—This paper presents a signal processing approach to analyse and identify accent discriminative features of four groups of English as a second language (ESL) speakers, including Chinese, Indian, Japanese, and Korean. The features used for speech recognition include pitch, stress, formant frequencies, the Mel frequency coefficient, log frequency coefficient, and the intensity and duration of vowels spoken. This paper presents our study using the Matlab Speech Analysis Toolbox, and highlights how data processing can be automated and results visualised. The proposed algorithm achieved an average success rate of 57.3% in identifying vowels spoken in a speech by the four nonnative English speaker groups.
Kannada Phonemes to Speech Dictionary: Statistical ApproachIJERA Editor
The input or output of a natural Language processing system can be either written text or speech. To process written text we need to analyze: lexical, syntactic, semantic knowledge about the language, discourse information, real world knowledge to process spoken language, we need to analyze everything required to process written text, along with the challenges of speech recognition and speech synthesis. This paper describes how articulatory phonetics of Kannada is used to generate the phoneme to speech dictionary for Kannada; the statistical computational approach is used to map the elements which are taken from input query or documents. The articulatory phonetics is the place of articulation of a consonant. It is the point of contact where an obstruction occurs in the vocal tract between an articulatory gesture, an active articulator, typically some part of the tongue, and a passive location, typically some part of the roof of the mouth. Along with the manner of articulation and the phonation, this gives the consonant its distinctive sound. The results are presented for the same.
ADVANCEMENTS ON NLP APPLICATIONS FOR MANIPURI LANGUAGEkevig
Manipuri is both a minority and morphologically rich language with genetic features similar to Tibeto Burman languages. It has Subject-Object-Verb (SOV) order, agglutinative verb morphology and ismonosyllabic. Morphology and syntax are not clearly distinguished in this language. Natural Language Processing (NLP) is a useful research field of computer science that deals with processing of a large amount of natural language corpus. The NLP applications encompass E-Dictionary, Morphological
Analyzer, Reduplicated Multi-Word Expression (RMWE), Named Entity Recognition (NER), Part of Speech (POS) Tagging, Machine Translation (MT), Word Net, Word Sense Disambiguation (WSD) etc. In this paper, we present a study on the advancements in NLP applications for Manipuri language, at the same time presenting a comparison table of the approaches and techniques adopted and the results obtained of each of the applications followed by a detail discussion of each work.
ADVANCEMENTS ON NLP APPLICATIONS FOR MANIPURI LANGUAGEijnlc
Manipuri is both a minority and morphologically rich language with genetic features similar to Tibeto Burman languages. It has Subject-Object-Verb (SOV) order, agglutinative verb morphology and is monosyllabic. Morphology and syntax are not clearly distinguished in this language. Natural Language
Processing (NLP) is a useful research field of computer science that deals with processing of a large amount of natural language corpus. The NLP applications encompass E-Dictionary, Morphological Analyzer, Reduplicated Multi-Word Expression (RMWE), Named Entity Recognition (NER), Part of Speech
(POS) Tagging, Machine Translation (MT), Word Net, Word Sense Disambiguation (WSD) etc. In this paper, we present a study on the advancements in NLP applications for Manipuri language, at the same time presenting a comparison table of the approaches and techniques adopted and the results obtained of each of the applications followed by a detail discussion of each work.
On Developing an Automatic Speech Recognition System for Commonly used Englis...rahulmonikasharma
Speech is one of the easiest and the fastest way to communicate. Recognition of speech by computer for various languages is a challenging task. The accuracy of Automatic speech recognition system (ASR) remains one of the key challenges, even after years of research. Accuracy varies due to speaker and language variability, vocabulary size and noise. Also, due to the design of speech recognition that is based on issues like- speech database, feature extraction techniques and performance evaluation. This paper aims to describe the development of a speaker-independent isolated automatic speech recognition system for Indian English language. The acoustic model is build using Carnegie Mellon University (CMU) Sphinx tools. The corpus used is based on Most Commonly used English words in everyday life. Speech database includes the recordings of 76 Punjabi Speakers (north-west Indian English accent). After testing, the system obtained an accuracy of 85.20 %, when trained using 128 GMMs (Gaussian Mixture Models).
Comparative study of Text-to-Speech Synthesis for Indian Languages by using S...ravi sharma
This paper explores syllable approach to building language independent text to speech systems for Indian Languages. The use of common phone set, common question set and borrowing context-independent monophone models along with syllable approach across languages makes the procedure easier and less time-consuming, without compromising the synthesized speech quality. Systems can be built without even knowing the language. This is especially quite beneficial in the Indian scenario.
Accents of English have been investigated for many years both from the perspective of native and non-native speakers of the language. Various research results imply that non-native speakers of English language produce certain speech characteristics which are uncommon in native speakers’ speech. This is because non-native speakers do not produce the same tongue movement as native speakers. This paper presents an isolated English word recognition system devised with the speech of local Bangladeshi people, who are also non-native speakers of English language. Here, we have also noticed a different speech characteristic which is not available within the speech of native English speakers. Two acoustic features, ‘pitch’ and ‘formants’ have been utilized to develop the system. The system is speaker-independent and stands on Template based approach. The recognition method applied here is very simple and the recognition accuracy is also very satisfactory.
Kannada Phonemes to Speech Dictionary: Statistical ApproachIJERA Editor
The input or output of a natural Language processing system can be either written text or speech. To process written text we need to analyze: lexical, syntactic, semantic knowledge about the language, discourse information, real world knowledge to process spoken language, we need to analyze everything required to process written text, along with the challenges of speech recognition and speech synthesis. This paper describes how articulatory phonetics of Kannada is used to generate the phoneme to speech dictionary for Kannada; the statistical computational approach is used to map the elements which are taken from input query or documents. The articulatory phonetics is the place of articulation of a consonant. It is the point of contact where an obstruction occurs in the vocal tract between an articulatory gesture, an active articulator, typically some part of the tongue, and a passive location, typically some part of the roof of the mouth. Along with the manner of articulation and the phonation, this gives the consonant its distinctive sound. The results are presented for the same.
ADVANCEMENTS ON NLP APPLICATIONS FOR MANIPURI LANGUAGEkevig
Manipuri is both a minority and morphologically rich language with genetic features similar to Tibeto Burman languages. It has Subject-Object-Verb (SOV) order, agglutinative verb morphology and ismonosyllabic. Morphology and syntax are not clearly distinguished in this language. Natural Language Processing (NLP) is a useful research field of computer science that deals with processing of a large amount of natural language corpus. The NLP applications encompass E-Dictionary, Morphological
Analyzer, Reduplicated Multi-Word Expression (RMWE), Named Entity Recognition (NER), Part of Speech (POS) Tagging, Machine Translation (MT), Word Net, Word Sense Disambiguation (WSD) etc. In this paper, we present a study on the advancements in NLP applications for Manipuri language, at the same time presenting a comparison table of the approaches and techniques adopted and the results obtained of each of the applications followed by a detail discussion of each work.
ADVANCEMENTS ON NLP APPLICATIONS FOR MANIPURI LANGUAGEijnlc
Manipuri is both a minority and morphologically rich language with genetic features similar to Tibeto Burman languages. It has Subject-Object-Verb (SOV) order, agglutinative verb morphology and is monosyllabic. Morphology and syntax are not clearly distinguished in this language. Natural Language
Processing (NLP) is a useful research field of computer science that deals with processing of a large amount of natural language corpus. The NLP applications encompass E-Dictionary, Morphological Analyzer, Reduplicated Multi-Word Expression (RMWE), Named Entity Recognition (NER), Part of Speech
(POS) Tagging, Machine Translation (MT), Word Net, Word Sense Disambiguation (WSD) etc. In this paper, we present a study on the advancements in NLP applications for Manipuri language, at the same time presenting a comparison table of the approaches and techniques adopted and the results obtained of each of the applications followed by a detail discussion of each work.
On Developing an Automatic Speech Recognition System for Commonly used Englis...rahulmonikasharma
Speech is one of the easiest and the fastest way to communicate. Recognition of speech by computer for various languages is a challenging task. The accuracy of Automatic speech recognition system (ASR) remains one of the key challenges, even after years of research. Accuracy varies due to speaker and language variability, vocabulary size and noise. Also, due to the design of speech recognition that is based on issues like- speech database, feature extraction techniques and performance evaluation. This paper aims to describe the development of a speaker-independent isolated automatic speech recognition system for Indian English language. The acoustic model is build using Carnegie Mellon University (CMU) Sphinx tools. The corpus used is based on Most Commonly used English words in everyday life. Speech database includes the recordings of 76 Punjabi Speakers (north-west Indian English accent). After testing, the system obtained an accuracy of 85.20 %, when trained using 128 GMMs (Gaussian Mixture Models).
Comparative study of Text-to-Speech Synthesis for Indian Languages by using S...ravi sharma
This paper explores syllable approach to building language independent text to speech systems for Indian Languages. The use of common phone set, common question set and borrowing context-independent monophone models along with syllable approach across languages makes the procedure easier and less time-consuming, without compromising the synthesized speech quality. Systems can be built without even knowing the language. This is especially quite beneficial in the Indian scenario.
Accents of English have been investigated for many years both from the perspective of native and non-native speakers of the language. Various research results imply that non-native speakers of English language produce certain speech characteristics which are uncommon in native speakers’ speech. This is because non-native speakers do not produce the same tongue movement as native speakers. This paper presents an isolated English word recognition system devised with the speech of local Bangladeshi people, who are also non-native speakers of English language. Here, we have also noticed a different speech characteristic which is not available within the speech of native English speakers. Two acoustic features, ‘pitch’ and ‘formants’ have been utilized to develop the system. The system is speaker-independent and stands on Template based approach. The recognition method applied here is very simple and the recognition accuracy is also very satisfactory.
Automatic Speech Recognition of Malayalam Language Nasal Class PhonemesEditor IJCATR
Speech recognition applications are becoming common and useful in this day and age as many of the modern devices are designed and produced user-friendly for the convenience of general public. Speaking/communicating directly with the machine to achieve desired objectives make usage of modern devices easier and convenient. Although may interactive software applications are available, the use of these applications is limited due to language barriers. Hence development of speech recognition systems in local languages will help anyone to make use of this technological advancement. In this paper we discuss the results of the Nasal phonetic class wise speech recognition performance of Malayalam language
A Context-based Numeral Reading Technique for Text to Speech Systems IJECEIAES
This paper presents a novel technique for context based numeral reading in Indian language text to speech systems. The model uses a set of rules to determine the context of the numeral pronunciation and is being integrated with the waveform concatenation technique to produce speech out of the input text in Indian languages. For this purpose, the three Indian languages Odia, Hindi and Bengali are considered. To analyze the performance of the proposed technique, a set of experiments are performed considering different context of numeral pronunciations and the results are compared with existing syllable-based technique. The results obtained from different experiments shows the effectiveness of the proposed technique in producing intelligible speech out of the entered text utterances compared to the existing technique even with very less storage and execution time.
DETECTION OF AUTOMATIC THE VOT VALUE FOR VOICED STOP SOUNDS IN MODERN STANDAR...cscpconf
Signal processing in current days is under studying. One of these studies focuses on speech processing. Speech signal have many important features. One of them is Voice Onset Time (VOT). This feature only appears in stop sounds. The human auditory system can utilize the VOT to differentiate between voiced and unvoiced stops like /p/ and /b/ in the English language. By VOT feature we can classify and detect languages and dialects. The main reason behind choosing this subject is that the researches in analyzing Arabic language in this field are not enough and automatic detection of VOT value in Modern Standard Arabic (MSA) is a new idea.
In this paper, we will focus on designing an algorithm that will be used to detect the VOT value in MSA language automatically depending on the power signal. We apply this algorithm only on the voiced stop sounds /b/, /d/ and /d?/, and compare that VOT values automatically generated by the algorithm with the manual values calculated by reading the spectrogram. We created the corpus, and used CV-CV-CV format for each word, the target stop consonant is in the middle of word. The algorithm resulted in a high accuracy, and the error rate was 0.80%, 26.62% and 11.71% for the three stop voiced sounds /d/, /d? / and /b/ respectively . The standard deviation was low in /d/ sound because it is easy to pronounce, and high in /d?/ sound because it is uniqueand difficult to pronounce.
INTEGRATION OF PHONOTACTIC FEATURES FOR LANGUAGE IDENTIFICATION ON CODE-SWITC...kevig
In this paper, phoneme sequences are used as language information to perform code-switched language
identification (LID). With the one-pass recognition system, the spoken sounds are converted into
phonetically arranged sequences of sounds. The acoustic models are robust enough to handle multiple
languages when emulating multiple hidden Markov models (HMMs). To determine the phoneme similarity
among our target languages, we reported two methods of phoneme mapping. Statistical phoneme-based
bigram language models (LM) are integrated into speech decoding to eliminate possible phone
mismatches. The supervised support vector machine (SVM) is used to learn to recognize the phonetic
information of mixed-language speech based on recognized phone sequences. As the back-end decision is
taken by an SVM, the likelihood scores of segments with monolingual phone occurrence are used to
classify language identity. The speech corpus was tested on Sepedi and English languages that are often
mixed. Our system is evaluated by measuring both the ASR performance and the LID performance
separately. The systems have obtained a promising ASR accuracy with data-driven phone merging
approach modelled using 16 Gaussian mixtures per state. In code-switched speech and monolingual
speech segments respectively, the proposed systems achieved an acceptable ASR and LID accuracy.
Speech is the important mode of communication and is the current research
topic. The concentration is mostly focused on synthesis and analyzing part.Apart of
synthesizing, text to speech system is developed.Speech synthesis is an artificial production
of human speech.A text to speech system (TTS) is to convert an arbitrary text into speech.In
India different languages have been spoken each being the mother tongue of tens of millions
of people.In this paper,the text to speech system is primarily developed for Telugu, a
Dravidian language predominantly spoken in Indian state of Andhra Pradesh.The
important qualities expected from this system are naturalness and intelligibility.Telugu TTS
can be developed using other synthesis methods like articulatory synthesis,formant synthesis
and concatenative synthesis.This paper describes a development of a Telugu text to speech
system using concatenative synthesis method on mobile based system OMAP 3530 (ARM
Cortex A-8 core) in Linux.
S URVEY O N M ACHINE T RANSLITERATION A ND M ACHINE L EARNING M ODELSijnlc
Globalization and growth of Internet users truly demands for almost all internet based applications to
support
l
oca
l l
anguages. Support
of
l
oca
l
l
anguages can be
given in all internet based applications by
means of Machine Transliteration
and
Machine Translation
.
This paper provides the thorough survey on
machine transliteration models and machine learning
approaches
used for machine transliteration
over the
period
of more than two decades
for internationally used languages as well as Indian languages.
Survey
shows that linguistic approach provides better results for the closely related languages and probability
based statistical approaches are good when one of the
languages is phonetic and other is non
-
phonetic.
B
etter accuracy can be achieved only by using Hybrid and Combined models.
Sipij040305SPEECH EVALUATION WITH SPECIAL FOCUS ON CHILDREN SUFFERING FROM AP...sipij
Speech disorders are very complicated in individuals suffering from Apraxia of Speech-AOS. In this paper ,
the pathological cases of speech disabled children affected with AOS are analyzed. The speech signal
samples of children of age between three to eight years are considered for the present study. These speech
signals are digitized and enhanced using the using the Speech Pause Index, Jitter,Skew ,Kurtosis analysis
This analysis is conducted on speech data samples which are concerned with both place of articulation and
manner of articulation. The speech disability of pathological subjects was estimated using results of above
analysis.
Implementation of Text To Speech for Marathi Language Using Transcriptions Co...IJERA Editor
This research paper presents the approach towards converting text to speech using new methodology. The text to
speech conversion system enables user to enter text in Marathi and as output it gets sound. The paper presents
the steps followed for converting text to speech for Marathi language and the algorithm used for it. The focus of
this paper is based on the tokenisation process and the orthographic representation of the text that shows the
mapping of letter to sound using the description of language’s phonetics. Here the main focus is on the text to
IPA transcription concept. It is in fact, a system that translates text to IPA transcription which is the primary
stage for text to speech conversion. The whole procedure for converting text to speech involves a great deal of
time as it’s not an easy task and requires efforts.
Artificially Generatedof Concatenative Syllable based Text to Speech Synthesi...iosrjce
IOSR journal of VLSI and Signal Processing (IOSRJVSP) is a double blind peer reviewed International Journal that publishes articles which contribute new results in all areas of VLSI Design & Signal Processing. The goal of this journal is to bring together researchers and practitioners from academia and industry to focus on advanced VLSI Design & Signal Processing concepts and establishing new collaborations in these areas.Design and realization of microelectronic systems using VLSI/ULSI technologies require close collaboration among scientists and engineers in the fields of systems architecture, logic and circuit design, chips and wafer fabrication, packaging, testing and systems applications. Generation of specifications, design and verification must be performed at all abstraction levels, including the system, register-transfer, logic, circuit, transistor and process levels.
Transliteration by orthography or phonology for hindi and marathi to english ...ijnlc
e-Governance and Web based online commercial multilingual applications has given utmost importance to
the task of translation and transliteration. The Named Entities and Technical Terms occur in the source
language of translation are called out of vocabulary words as they are not available in the multilingual
corpus or dictionary used to support translation process. These Named Entities and Technical Terms need
to be transliterated from source language to target language without losing their phonetic properties. The
fundamental problem in India is that there is no set of rules available to write the spellings in English for
Indian languages according to the linguistics. People are writing different spellings for the same name at
different places. This fact certainly affects the Top-1 accuracy of the transliteration and in turn the
translation process. Major issue noticed by us is the transliteration of named entities consisting three
syllables or three phonetic units in Hindi and Marathi languages where people use mixed approach to
write the spelling either by orthographical approach or by phonological approach. In this paper authors
have provided their opinion through experimentation about appropriateness of either approach.
Monitoring and feedback in the process of language acquisition analysis and ...ijnlc
Previously studies have shown that native Japanese and English speakers, constantly monitor their speech,
provide feedback and then correct. Japanese and English have different word orders which make speakers
of both languages monitor their own speech, give feedback and make corrections at different key surface
points. However, structurally, speakers from both languages check their speech and make correction at the
complementizers.1 From there, they continue to produce sentences. As a result, we may say that in order to
efficientlyproduce sentences ( in time and energy ),native Japanese and English speakerscheck their speech
and correct it at the sentence level.
Segmentation Words for Speech Synthesis in Persian Language Based On Silencepaperpublications3
Abstract: In speech synthesis in text to speech systems, the words usually break to different parts and use from recorded sound of each part for play words. This paper use silent in word's pronunciation for better quality of speech. Most algorithms divide words to syllable and some of them divide words to phoneme, but This paper benefit from silent in intonation and divide words at silent region and then set equivalent sound of each parts whereupon joining the parts is trusty and speech quality being more smooth . this paper concern Persian language but extendable to another language. This method has been tested with MOS test and intelligibility, naturalness and fluidity are better.
Keywords:TTS, SBS, Sillable, Diphone.
DYNAMIC PHONE WARPING – A METHOD TO MEASURE THE DISTANCE BETWEEN PRONUNCIATIONS cscpconf
Human beings generate different speech waveforms while speaking the same word at different times. Also, different human beings have different accents and generate significantly varying speech waveforms for the same word. There is a need to measure the distances between various words which facilitate preparation of pronunciation dictionaries. A new algorithm called Dynamic Phone Warping (DPW) is presented in this paper. It uses dynamic programming technique for global alignment and shortest distance measurements. The DPW algorithm can be used to enhance the pronunciation dictionaries of the well-known languages like English or to build pronunciation dictionaries to the less known sparse languages. The precision measurement experiments show 88.9% accuracy.
Research Inventy : International Journal of Engineering and Science is published by the group of young academic and industrial researchers with 12 Issues per year. It is an online as well as print version open access journal that provides rapid publication (monthly) of articles in all areas of the subject such as: civil, mechanical, chemical, electronic and computer engineering as well as production and information technology. The Journal welcomes the submission of manuscripts that meet the general criteria of significance and scientific excellence. Papers will be published by rapid process within 20 days after acceptance and peer review process takes only 7 days. All articles published in Research Inventy will be peer-reviewed.
Research Inventy : International Journal of Engineering and Science is published by the group of young academic and industrial researchers with 12 Issues per year. It is an online as well as print version open access journal that provides rapid publication (monthly) of articles in all areas of the subject such as: civil, mechanical, chemical, electronic and computer engineering as well as production and information technology. The Journal welcomes the submission of manuscripts that meet the general criteria of significance and scientific excellence. Papers will be published by rapid process within 20 days after acceptance and peer review process takes only 7 days. All articles published in Research Inventy will be peer-reviewed.
American Standard Sign Language Representation Using Speech Recognitionpaperpublications3
Abstract: For many deaf people, sign language is the principle means of communication. This increases the isolation of hearing impaired people. This paper presents a system prototype that is able to automatically recognize speech which helps to communicate more effectively with the hearing or speech impaired people. This system recognizes speech signal . Recognized spoken words are represented using American standard sign language via a robotic arm and also on the computer using visual basic .In this project a software package is provided to convert the speech signal, (which does not have any meaning for the deaf and the dumb) into the sign language. The main purpose of this project is to bridge the communication and expression gap between the normal people who cannot understand the sign language, and the deaf and dumb who cannot understand the normal speech.
Automatic Speech Recognition of Malayalam Language Nasal Class PhonemesEditor IJCATR
Speech recognition applications are becoming common and useful in this day and age as many of the modern devices are designed and produced user-friendly for the convenience of general public. Speaking/communicating directly with the machine to achieve desired objectives make usage of modern devices easier and convenient. Although may interactive software applications are available, the use of these applications is limited due to language barriers. Hence development of speech recognition systems in local languages will help anyone to make use of this technological advancement. In this paper we discuss the results of the Nasal phonetic class wise speech recognition performance of Malayalam language
A Context-based Numeral Reading Technique for Text to Speech Systems IJECEIAES
This paper presents a novel technique for context based numeral reading in Indian language text to speech systems. The model uses a set of rules to determine the context of the numeral pronunciation and is being integrated with the waveform concatenation technique to produce speech out of the input text in Indian languages. For this purpose, the three Indian languages Odia, Hindi and Bengali are considered. To analyze the performance of the proposed technique, a set of experiments are performed considering different context of numeral pronunciations and the results are compared with existing syllable-based technique. The results obtained from different experiments shows the effectiveness of the proposed technique in producing intelligible speech out of the entered text utterances compared to the existing technique even with very less storage and execution time.
DETECTION OF AUTOMATIC THE VOT VALUE FOR VOICED STOP SOUNDS IN MODERN STANDAR...cscpconf
Signal processing in current days is under studying. One of these studies focuses on speech processing. Speech signal have many important features. One of them is Voice Onset Time (VOT). This feature only appears in stop sounds. The human auditory system can utilize the VOT to differentiate between voiced and unvoiced stops like /p/ and /b/ in the English language. By VOT feature we can classify and detect languages and dialects. The main reason behind choosing this subject is that the researches in analyzing Arabic language in this field are not enough and automatic detection of VOT value in Modern Standard Arabic (MSA) is a new idea.
In this paper, we will focus on designing an algorithm that will be used to detect the VOT value in MSA language automatically depending on the power signal. We apply this algorithm only on the voiced stop sounds /b/, /d/ and /d?/, and compare that VOT values automatically generated by the algorithm with the manual values calculated by reading the spectrogram. We created the corpus, and used CV-CV-CV format for each word, the target stop consonant is in the middle of word. The algorithm resulted in a high accuracy, and the error rate was 0.80%, 26.62% and 11.71% for the three stop voiced sounds /d/, /d? / and /b/ respectively . The standard deviation was low in /d/ sound because it is easy to pronounce, and high in /d?/ sound because it is uniqueand difficult to pronounce.
INTEGRATION OF PHONOTACTIC FEATURES FOR LANGUAGE IDENTIFICATION ON CODE-SWITC...kevig
In this paper, phoneme sequences are used as language information to perform code-switched language
identification (LID). With the one-pass recognition system, the spoken sounds are converted into
phonetically arranged sequences of sounds. The acoustic models are robust enough to handle multiple
languages when emulating multiple hidden Markov models (HMMs). To determine the phoneme similarity
among our target languages, we reported two methods of phoneme mapping. Statistical phoneme-based
bigram language models (LM) are integrated into speech decoding to eliminate possible phone
mismatches. The supervised support vector machine (SVM) is used to learn to recognize the phonetic
information of mixed-language speech based on recognized phone sequences. As the back-end decision is
taken by an SVM, the likelihood scores of segments with monolingual phone occurrence are used to
classify language identity. The speech corpus was tested on Sepedi and English languages that are often
mixed. Our system is evaluated by measuring both the ASR performance and the LID performance
separately. The systems have obtained a promising ASR accuracy with data-driven phone merging
approach modelled using 16 Gaussian mixtures per state. In code-switched speech and monolingual
speech segments respectively, the proposed systems achieved an acceptable ASR and LID accuracy.
Speech is the important mode of communication and is the current research
topic. The concentration is mostly focused on synthesis and analyzing part.Apart of
synthesizing, text to speech system is developed.Speech synthesis is an artificial production
of human speech.A text to speech system (TTS) is to convert an arbitrary text into speech.In
India different languages have been spoken each being the mother tongue of tens of millions
of people.In this paper,the text to speech system is primarily developed for Telugu, a
Dravidian language predominantly spoken in Indian state of Andhra Pradesh.The
important qualities expected from this system are naturalness and intelligibility.Telugu TTS
can be developed using other synthesis methods like articulatory synthesis,formant synthesis
and concatenative synthesis.This paper describes a development of a Telugu text to speech
system using concatenative synthesis method on mobile based system OMAP 3530 (ARM
Cortex A-8 core) in Linux.
S URVEY O N M ACHINE T RANSLITERATION A ND M ACHINE L EARNING M ODELSijnlc
Globalization and growth of Internet users truly demands for almost all internet based applications to
support
l
oca
l l
anguages. Support
of
l
oca
l
l
anguages can be
given in all internet based applications by
means of Machine Transliteration
and
Machine Translation
.
This paper provides the thorough survey on
machine transliteration models and machine learning
approaches
used for machine transliteration
over the
period
of more than two decades
for internationally used languages as well as Indian languages.
Survey
shows that linguistic approach provides better results for the closely related languages and probability
based statistical approaches are good when one of the
languages is phonetic and other is non
-
phonetic.
B
etter accuracy can be achieved only by using Hybrid and Combined models.
Sipij040305SPEECH EVALUATION WITH SPECIAL FOCUS ON CHILDREN SUFFERING FROM AP...sipij
Speech disorders are very complicated in individuals suffering from Apraxia of Speech-AOS. In this paper ,
the pathological cases of speech disabled children affected with AOS are analyzed. The speech signal
samples of children of age between three to eight years are considered for the present study. These speech
signals are digitized and enhanced using the using the Speech Pause Index, Jitter,Skew ,Kurtosis analysis
This analysis is conducted on speech data samples which are concerned with both place of articulation and
manner of articulation. The speech disability of pathological subjects was estimated using results of above
analysis.
Implementation of Text To Speech for Marathi Language Using Transcriptions Co...IJERA Editor
This research paper presents the approach towards converting text to speech using new methodology. The text to
speech conversion system enables user to enter text in Marathi and as output it gets sound. The paper presents
the steps followed for converting text to speech for Marathi language and the algorithm used for it. The focus of
this paper is based on the tokenisation process and the orthographic representation of the text that shows the
mapping of letter to sound using the description of language’s phonetics. Here the main focus is on the text to
IPA transcription concept. It is in fact, a system that translates text to IPA transcription which is the primary
stage for text to speech conversion. The whole procedure for converting text to speech involves a great deal of
time as it’s not an easy task and requires efforts.
Artificially Generatedof Concatenative Syllable based Text to Speech Synthesi...iosrjce
IOSR journal of VLSI and Signal Processing (IOSRJVSP) is a double blind peer reviewed International Journal that publishes articles which contribute new results in all areas of VLSI Design & Signal Processing. The goal of this journal is to bring together researchers and practitioners from academia and industry to focus on advanced VLSI Design & Signal Processing concepts and establishing new collaborations in these areas.Design and realization of microelectronic systems using VLSI/ULSI technologies require close collaboration among scientists and engineers in the fields of systems architecture, logic and circuit design, chips and wafer fabrication, packaging, testing and systems applications. Generation of specifications, design and verification must be performed at all abstraction levels, including the system, register-transfer, logic, circuit, transistor and process levels.
Transliteration by orthography or phonology for hindi and marathi to english ...ijnlc
e-Governance and Web based online commercial multilingual applications has given utmost importance to
the task of translation and transliteration. The Named Entities and Technical Terms occur in the source
language of translation are called out of vocabulary words as they are not available in the multilingual
corpus or dictionary used to support translation process. These Named Entities and Technical Terms need
to be transliterated from source language to target language without losing their phonetic properties. The
fundamental problem in India is that there is no set of rules available to write the spellings in English for
Indian languages according to the linguistics. People are writing different spellings for the same name at
different places. This fact certainly affects the Top-1 accuracy of the transliteration and in turn the
translation process. Major issue noticed by us is the transliteration of named entities consisting three
syllables or three phonetic units in Hindi and Marathi languages where people use mixed approach to
write the spelling either by orthographical approach or by phonological approach. In this paper authors
have provided their opinion through experimentation about appropriateness of either approach.
Monitoring and feedback in the process of language acquisition analysis and ...ijnlc
Previously studies have shown that native Japanese and English speakers, constantly monitor their speech,
provide feedback and then correct. Japanese and English have different word orders which make speakers
of both languages monitor their own speech, give feedback and make corrections at different key surface
points. However, structurally, speakers from both languages check their speech and make correction at the
complementizers.1 From there, they continue to produce sentences. As a result, we may say that in order to
efficientlyproduce sentences ( in time and energy ),native Japanese and English speakerscheck their speech
and correct it at the sentence level.
Segmentation Words for Speech Synthesis in Persian Language Based On Silencepaperpublications3
Abstract: In speech synthesis in text to speech systems, the words usually break to different parts and use from recorded sound of each part for play words. This paper use silent in word's pronunciation for better quality of speech. Most algorithms divide words to syllable and some of them divide words to phoneme, but This paper benefit from silent in intonation and divide words at silent region and then set equivalent sound of each parts whereupon joining the parts is trusty and speech quality being more smooth . this paper concern Persian language but extendable to another language. This method has been tested with MOS test and intelligibility, naturalness and fluidity are better.
Keywords:TTS, SBS, Sillable, Diphone.
DYNAMIC PHONE WARPING – A METHOD TO MEASURE THE DISTANCE BETWEEN PRONUNCIATIONS cscpconf
Human beings generate different speech waveforms while speaking the same word at different times. Also, different human beings have different accents and generate significantly varying speech waveforms for the same word. There is a need to measure the distances between various words which facilitate preparation of pronunciation dictionaries. A new algorithm called Dynamic Phone Warping (DPW) is presented in this paper. It uses dynamic programming technique for global alignment and shortest distance measurements. The DPW algorithm can be used to enhance the pronunciation dictionaries of the well-known languages like English or to build pronunciation dictionaries to the less known sparse languages. The precision measurement experiments show 88.9% accuracy.
Research Inventy : International Journal of Engineering and Science is published by the group of young academic and industrial researchers with 12 Issues per year. It is an online as well as print version open access journal that provides rapid publication (monthly) of articles in all areas of the subject such as: civil, mechanical, chemical, electronic and computer engineering as well as production and information technology. The Journal welcomes the submission of manuscripts that meet the general criteria of significance and scientific excellence. Papers will be published by rapid process within 20 days after acceptance and peer review process takes only 7 days. All articles published in Research Inventy will be peer-reviewed.
Research Inventy : International Journal of Engineering and Science is published by the group of young academic and industrial researchers with 12 Issues per year. It is an online as well as print version open access journal that provides rapid publication (monthly) of articles in all areas of the subject such as: civil, mechanical, chemical, electronic and computer engineering as well as production and information technology. The Journal welcomes the submission of manuscripts that meet the general criteria of significance and scientific excellence. Papers will be published by rapid process within 20 days after acceptance and peer review process takes only 7 days. All articles published in Research Inventy will be peer-reviewed.
American Standard Sign Language Representation Using Speech Recognitionpaperpublications3
Abstract: For many deaf people, sign language is the principle means of communication. This increases the isolation of hearing impaired people. This paper presents a system prototype that is able to automatically recognize speech which helps to communicate more effectively with the hearing or speech impaired people. This system recognizes speech signal . Recognized spoken words are represented using American standard sign language via a robotic arm and also on the computer using visual basic .In this project a software package is provided to convert the speech signal, (which does not have any meaning for the deaf and the dumb) into the sign language. The main purpose of this project is to bridge the communication and expression gap between the normal people who cannot understand the sign language, and the deaf and dumb who cannot understand the normal speech.
Hybrid Phonemic and Graphemic Modeling for Arabic Speech RecognitionWaqas Tariq
In this research, we propose a hybrid approach for acoustic and pronunciation modeling for Arabic speech recognition. The hybrid approach benefits from both vocalized and non-vocalized Arabic resources, based on the fact that the amount of non-vocalized resources is always higher than vocalized resources. Two speech recognition baseline systems were built: phonemic and graphemic. The two baseline acoustic models were fused together after two independent trainings to create a hybrid acoustic model. Pronunciation modeling was also hybrid by generating graphemic pronunciation variants as well as phonemic variants. Different techniques are proposed for pronunciation modeling to reduce model complexity. Experiments were conducted on large vocabulary news broadcast speech domain. The proposed hybrid approach has shown a relative reduction in WER of 8.8% to 12.6% based on pronunciation modeling settings and the supervision in the baseline systems.
Upgrading the Performance of Speech Emotion Recognition at the Segmental Level IOSR Journals
Abstract: This paper presents an efficient approach for maximizing the accuracy of automatic speech emotion
recognition in English, using minimal inputs, minimal features, lesser algorithmic complexity and reduced
processing time. Whereas the findings reported here are based on the exclusive use of vowel formants, most of
the related previous works used tens or even hundreds of other features. In spite of using a greater level of
signal processing, the recognition accuracy reported earlier was often lesser than that obtained by our
approach. This method is based on vowel utterances and the first step comprises statistical pre-processing of
the vowel formants. This is followed by the identification of the best formants using the KMeans, K-nearest
neighbor and Naive Bayes classifiers. The Artificial neural network that was used for the final classification
gave an accuracy of 95.6% on elicited emotional speech. Nearly 1500 speech files from ten female speakers in
the neutral and six basic emotions were used to prove the efficiency of the proposed approach. Such a result
has not been reported earlier for English and is of significance to researchers, sociologists and others interested in speech.
Keywords: Artificial Neural Networks, Emotions, Formants, Preprocessing,Vowels.
English speaking proficiency assessment using speech and electroencephalograp...IJECEIAES
In this paper, the English speaking proficiency level of non-native English speakers was automatically estimated as high, medium, or low performance. For this purpose, the speech of 142 non-native English speakers was recorded and electroencephalography (EEG) signals of 58 of them were recorded while speaking in English. Two systems were proposed for estimating the English proficiency level of the speaker; one used 72 audio features, extracted from speech signals, and the other used 112 features extracted from EEG signals. Multi-class support vector machines (SVM) was used for training and testing both systems using a cross-validation strategy. The speech-based system outperformed the EEG system with 68% accuracy on 60 testing audio recordings, compared with 56% accuracy on 30 testing EEG recordings.
PERFORMANCE ANALYSIS OF DIFFERENT ACOUSTIC FEATURES BASED ON LSTM FOR BANGLA ...ijma
In this work a new Bangla speech corpus along with proper transcriptions has been developed; also
various acoustic feature extraction methods have been investigated using Long Short-Term Memory
(LSTM) neural network to find their effective integration into a state-of-the-art Bangla speech recognition
system. The acoustic features are usually a sequence of representative vectors that are extracted from
speech signals and the classes are either words or sub word units such as phonemes. The most commonly
used feature extraction method, known as linear predictive coding (LPC), has been used first in this work.
Then the other two popular methods, namely, the Mel frequency cepstral coefficients (MFCC) and
perceptual linear prediction (PLP) have also been applied. These methods are based on the models of the
human auditory system. A detailed review of the implementation of these methods have been described
first. Then the steps of the implementation have been elaborated for the development of an automatic
speech recognition system (ASR) for Bangla speech.
PERFORMANCE ANALYSIS OF DIFFERENT ACOUSTIC FEATURES BASED ON LSTM FOR BANGLA ...ijma
In this work a new Bangla speech corpus along with proper transcriptions has been developed; also
various acoustic feature extraction methods have been investigated using Long Short-Term Memory
(LSTM) neural network to find their effective integration into a state-of-the-art Bangla speech recognition
system. The acoustic features are usually a sequence of representative vectors that are extracted from
speech signals and the classes are either words or sub word units such as phonemes. The most commonly
used feature extraction method, known as linear predictive coding (LPC), has been used first in this work.
Then the other two popular methods, namely, the Mel frequency cepstral coefficients (MFCC) and
perceptual linear prediction (PLP) have also been applied. These methods are based on the models of the
human auditory system. A detailed review of the implementation of these methods have been described
first. Then the steps of the implementation have been elaborated for the development of an automatic
speech recognition system (ASR) for Bangla speech.
Energy distribution in formant bands for arabic vowelsIJECEIAES
The acoustic cues play a major role in speech segmentation phase; the extraction of these indexes facilitates the characterization of the speech signal. In this work, we aim to study Arabic vowels (/a/, /a:/, /i/, /i:/, /u/ and /u:/), especially the long ones. We are interested in characterizing this type of vowels in terms of time, frequency and energy. The cues extracted and analyzed in this work are: segment length, voicing degree and formants values.
High Level Speaker Specific Features as an Efficiency Enhancing Parameters in...IJECEIAES
In this paper, I present high-level speaker specific feature extraction considering intonation, linguistics rhythm, linguistics stress, prosodic features directly from speech signals. I assume that the rhythm is related to language units such as syllables and appears as changes in measurable parameters such as fundamental frequency ( ), duration, and energy. In this work, the syllable type features are selected as the basic unit for expressing the prosodic features. The approximate segmentation of continuous speech to syllable units is achieved by automatically locating the vowel starting point. The knowledge of high-level speaker’s specific speakers is used as a reference for extracting the prosodic features of the speech signal. High-level speaker-specific features extracted using this method may be useful in applications such as speaker recognition where explicit phoneme/syllable boundaries are not readily available. The efficiency of the particular characteristics of the specific features used for automatic speaker recognition was evaluated on TIMIT and HTIMIT corpora initially sampled in the TIMIT at 16 kHz to 8 kHz. In summary, the experiment, the basic discriminating system, and the HMM system are formed on TIMIT corpus with a set of 48 phonemes. Proposed ASR system shows 1.99%, 2.10%, 2.16% and 2.19 % of efficiency improvements compared to traditional ASR system for and of 16KHz TIMIT utterances.
G2 pil a grapheme to-phoneme conversion tool for the italian languageijnlc
This paper presents a knowledge-based approach for the grapheme to-phoneme conversion (G2P) of isolated words of the Italian language. With more than 7,000 languages in the world, the biggest challenge today is to rapidly port speech processing systems to new languages with low human effort and at reasonable cost. This includes the creation of qualified pronunciation dictionaries. The dictionaries provide the mapping from the orthographic form of a word to its pronunciation, which is useful in both speech synthesis and automatic speech recognition (ASR) systems. For training the acoustic models we need an automatic routine that maps the spelling of training set to a string of phonetic symbols representing the pronunciation.
Efficiency of the energy contained in modulators in the Arabic vowels recogni...IJECEIAES
The speech signal is described as many acoustic properties that may contribute differently to spoken word recognition. Vowel characterization is an important process of studying the acoustic characteristics or behaviors of speech within different contexts. This current study focuses on the modulators characteristics of three Arabic vowels, we proposed a new approach to characterize the three Arabic vowels /a/, /i/ and /u/. The proposed method is based on the energy contained in the speech modulators. The coherent subband demodulation method related to the spectral center of gravity (COG) was used to calculate the energy of the speech modulators. The obtained results showed that the modulators energy help characterize the Arabic vowels /a/, /i/ and /u/ with an interesting recognition rate ranging from 86% to 100%.
Research Inventy : International Journal of Engineering and Scienceinventy
Research Inventy : International Journal of Engineering and Science
Research Inventy : International Journal of Engineering and Science is published by the group of young academic and industrial researchers with 12 Issues per year. It is an online as well as print version open access journal that provides rapid publication (monthly) of articles in all areas of the subject such as: civil, mechanical, chemical, electronic and computer engineering as well as production and information technology. The Journal welcomes the submission of manuscripts that meet the general criteria of significance and scientific excellence. Papers will be published by rapid process within 20 days after acceptance and peer review process takes only 7 days. All articles published in Research Inventy will be peer-reviewed
High Quality Arabic Concatenative Speech Synthesissipij
This paper describes the implementation of TD-PSOLA tools to improve the quality of the Arabic Text-tospeech (TTS) system. This system based on Diphone concatenation with TD-PSOLA modifier synthesizer. This paper describes techniques to improve the precision of prosodic modifications in the Arabic speech synthesis using the TD-PSOLA (Time Domain Pitch Synchronous Overlap-Add) method. This approach is based on the decomposition of the signal into overlapping frames synchronized with the pitch period. The main objective is to preserve the consistency and accuracy of the pitch marks after prosodic modifications of the speech signal and diphone with vowel integrated database adjustment and optimisation.
Similar to Speech Feature Extraction and Data Visualisation (20)
Securing Cloud Computing Through IT GovernanceITIIIndustries
Lack of alignment between information technology (IT) and the business is a problem facing many organizations. Most organizations, today, fundamentally depend on IT. When IT and the business are aligned in an organization, IT delivers what the business needs and the business is able to deliver what the market needs. IT has become a strategic function for most organizations, and it is imperative that IT and business are aligned. IT governance is one of the most powerful ways to achieve IT to business alignment. Furthermore, as the use of cloud computing for delivering IT functions becomes pervasive, organizations using cloud computing must effectively apply IT governance to it. While cloud computing presents tremendous
opportunities, it comes with risks as well. Information security
is one of the top risks in cloud computing. Thus, IT governance must be applied to cloud computing information security to help manage the risks associated with cloud computing information security. This study advances knowledge by extending IT governance to cloud computing and information security governance.
Information Technology in Industry(ITII) - November Issue 2018ITIIIndustries
IT Industry publishes original research articles, review articles, and extended versions of conference papers. Articles resulting from research of both theoretical and/or practical natures performed by academics and/or industry practitioners are welcome. IT in Industry aims to become a leading IT journal with a high impact factor.
Design of an IT Capstone Subject - Cloud RoboticsITIIIndustries
This paper describes the curriculum of the three year IT undergraduate program at La Trobe University, and the faculty requirements in designing a capstone subject, followed by the ACM’s recommended IT curriculum covering the five pillars of the IT discipline. Cloud robotics, a broad multidisciplinary research area, requiring expertise in all five pillars with mechatronics, is an ideal candidate to offer capstone experiences to IT students. Therefore, in this paper, we propose a long term
master project in developing a cloud robotics testbed, with many capstone sub-projects spanning across the five IT pillars, to meet the objectives of capstone experience. This paper also describes the design and implementation of the testbed, and proposes potential capstone projects for students with different interests.
Dimensionality Reduction and Feature Selection Methods for Script Identificat...ITIIIndustries
The goal of this research is to explore effects of dimensionality reduction and feature selection on the problem of script identification from images of printed documents. The kadjacent segment is ideal for this use due to its ability to capture visual patterns. We have used principle component analysis to reduce the size of our feature matrix to a handier size that can be trained easily, and experimented by including varying combinations of dimensions of the super feature set. A modular
approach in neural network was used to classify 7 languages – Arabic, Chinese, English, Japanese, Tamil, Thai and Korean.
Image Matting via LLE/iLLE Manifold LearningITIIIndustries
Accurately extracting foreground objects is the problem of isolating the foreground in images and video, called image matting which has wide applications in digital photography. This problem is severely ill-posed in the sense that, at each pixel, one must estimate the foreground and background pixels and the so-called alpha value from only pixel information. The most recent work in natural image matting rely on local smoothness assumptions about foreground and background colours on which a cost function has been established. In this paper, we propose an extension to the class of affinity based matting techniques by incorporating local manifold structural
information to produce both a smoother matte based on the socalled improved Locally Linear Embedding. We illustrate our new algorithm using the standard benchmark images and very comparable results have been obtained.
Annotating Retina Fundus Images for Teaching and Learning Diabetic Retinopath...ITIIIndustries
With the improvement in IT industry, more and more application of computer software is introduced in teaching and learning. In this paper, we discuss the development process of such software. Diabetic Retinopathy is a common complication for diabetic patients. It may cause sight loss if not treated early. There are several stages of this disease. Fundus imagery is required to identify the stage and severity of the disease. Due to the lack of proper dataset of the fundus images and proper annotation, it is very difficult to perform research on this topic. Moreover, medical students are often facing difficulty with identifying the diseases in later stage of their practice as they may not have seen a sample of all of the stages of Diabetic Retinopathy problems. To mitigate the problem, we have collected fundus images from different geographic area of Bangladesh and designed an annotation software to store information about the patient, the infection level and their locations in the images. Sometimes, it is difficult to select all appropriate pixels of the infected region. To resolve the issue, we have introduced a K nearest neighbor (KNN) based technique to accurately select the region of interest (ROI). Once an expert (ophthalmologist) has annotated the images, the software can be used by the students for learning.
A Framework for Traffic Planning and Forecasting using Micro-Simulation Calib...ITIIIndustries
This paper presents the application of microsimulation for traffic planning and forecasting, and proposes a new framework to model complex traffic conditions by calibrating and adjusting traffic parameters of a microsimulation model. By using an open source micro-simulator package, TRANSIMS, in this study, animated and numerical results were produced and analysed. The framework of traffic model calibration was evaluated for its usefulness and practicality. Finally, we discuss future applications such as providing end users with real time traffic information through Intelligent Transport System (ITS) integration.
Investigating Tertiary Students’ Perceptions on Internet SecurityITIIIndustries
Internet security threats have grown from just simple viruses to various forms of computer hacking, scams, impersonation, cyber bullying, and spyware. The Internet has great influence on most people. It has profound influence and one can spend endless hours on internet activities. In particular, youth engage in more online activities than any other age group. Excessive internet usage is an emerging threat that has negative impacts on these youth; hence it is vital to investigate youths' online behavior. This work studies tertiary students’ risk awareness, and provides some findings that allow us to understand their knowledge on risks and their behavior towards online activities. It reveals several important online issues amongst tertiary students; Firstly, the lack of online security awareness; second, a lack of awareness and information about the dangers of rootkits, internet cookies and spyware; thirdly, female students are more unflinching than male students when commenting on social networking sites; fourthly, students are cautious only when obvious security warnings are present; and finally, their usage of internet hotspots is common without fully understanding its associated danger. These findings enable us to recommend types of internet security habits and safety practices that students should adopt in future when they are exposed to online activities. A more holistic approach was considered which aims to minimize any future risks and dangers with online activities involving students.
Blind Image Watermarking Based on Chaotic MapsITIIIndustries
Security of a watermark refers to its resistance to unauthorized detecting and decoding, while watermark robustness refers to the watermark’s resistance against common processing. Many watermarking schemes emphasize robustness more than security. However, a robust watermark is not enough to accomplish protection because the range of hostile attacks is not limited to common processing and distortions. In this paper, we give consideration to watermark security. To achieve this, we employ chaotic maps due to their extreme sensitivity to the initial values. If one fails to provide these values, the watermark will be wrongly extracted. While the chaotic maps provide perfect watermarking security, the proposed scheme is also intended to achieve robustness.
Programmatic detection of spatial behaviour in an agent-based modelITIIIndustries
The automated detection of aspects of spatial behaviour in an agent-based model is necessary for model testing and analysis. In this paper we compare four predictors of herding behaviour in a model of a grazing herbivore. We find that a) the mean number of neighbours adjusted to account for population variation and b) the mean Hamming distance between rows of the two-dimensional environment can be used to detect herding. Visual inspection of the model behaviour revealed that herding occurs when the herbivore mobility reaches a threshold level. Using this threshold we identify a limits for these predictors to use in the program code. These results apply only to one set of parameters and environment size; future research will involve a wider parameter space.
Design of an IT Capstone Subject - Cloud RoboticsITIIIndustries
This paper describes the curriculum of the three year IT undergraduate program at La Trobe University, and the faculty requirements in designing a capstone subject, followed by the ACM’s recommended IT curriculum covering the five pillars of the IT discipline. Cloud robotics, a broad multidisciplinary research area, requiring expertise in all five pillars with mechatronics, is an ideal candidate to offer capstone experiences to IT students. Therefore, in this paper, we propose a long term master project in developing a cloud robotics testbed, with many capstone sub-projects spanning across the five IT pillars, to meet the objectives of capstone experience. This paper also describes the design and implementation of the testbed, and proposes potential capstone projects for students with different interests.
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Fuzzing is one of the most commonly used methods to detect software vulnerabilities, a major cause of information security incidents. Although it has advantages of simple design and low error report, its efficiency is usually poor. In this paper we present a smart fuzzing approach for integer overflow detection and a tool, SwordFuzzer, which implements this approach. Unlike standard fuzzing techniques, which randomly change parts of the input file with no information about the underlying syntactic structure of the file, SwordFuzzer uses online dynamic taint analysis to identify which bytes in the input file are used in security sensitive operations and then focuses on mutating such bytes. Thus, the generated inputs are more likely to trigger potential vulnerabilities. We evaluated SwordFuzzer with an example program and a number of real-world applications. The experimental results show that SwordFuzzer can accurately locate the key bytes of the input file and dramatically improve the effectiveness of fuzzing in detecting real-world vulnerabilities
The banking experience for many people today is fundamentally an application of technology to be able to carry out their financial tasks. While the need to visit a bank branch remains essential for a number of activities, increasingly the need to support mobile usage is becoming the central focus of many bank strategies. The core banking systems that process financial transactions must remain highly available and able to support large volumes of activity. These systems represent a long term investment for banks and when the need arises to modernize these large systems, the transformation initiative is often very expensive and of high risk. We present in this paper our experiences in bank modernization and transformation, and outline the strategies for rolling out these large programs. As banking institutions embark upon transformation programs to upgrade their banking channels and core banking systems, it is hoped that the insights presented here are useful as a framework to support these initiatives.
Detecting Fraud Using Transaction Frequency DataITIIIndustries
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Mapping the Cybernetic Principles of Viable System Model to Enterprise Servic...ITIIIndustries
This paper describes the results of a theoretical mapping of the cybernetic principles of the Viable System Model (VSM) to an Enterprise Service Bus (ESB) model, with the aim to identify the management principles for the integration of services at all levels in the enterprise. This enrichment directly contributes to the viability of service-oriented systems and the justification of Business/IT alignment within enterprise. The model was identified to be suitable for further adaption in the industrial setting planned within Australian governmental departments.
Using Watershed Transform for Vision-based Two-Hand Occlusion in an Interacti...ITIIIndustries
To achieve a natural interaction in augmented reality environment, we have suggested to use markerless visionbased two-handed gestures for the interaction; with an outstretched hand and a pointing hand used as virtual object registration plane and pointing device respectively. However, twohanded interaction always causes mutual occlusion which jeopardizes the hand gesture recognition. In this paper, we present a solution for two-hand occlusion by using watershed transform. The main idea is to start from a two-hand occlusion image in binary format, then form a grey-scale image based on the distance of each non-object pixel to object pixel. The watershed algorithm is applied to the negation of the grey scaled image to form watershed lines which separate the two hands. Fingertips are then identified and each hand is recognized based on the number of fingertips on each hand. The outstretched hand is assumed to contain 5 fingertips and the pointing device contains less than 5 fingertips. An example of applying our result in hand and virtual object interaction is displayed at the end of the paper.
Bayesian-Network-Based Algorithm Selection with High Level Representation Fee...ITIIIndustries
A real-world intelligent system consists of three basic modules: environment recognition, prediction (or estimation), and behavior planning. To obtain high quality results in these modules, high speed processing and real time adaptability on a case by case basis are required. In the environment recognition module many different algorithms and algorithm networks exist with varying performance. Thus, a mechanism that selects the best possible algorithm is required. To solve this problem we are using an algorithm selection approach to the problem of natural image understanding. This selection mechanism is based on machine learning; a bottom-up algorithm selection from real-world image features and a top-down algorithm selection using information obtained from a high level symbolic world description and algorithm suitability. The algorithm selection method iterates for each input image until the high-level description cannot be improved anymore. In this paper we present a method of iterative composition of the high level description. This step by step approach allows us to select the best result for each region of the image by evaluating all the intermediary representations and finally keep only the best one.
Instance Selection and Optimization of Neural NetworksITIIIndustries
Credit scoring is an important tool in financial institutions, which can be used in credit granting decision. Credit applications are marked by credit scoring models and those with high marks will be treated as “good”, while those with low marks will be regarded as “bad”. As data mining technique develops, automatic credit scoring systems are warmly welcomed for their high efficiency and objective judgments. Many machine learning algorithms have been applied in training credit scoring models, and ANN is one of them with good performance. This paper presents a higher accuracy credit scoring model based on MLP neural networks trained with back propagation algorithm. Our work focuses on enhancing credit scoring models in three aspects: optimize data distribution in datasets using a new method called Average Random Choosing; compare effects of training-validation-test instances numbers; and find the most suitable number of hidden units. Another contribution of this paper is summarizing the tendency of scoring accuracy of models when the number of hidden units increases. The experiment results show that our methods can achieve high credit scoring accuracy with imbalanced datasets. Thus, credit granting decision can be made by data mining methods using MLP neural networks.
Signature Forgery and the Forger – An Assessment of Influence on Handwritten ...ITIIIndustries
Signatures are widely used as a form of personal authentication. Despite ubiquity in deployment, individual signatures are relatively easy to forge, especially when only the static ‘pictorial’ outcome of the signature is considered at verification time. In this study, we explore opinions on signature usage for verification purposes, and how individuals rate a particular third-party signature in terms of ease of forgeability and their own ability to forge. We examine responses with respect to an individual’s experience of the forgeability/complexity of their own signature. Our study shows that past experience does not generally have an effect on perceived signature complexity nor the perceived effectiveness of an individual to themselves forge a signature. In assessing forgeability, most subjects cite the overall signature complexity and distinguishing features in reaching this decision. Furthermore, our research indicates that individuals typically vary their signature according to the scenario but generally little effort into the production of the signature.
Stability of Individuals in a Fingerprint System across Force LevelsITIIIndustries
This research studied the question: “Are all
individual’s performance stable in a fingerprint recognition
system?” The fingerprints of 154 individuals, provided at
different force levels, were examined using the biometric
menagerie tool, first coined by Doddington et al. in 1998. The
Biometric Menagerie illustrates how each person in a given
dataset performs in a biometric system, by using their genuine
and impostor scores, and providing them a classification based
upon those scores. This research examined the biometric
menagerie classifications across different force levels in a
fingerprint recognition study to uncover if individuals performed
the same over five force levels. The study concluded that they did
not, and a new metric has been created to quantify this
phenomenon. As a result of this discovery, the new metric,
Stability Score Index is described to showcase the movement of
individuals in the menagerie.
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This presentation was delivered at K8SUG Singapore. See https://feryn.eu/presentations/accelerate-your-kubernetes-clusters-with-varnish-caching-k8sug-singapore-28-2024 for more details.
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This talk is aimed at encouraging a more independent approach to using PHP frameworks, moving towards a more flexible and future-proof approach to PHP development.
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