of mobility the topology changes very frequently. This raises a number of technical challenges including the stability of the network .There is a need for assemblage configuration leading to more stable realistic network. The paper provides investigation of various simulation scenarios in which cluster using k-means algorithm are generated and their numbers are varied to find the more stable configuration in real scenario of road.
Optimum Location of DG Units Considering Operation ConditionsEditor IJCATR
The optimal sizing and placement of Distributed Generation units (DG) are becoming very attractive to researchers these days. In this paper a two stage approach has been used for allocation and sizing of DGs in distribution system with time varying load model. The strategic placement of DGs can help in reducing energy losses and improving voltage profile. The proposed work discusses time varying loads that can be useful for selecting the location and optimizing DG operation. The method has the potential to be used for integrating the available DGs by identifying the best locations in a power system. The proposed method has been demonstrated on 9-bus test system.
Analysis of Comparison of Fuzzy Knn, C4.5 Algorithm, and Naïve Bayes Classifi...Editor IJCATR
Early detection of diabetes mellitus (DM) can prevent or inhibit complication. There are several laboratory test that must be done to detect DM. The result of this laboratory test then converted into data training. Data training used in this study generated from UCI Pima Database with 6 attributes that were used to classify positive or negative diabetes. There are various classification methods that are commonly used, and in this study three of them were compared, which were fuzzy KNN, C4.5 algorithm and Naïve Bayes Classifier (NBC) with one identical case. The objective of this study was to create software to classify DM using tested methods and compared the three methods based on accuracy, precision, and recall. The results showed that the best method was Fuzzy KNN with average and maximum accuracy reached 96% and 98%, respectively. In second place, NBC method had respective average and maximum accuracy of 87.5% and 90%. Lastly, C4.5 algorithm had average and maximum accuracy of 79.5% and 86%, respectively.
Web Scraping for Estimating new Record from Source SiteEditor IJCATR
Study in the Competitive field of Intelligent, and studies in the field of Web Scraping, have a symbiotic relationship mutualism. In the information age today, the website serves as a main source. The research focus is on how to get data from websites and how to slow down the intensity of the download. The problem that arises is the website sources are autonomous so that vulnerable changes the structure of the content at any time. The next problem is the system intrusion detection snort installed on the server to detect bot crawler. So the researchers propose the use of the methods of Mining Data Records and the method of Exponential Smoothing so that adaptive to changes in the structure of the content and do a browse or fetch automatically follow the pattern of the occurrences of the news. The results of the tests, with the threshold 0.3 for MDR and similarity threshold score 0.65 for STM, using recall and precision values produce f-measure average 92.6%. While the results of the tests of the exponential estimation smoothing using ? = 0.5 produces MAE 18.2 datarecord duplicate. It slowed down to 3.6 datarecord from 21.8 datarecord results schedule download/fetch fix in an average time of occurrence news.
Evaluating Semantic Similarity between Biomedical Concepts/Classes through S...Editor IJCATR
Most of the existing semantic similarity measures that use ontology structure as their primary source can measure semantic similarity between concepts/classes using single ontology. The ontology-based semantic similarity techniques such as structure-based semantic similarity techniques (Path Length Measure, Wu and Palmer’s Measure, and Leacock and Chodorow’s measure), information content-based similarity techniques (Resnik’s measure, Lin’s measure), and biomedical domain ontology techniques (Al-Mubaid and Nguyen’s measure (SimDist)) were evaluated relative to human experts’ ratings, and compared on sets of concepts using the ICD-10 “V1.0” terminology within the UMLS. The experimental results validate the efficiency of the SemDist technique in single ontology, and demonstrate that SemDist semantic similarity techniques, compared with the existing techniques, gives the best overall results of correlation with experts’ ratings.
Semantic Similarity Measures between Terms in the Biomedical Domain within f...Editor IJCATR
The techniques and tests are tools used to define how measure the goodness of ontology or its resources. The similarity between biomedical classes/concepts is an important task for the biomedical information extraction and knowledge discovery. However, most of the semantic similarity techniques can be adopted to be used in the biomedical domain (UMLS). Many experiments have been conducted to check the applicability of these measures. In this paper, we investigate to measure semantic similarity between two terms within single ontology or multiple ontologies in ICD-10 “V1.0” as primary source, and compare my results to human experts score by correlation coefficient.
A Strategy for Improving the Performance of Small Files in Openstack Swift Editor IJCATR
This is an effective way to improve the storage access performance of small files in Openstack Swift by adding an aggregate storage module. Because Swift will lead to too much disk operation when querying metadata, the transfer performance of plenty of small files is low. In this paper, we propose an aggregated storage strategy (ASS), and implement it in Swift. ASS comprises two parts which include merge storage and index storage. At the first stage, ASS arranges the write request queue in chronological order, and then stores objects in volumes. These volumes are large files that are stored in Swift actually. During the short encounter time, the object-to-volume mapping information is stored in Key-Value store at the second stage. The experimental results show that the ASS can effectively improve Swift's small file transfer performance.
Integrated System for Vehicle Clearance and RegistrationEditor IJCATR
Efficient management and control of government's cash resources rely on government banking arrangements. Nigeria, like many low income countries, employed fragmented systems in handling government receipts and payments. Later in 2016, Nigeria implemented a unified structure as recommended by the IMF, where all government funds are collected in one account would reduce borrowing costs, extend credit and improve government's fiscal policy among other benefits to government. This situation motivated us to embark on this research to design and implement an integrated system for vehicle clearance and registration. This system complies with the new Treasury Single Account policy to enable proper interaction and collaboration among five different level agencies (NCS, FRSC, SBIR, VIO and NPF) saddled with vehicular administration and activities in Nigeria. Since the system is web based, Object Oriented Hypermedia Design Methodology (OOHDM) is used. Tools such as Php, JavaScript, css, html, AJAX and other web development technologies were used. The result is a web based system that gives proper information about a vehicle starting from the exact date of importation to registration and renewal of licensing. Vehicle owner information, custom duty information, plate number registration details, etc. will also be efficiently retrieved from the system by any of the agencies without contacting the other agency at any point in time. Also number plate will no longer be the only means of vehicle identification as it is presently the case in Nigeria, because the unified system will automatically generate and assigned a Unique Vehicle Identification Pin Number (UVIPN) on payment of duty in the system to the vehicle and the UVIPN will be linked to the various agencies in the management information system.
Assessment of the Efficiency of Customer Order Management System: A Case Stu...Editor IJCATR
The Supermarket Management System deals with the automation of buying and selling of good and services. It includes both sales and purchase of items. The project Supermarket Management System is to be developed with the objective of making the system reliable, easier, fast, and more informative.
Energy-Aware Routing in Wireless Sensor Network Using Modified Bi-Directional A*Editor IJCATR
Energy is a key component in the Wireless Sensor Network (WSN)[1]. The system will not be able to run according to its function without the availability of adequate power units. One of the characteristics of wireless sensor network is Limitation energy[2]. A lot of research has been done to develop strategies to overcome this problem. One of them is clustering technique. The popular clustering technique is Low Energy Adaptive Clustering Hierarchy (LEACH)[3]. In LEACH, clustering techniques are used to determine Cluster Head (CH), which will then be assigned to forward packets to Base Station (BS). In this research, we propose other clustering techniques, which utilize the Social Network Analysis approach theory of Betweeness Centrality (BC) which will then be implemented in the Setup phase. While in the Steady-State phase, one of the heuristic searching algorithms, Modified Bi-Directional A* (MBDA *) is implemented. The experiment was performed deploy 100 nodes statically in the 100x100 area, with one Base Station at coordinates (50,50). To find out the reliability of the system, the experiment to do in 5000 rounds. The performance of the designed routing protocol strategy will be tested based on network lifetime, throughput, and residual energy. The results show that BC-MBDA * is better than LEACH. This is influenced by the ways of working LEACH in determining the CH that is dynamic, which is always changing in every data transmission process. This will result in the use of energy, because they always doing any computation to determine CH in every transmission process. In contrast to BC-MBDA *, CH is statically determined, so it can decrease energy usage.
Security in Software Defined Networks (SDN): Challenges and Research Opportun...Editor IJCATR
In networks, the rapidly changing traffic patterns of search engines, Internet of Things (IoT) devices, Big Data and data centers has thrown up new challenges for legacy; existing networks; and prompted the need for a more intelligent and innovative way to dynamically manage traffic and allocate limited network resources. Software Defined Network (SDN) which decouples the control plane from the data plane through network vitalizations aims to address these challenges. This paper has explored the SDN architecture and its implementation with the OpenFlow protocol. It has also assessed some of its benefits over traditional network architectures, security concerns and how it can be addressed in future research and related works in emerging economies such as Nigeria.
Measure the Similarity of Complaint Document Using Cosine Similarity Based on...Editor IJCATR
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Assistive Examination System for Visually Impaired
1. International Journal of Computer Applications Technology and Research
Volume 4– Issue 2, 157 - 164, 2015, ISSN:- 2319–8656
www.ijcat.com 157
Assistive Examination System for Visually Impaired
Manvi Breja
Manav Rachna College of
Engineering
Faridabad, Haryana, India
Abstract: This paper presents a design of voice enabled examination system which can be used by the visually challenged students.
The system uses Text-to-Speech (TTS) and Speech-to-Text (STT) technology. The text-to-speech and speech-to-text web based
academic testing software would provide an interaction for blind students to enhance their educational experiences by providing them
with a tool to give the exams. This system will aid the differently-abled to appear for online tests and enable them to come at par with
the other students. This system can also be used by students with learning disabilities or by people who wish to take the examination in
a combined auditory and visual way.
Keywords: Speech recognition, speech synthesis, prosody analysis, phonemes, speech API.
1. INTRODUCTION
In today‘s era of rapidly evolving technological advances, a
major change has occurred in the educational system
prevalent in schools and colleges. Along with other changes in
the teaching system, the examination system has evolved
significantly Voice Enabled Examination System for the
visually impaired people has been a very active research area
for a long time and much more success is also achieved in this
area.. According to the National Center for Educational
Statistics , ―The number of students with disabilities attending
higher education has increased. In a recent study, the number
of postsecondary undergraduate students identified as having
disabilities in the United States was found to be 428,280,
representing 6% of the student body.‖ With the growing
number of blind people attending college, there has been a
growing need for such system that can aid these visually
impaired students .In today‘s era, the conventional pen and
paper tests have been replaced by online examination systems.
The word ‗Online‘ here refers to not necessarily web-based or
browser-dependent, but to a network that links all of the test
takers to a common server. As of now, there are many
companies in the industry that offer testing solutions, but none
of them has on offer an examination system which is voice-
enabled and assists the differently abled.
2. RELATED WORK
Technology has removed many barriers to education and
employment for visually impaired individuals. Various
technology exists for students with visual impairements.
These include:
2.1 Screen Magnification
Screen magnification software is used by people with visual
impairments to access information on computer screens. The
software enlarges information on the screen by incremental
factors (2 x magnification, 3x up to 20x magnification). Most
screen magnification programs have the flexibility to magnify
the full screen, parts of the screen, or a magnifying glass view
of the area around the cursor or pointer. Commonly used
screen magnification software are-
• MAGic, developed by Freedom Scientific Inc. Blind/Low
Vision Group.
• ZoomText, developed by Ai Squared
• BigShot, developed by Ai Squared .
2.2 Screen Readers
A screen reader is a software application that attempts to
identify and interpret what is being displayed on the screen.
Screen reading software reads aloud everything on computer
screens, including text, pull-down menus, icons, dialog boxes,
and web pages. Screen readers run simultaneously with the
computer's operating system and applications. There are
mainly two types of screen readers- the CLI screen readers
and the GUI screen readers.
2.3 Optical Character Recognition Systems
Optical character recognition (OCR) technology offers blind
and visually impaired persons the capacity to scan printed text
and then speak it back in synthetic speech or save it to a
computer. There are three essential elements to OCR
technology—scanning, recognition, and reading text. Initially,
a printed document is scanned by a camera. OCR software
then converts the images into recognized characters and
words. Some of the most popular OCR systems are:
Kurzweil 1000, developed by Kurzweil Educational
Systems
OpenBook, developed by Freedom Scientific Inc.
Eye-Pal, developed by ABISee, Inc.
2.4 Electronic Portable Note-Takers
Electronic Braille note takers are small, portable devices with
Braille keyboards for entering information. They use a speech
synthesizer or Braille display for output. The user enters the
information on the Braille keyboard and has the option of
transferring it to a larger computer with more memory,
reviewing it using the built in speech synthesizer or Braille
display, or printing it on a Braille or ink print printer.
The commonly used note takers are:
Braille ‗n‘ Speak, developed by Freedom Scientific,
Inc.
Type ‗n‘ Speak, developed by Freedom Scientific,
Inc.
PacMate Series, developed by Freedom Scientific,
Inc.
VoiceNote, developed by Pulse Data.
2. International Journal of Computer Applications Technology and Research
Volume 4– Issue 2, 157 - 164, 2015, ISSN:- 2319–8656
www.ijcat.com 158
2.5 Portable Reading Devices
One of newer blind technologies is the portable reading
device, which downloads books and then reads them out loud
in a synthesized voice. They are specially designed with the
blind and visually impaired community in mind. The Victor
Reader Stream and the BookSense audio book are popular
models.
3. PROPOSED SYSTEM
The prime interest behind the development of this system is to
implement speech technology in an application in such a way
that it enables the visually-challenged candidates to appear for
a computer-adaptive online examination. The system is a
stand-alone application which uses Speech-To-Text (STT)
and Text-To-Speech (TTS) technology to provide the users
almost all of the capabilities of a conventional online
examination.
The online examination system is adaptable to different types
of questions pertaining to different subjects, different time
limits and different marking schemes, and can be customized
according to the needs of any organization. All the data
pertaining to the test is stored in a database which is linked to
the application.
The Voice Enabled Examination System is able to read aloud
the questions and the different options available to the test-
taker. The candidate has to answer the question by speaking
out the option number. The system registers the answer given
by the candidate and moves on to the next question. At the
end of the test, a report is generated by the system.
This system is equipped with the following functionalities:-
Authentication of candidates via a mechanism of
unique registration id and FolderLocking.
Reading out of the questions by the application.
Registering the answer of the candidate which has
been spoken by him/her.
Announcement of score to the candidate at the end
of the exam.
Folder Locking which ensures encryption of
candidates‘ data in a folder and allows only the
administrator to unlock that folder containing the
candidates‘ details of the exam.
Sending the resultant score sheet of the
Examination to the registered mail id of the
Candidate as well as to the Administrator.
Generating the Certificate of scored marks and
printing it at the end of the exam.
Voice notifications to the candidates about the
status of time left for each question.
will inculcate the feature of Photograph Matching of
the candidate while he appears for an exam from
the administrator side for the security purposes.
The remainder of this paper is organized as follows. Section
IV introduces some introduction on speech synthesis, Section
V presents the concepts of speech recognition, Section VI
discusses the application frameworks used, Section VII
presents the snapshots of the results, Section VIII presents the
future scope. Finally Section IX concludes the paper.
4. SPEECH SYNTHESIS
Speech synthesis is the artificial production of human speech.
A synthesizer can be implemented in software or hardware.. A
Text-To-Speech (TTS) synthesizer is a computer-based
system that should be able to read any text aloud, whether it
was directly introduced in the computer by an operator or
scanned and submitted to an Optical Character Recognition
(OCR) system. A text-to-speech (TTS) system converts
normal language text into speech; other systems
render symbolic linguistic representations like phonetic
transcriptions into speech Synthesized speech can be created
by concatenating pieces of recorded speech that are stored in
a database. Systems that simply concatenate isolated words or
parts of sentences, denoted as Voice Response Systems, are
only applicable when a limited vocabulary is required
(typically a few one hundreds of words), and when the
sentences to be pronounced respect a very restricted structure.
It is thus more suitable to define Text-To-Speech as the
automatic production of speech, through a grapheme-to-
phoneme transcription of the sentences to utter [4].
The quality of a speech synthesizer is judged by its similarity
to the human voice and by its ability to be understood. An
intelligible text-to-speech program allows people with visual
impairments or reading disabilities to listen to written works
on a home computer. Many computer operating systems have
included speech synthesizers since the early 1980s. The text-
to-speech (TTS) synthesis procedure consists of two main
phases. The first one is text analysis, where the input text is
transcribed into a phonetic or some other linguistic
representation, and the second one is the generation of speech
waveforms, where the acoustic output is produced from this
phonetic and prosodic information. These two phases are
usually called as high- and low-level synthesis. A simplified
version of the procedure is presented in Figure.
Figure 1: Process of Text-to-Speech Synthesizer
The input text might be for example data from a word
processor, standard ASCII from e-mail, a mobile text-
message, orscanned text from a newspaper. The character
string is then preprocessed and analyzed into phonetic
representation which is usually a string of phonemes with
some additional information for correct intonation, duration,
and stress. Speech sound is finally generated with the low-
level synthesizer by the information from high-level one [8].
4.1. Process of Speech Synthesis
4.1.1. Structure analysis
Processes the input text to determine where paragraphs,
sentences, and other structures start and end. For most
languages, punctuation and formatting data are used in this
stage.
4.1.2. Text pre-processing:
Analyzes the input text for special constructs of the language.
In English, special treatment is required for abbreviations,
acronyms, dates, times, numbers, currency amounts, e-mail
addresses, and many other forms. Other languages need
special processing for these forms, and most languages have
other specialized requirements.
The remaining steps convert the spoken text to speech:
4.1.3. Text-to-phoneme conversion:
3. International Journal of Computer Applications Technology and Research
Volume 4– Issue 2, 157 - 164, 2015, ISSN:- 2319–8656
www.ijcat.com 159
Converts each word to phonemes. A phoneme is a basic unit
of sound in a language.
4.1.4. Prosody analysis:
Processes the sentence structure, words, and phonemes to
determine the appropriate prosody for the sentence.
4.1.5. Waveform production:
Uses the phonemes and prosody information to produce the
audio waveform for each sentence [3].
4.2. Synthesizer Technologies
The most important qualities of a speech synthesis system
are naturalness and intelligibility. The ideal speech
synthesizer is both natural and intelligible. Speech synthesis
systems usually try to maximize both characteristics.
The two primary technologies for generating synthetic speech
waveforms are concatenative synthesis and formant synthesis.
Each technology has strengths and weaknesses, and the
intended uses of a synthesis system will typically determine
which approach is used [7].
5. SPEECH RECOGNITION SYSTEM
The speech is primary mode of communication among human
being and also the most natural and efficient form of
exchanging information among human in speech. Speech
Recognition can be defined as the process of converting
speech signal to a sequence of words by means of an
algorithm implemented as a computer program. Speech
processing is one of the exciting areas of signal processing[1].
Since the 1960s computer scientists have been researching
ways and means to make computers able to record interpret
and understand human speech. Throughout the decades this
has been a daunting task. Even the most rudimentary in the
early years. It took until the 1980s before the first systems
problem such as digitalizing (sampling) voice was a huge
challenge arrived which could actually decipher speech. Off
course these early systems were very limited in scope and
power. Communication among the human being is dominated
by spoken language, therefore it is natural for people to expect
speech interfaces with computer .computer which can speak
and recognize speech in native language[2]. Machine
recognisation of speech involves generating a sequence of
words best matches the given speech signal. Some of known
applications include virtual reality, Multimedia searches, auto-
attendants, travel Information and reservation, translators,
natural language understanding and many more Applications.
Figure 2: Structure of a standard speech recognition
system.
A standard speech recognition system consists of the
following components [5] :-
Raw speech. Speech is typically sampled at a high
frequency, e.g., 16 KHz over a microphone or 8 KHz over a
telephone. This yields a sequence of amplitude values over
time.
Signal analysis. Raw speech should be initially transformed
and compressed, in order to simplify subsequent processing.
Many signal analysis techniques are available which can
extract useful features and compress the data by a factor of ten
without losing any important information.
Figure 3: Conversion of raw speech to speech frames
through signal anaysis
Speech frames. The result of signal analysis is a sequence of
speech frames, typically at 10 msec intervals, with about 16
coefficients per frame. These frames may be augmented by
their own first and/or second derivatives, providing explicit
information about speech dynamics; this typically leads to
improved performance. The speech frames are used for
acoustic analysis.
Acoustic models. In order to analyze the speech frames for
their acoustic content, we need a set of acoustic models. There
are many kinds of acoustic models, varying in their
representation, granularity, context dependence, and other
properties.
The major steps of a typical speech recognizer are as follows:
Grammar design:
Defines the words that may be spoken by a user and the
patterns in which they may be spoken.
Signal processing:
Analyzes the spectrum (i.e., the frequency) characteristics of
the incoming audio.
Phoneme recognition:
Compares the spectrum patterns to the patterns of the
phonemes of the language being recognized.
Word recognition:
Compares the sequence of likely phonemes against the words
and patterns of words specified by the active grammars.
Result generation:
Provides the application with information about the words the
recognizer has detected in the incoming audio.
Figure 4: Process of Speech Recognition and Speech
synthesis
5.1. Speech Recognition Techniques
The goal of speech recognition is for a machine to be able to
"hear,‖ understand," and "act upon" spoken information. The
4. International Journal of Computer Applications Technology and Research
Volume 4– Issue 2, 157 - 164, 2015, ISSN:- 2319–8656
www.ijcat.com 160
earliest speech recognition systems were first attempted in the
early 1950s at Bell Laboratories, Davis, Biddulph and
Balashek developed an isolated digit Recognition system for a
single speaker. The goal of automatic speaker recognisation is
to analyze, extract characterize and recognize information
about the speaker identity. The speaker recognisation system
may be viewed as working in a four stages
1. Analysis
2. Feature extraction
3. Modeling
4. Testing
5.1.1. Speech analysis technique
Speech data contain different type of information that shows a
speaker identity. This includes speaker specific information
due to vocal tract, excitation source and behavior feature. The
information about the behavior feature also embedded in
signal and that can be used for speaker recognition. The
speech analysis stage deals with stage with suitable frame size
for segmenting speech signal for further analysis and
extracting [6] .
5.1.2. Feature Extraction Technique
The speech feature extraction in a categorization problem is
about reducing the dimensionality of the input vector while
maintaining the discriminating power of the signal. As we
know from fundamental formation of speaker identification
and verification system, that the number of training and test
vector needed for the classification problem grows with the
dimension of the given input so we need feature extraction of
speech signal
5.1.3 Modeling Technique
The objective of modeling technique is to generate speaker
models using speaker specific feature vector. The speaker
modeling technique divided into two classification speaker
recognition and speaker identification. The speaker
identification technique automatically identify who is
speaking on basis of individual information integrated in
speech signal The speaker recognisation is also divided into
two parts that means speaker dependant and speaker
independent. In the speaker independent mode of the speech
recognisation the computer should ignore the speaker specific
characteristics of the speech signal and extract the intended
message .on the other hand in case of speaker recognisation
machine should extract speaker characteristics in the acoustic
signal. The main aim of speaker identification is comparing a
speech signal from an unknown speaker to a database of
known speaker. Speaker recognition can also be divide into
two methods, text- dependent and text independent methods.
In text dependent method the speaker say key words or
sentences having the same text for both training and
recognition trials. Whereas text independent does not rely on a
specific texts being spoken
5.1.4 Matching Techniques
Speech-recognition engines match a detected word to a known
word using one of the following techniques:
.
5.1.4.1. Whole-word matching
The engine compares the incoming digital-audio signal
against a prerecorded template of the word. This technique
takes much less processing than sub-word matching, but it
requires that the user (or someone) prerecord every word that
will be recognized - sometimes several hundred thousand
words. Whole-word templates also require large amounts of
storage (between 50 and 512 bytes per word) and are practical
only if the recognition vocabulary is known when the
application is developed .
5.1.4.2. Sub-word matching
The engine looks for sub-words – usually phonemes and then
performs further pattern recognition on those. This technique
takes more processing than whole-word matching, but it
requires much less storage (between 5 and 20 bytes per word).
In addition, the pronunciation of the word can be guessed
from English text without requiring the user to speak the word
beforehand to discuss that research in the area of automatic
speech recognition had been pursued for the last three
decades.
Figure 5: The general scheme of a concatenative text-to-
speech system.
6. APPLICATION FRAMEWORKS
Several methods and interfaces for making the
implementation of synthesized speech in desired applications
easier have been developed during this decade. It is quite clear
that it is impossible to create a standard for speech synthesis
methods because most systems
act as stand alone device which means they are incompatible
with each other and do not share common parts. However, it
is possible to standardize the interface of data flow between
the application and the synthesizer. Usually, the interface
contains a set of control characters or variables for controlling
the synthesizer output and features. The output is usually
controlled by normal play, stop, pause, and resume type
commands and the controllable features are usually pitch
baseline and range, speech rate, volume, and in some cases
even different voices, ages, and genders are available. Most of
the present synthesis systems support so called Speech
Application Programming Interface (SAPI) which makes
easier the implementation of speech in any kind of
application. For Internet purposes several kind of speech
synthesis markup languages have been developed to make it
possible to listen to synthesized speech without having to
transfer the actual speech signal through network.
6.1. Speech Application Programming
Interface
SAPI is an interface between applications and speech
technology engines, both text-tospeech 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.
Speech synthesis and recognition applications usually require
plenty of computational resources and with SAPI approach
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lots of these resources may be saved. The user of an
application can also choose the synthesizer used as long as it
supports SAPI. Currently SAPIs are available for several
environments, such as MS-SAPI for Microsoft Windows
operating systems and Sun Microsystems Java SAPI (JSAPI)
for JAVA based applications. SAPI text-to-speech part
consists of three interfaces. The voice text interface which
provides methods to start, pause, resume, fast attribute
interface allows access to control the basic behavior of the
forward, rewind, and stop the TTS engine during speech. The
TTS engine, such as the audio device to be used, the playback
speed (in words per minute), and turning the speech on and
off. With some TTS systems the attribute interface may also
be used to select the speaking mode from predefined list of
voices, such as female, male, child, or alien. Finally, the
dialog interface can be used to set and retrieve information
regarding the TTS engine to for example identify the TTS
engine and alter the pronunciation lexicon.
7. RESULTS
The proposed system opens up with the interface which has
the two functionalities: login for the administrator and the
student and watching the video tutorials on the related test
topic.
Figure 6: Cover Page that is opened when the project is
run
After clicking the login button, login page for candidate opens
and from that administrator can also login. Login requires the
authenticated username and password.
Figure 7: Login form for candidate
Figure 8: Login form for administrator
The administrator has the functionalities of registering the
user for the system, adding the details of the users, ability to
view all the user‘s details, folder lock facility so that the
folder is authenticated, can be viewed by only the
adminstrator.
Figure 9: Admin panel that opens up whenever
administrator logins
The users who have come to appear for the test need to give
their details to administrator for maintaining the record.
Figure 10: Display of user’s details in grid layout format
The administrator has the ability to search the record of any
candidate, adding the new candidate, deleting, updating the
details, adding the photograph of the candidate.
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Figure 11: Add user detail form to be filled for each
candidate
The administrator uses the facility of the folder lock i.e. folder
in which candidate‘s records are kept can be accessible by
only the administrator.
Figure 12: When the administrator clicks on folder lock
Figure 13: choosing the password to lock the desired
folder
Figure 14: Displaying the locked status of folder
Authentication for the candidate who have come to appear for
the test is also done. Since the candidate are blind, more
security needs to be encorporated. Voice clip is taken when
the students comes for registration. When the student will
come to appear for the test, his voice clip is matched. Only
when the authentication is done, the user is allowed to appear
for the test.
Figure 15: If authentication succeeds, matching video clip
is to be browsed
Figure 16: Displaying the status that the voice clip is
matched
Now when the authenticated user login for the test, the
system read aloud the initial instructions for the test.
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Figure 17: Instructions and guidelines for the exam
When the student listens the instruction and speaks Next, the
test starts. The system reads aloud the questions and options,
the student in turns speaks the option for answer. If the
candidate wants to skip a certain question, he can simply go to
the next one by speaking ‗Next Question‘ or to the previous
one by saying ‗Previous Question‘. The sytem is having the
facility to tell the status of the left out time also.
Figure 18: Displaying the voice enabled exam screen
After the completion of the test, the system will announce the
candidates score to him. Also, the system will generate a print
of his score card and will mail a copy of the same to his e-
mail id also. When the test has been completed, the system
will encrypt the candidate‘s score and store it in the database.
8. FUTURE SCOPE
The applications of speaker recognition technology are quite
varied and continually growing. Below is an outline of some
broad areas where speaker recognition technology has been or
is currently used.
8.1. Access Control:
Originally for physical facilities, more recent applications are
for controlling access to computer networks (add biometric
factor to usual password and/or token) or websites (thwart
password sharing for access to subscription sites). Also used
for automated password reset services.
8.2. Transaction Authentication:
For telephone banking, in addition to account access control,
higher levels of verification can be used for more sensitive
transactions. More recent applications are in user verification
for remote electronic and mobile purchases (e- and m-
commerce).
8.3. Law Enforcement:
Some applications are home-parole monitoring (call parolees
at random times to verify they are at home) and prison call
monitoring (validate inmate prior to outbound call). There has
also been discussion of using automatic systems to
corroborate aural/spectral inspections of voice samples for
forensic analysis.
8.4. Speech Data Management:
In voice mail browsing or intelligent answering machines, use
speaker recognition to label incoming voice mail with speaker
name for browsing and/or action (personal reply). For speech
skimming or audio mining applications, annotate recorded
meetings or video with speaker labels for quick indexing and
filing.
8.5. Personalization:
In voice-web or device customization, store and retrieve
personal setting/preferences based on user verification for
multi-user site or device (car climate and radio settings).
There is also interest in using recognition techniques for
directed advertisement or services, where, for example, repeat
users could be recognized or advertisements focused based on
recognition of broad speaker characteristics (e.g. gender or
age).
8.6. Aids for the disabled:
One of the longest-established applications of TTS synthesis
is in reading machines for the blind. The first such machine,
combining an optical character reader with a TTS synthesizer,
was produced by Kurzweil Computer Products in the 1970s.
Even now, this speech synthesis task is very difficult as the
machine must cope with any arbitrary text, and the quality of
the speech that is generated would be regarded as insufficient
by many people. However, these systems provide the visually
impaired with the facility to read text that would not otherwise
be available to them.
8.7. Remote e-mail readers:
A specialized but very useful application of TTS synthesis is
to provide remote access to e-mail from any fixed or mobile
telephone. For an e-mail reader, a full TTS conversion facility
is required because the messages may contain any text
characters.
E-mail messages are often especially challenging, due to the
tendency to errors of spelling and grammar as well as the
special nature of the language, abbreviations and so on that
are often used. There are also many formatting features that
are specific to e-mail.
9. CONCLUSION
The system has led to the development of a voice
enabled examination system, as a tool for giving voice
enabled exam. The system has been designed keeping
in view the requirements of visually impaired students
to aid them to keep pace with ordinary people in the
field of education.
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The testing application was based on Text-to-
Speech(TTS) and Speech-to-Text(STT) technology
which was implemented using Microsoft Speech
API(SAPI). This web based academic testing software
would provide an interaction medium for blind or
partially sighted students to enhance their educational
experiences.
While designing the software, the developers have used
the natural voice that read aloud the questions in the test
which the blind students have to answer and taken care
of the accuracy of the synthetic pronunciation so as to
provide the appropriate answer to the question.
The developers have made the use of Speech
Application Programming Interface (SAPI) which act as
an interface between applications and speech
technology engines, both text-to-speech and speech
recognition.
Many assistive technologies was present but no such
system was offering an easy, accessible and intelligible
interaction of the visually challenged with the
computer. This system will prove to be an indispensable
tool.
Incorporating and implementing the accessibility
technology during the development of this testing
application provided numerous advantages to the
sighted users, thereby giving the test more efficiently.
Thus voice enabled examination system, is a vital
technology that can be beneficial for all types of users.
10. REFERENCES
[1] R. Klevansand, R. Rodman, ―Voice Recognition, Artech
House, Boston, London,1997.
[2] Samudravijaya K. Speech and Speaker recognition
tutorial TIFR
[3] http://www.w3.org/TR/speech-synthesis/
[4] Nurulisma Ismail and Halimah Badioze Zaman, Search
Engine Module in Voice Recognition Browser to
Facilitate the Visually Impaired in Virtual Learning
(MGSYS VISI-VL), World Academy of Science,
Engineering and Technology, Volume 71, 2010.
[5] Michael Koerner, 1996, Speech Recognition: The Future
Now, Prentice Hall Professional Technical Reference,
306.
[6] GIN-DER WU AND YING LEI ― A Register Array
based Low power FFT Processor for speech recognition‖
Department of Electrical engineering national Chi Nan
university Puli ,545 Taiwan
[7] .Dutoit, T. (1999). A short introduction to text-to-speech
synthesis [Web page]. Mons: TCTS Lab, Faculté
Polytechnique de Mons. Retrieved from
http://tcts.fpms.ac.be/synthesis/introtts_old.html.
[8] http://en.wikipedia.org/wiki/Speech_synthesis- speech