In this paper we report the experiment carried out on recently collected speaker recognition database
namely Arunachali Language Speech Database (ALS-DB)to make a comparative study on the
performance of acoustic and prosodic features for speaker verification task.The speech database consists
of speech data recorded from 200 speakers with Arunachali languages of North-East India as mother
tongue. The collected database is evaluated using Gaussian mixture model-Universal Background Model
(GMM-UBM) based speaker verification system. The acoustic feature considered in the present study is
Mel-Frequency Cepstral Coefficients (MFCC) along with its derivatives.The performance of the system
has been evaluated for both acoustic feature and prosodic feature individually as well as in
combination.It has been observed that acoustic feature, when considered individually, provide better
performance compared to prosodic features. However, if prosodic features are combined with acoustic
feature, performance of the system outperforms both the systems where the features are considered
individually. There is a nearly 5% improvement in recognition accuracy with respect to the system where
acoustic features are considered individually and nearly 20% improvement with respect to the system
where only prosodic features are considered.
Performance Calculation of Speech Synthesis Methods for Hindi languageiosrjce
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
IMPROVING MYANMAR AUTOMATIC SPEECH RECOGNITION WITH OPTIMIZATION OF CONVOLUTI...ijnlc
Researchers of many nations have developed automatic speech recognition (ASR) to show their national improvement in information and communication technology for their languages. This work intends to improve the ASR performance for Myanmar language by changing different Convolutional Neural Network (CNN) hyperparameters such as number of feature maps and pooling size. CNN has the abilities of reducing in spectral variations and modeling spectral correlations that exist in the signal due to the locality and pooling operation. Therefore, the impact of the hyperparameters on CNN accuracy in ASR tasks is investigated. A 42-hr-data set is used as training data and the ASR performance was evaluated on two open
test sets: web news and recorded data. As Myanmar language is a syllable-timed language, ASR based on syllable was built and compared with ASR based on word. As the result, it gained 16.7% word error rate (WER) and 11.5% syllable error rate (SER) on TestSet1. And it also achieved 21.83% WER and 15.76% SER on TestSet2.
SMATalk: Standard Malay Text to Speech Talk SystemCSCJournals
This paper presents a rule-based text- to- speech (TTS) Synthesis System for Standard Malay, namely SMaTTS. The proposed system using sinusoidal method and some pre- recorded wave files in generating speech for the system. The use of phone database significantly decreases the amount of computer memory space used, thus making the system very light and embeddable. The overall system was comprised of two phases the Natural Language Processing (NLP) that consisted of the high-level processing of text analysis, phonetic analysis, text normalization and morphophonemic module. The module was designed specially for SM to overcome few problems in defining the rules for SM orthography system before it can be passed to the DSP module. The second phase is the Digital Signal Processing (DSP) which operated on the low-level process of the speech waveform generation. A developed an intelligible and adequately natural sounding formant-based speech synthesis system with a light and user-friendly Graphical User Interface (GUI) is introduced. A Standard Malay Language (SM) phoneme set and an inclusive set of phone database have been constructed carefully for this phone-based speech synthesizer. By applying the generative phonology, a comprehensive letter-to-sound (LTS) rules and a pronunciation lexicon have been invented for SMaTTS. As for the evaluation tests, a set of Diagnostic Rhyme Test (DRT) word list was compiled and several experiments have been performed to evaluate the quality of the synthesized speech by analyzing the Mean Opinion Score (MOS) obtained. The overall performance of the system as well as the room for improvements was thoroughly discussed.
An expert system for automatic reading of a text written in standard arabicijnlc
In this work we present our expert system of Automatic reading or speech synthesis based on a text
written in Standard Arabic, our work is carried out in two great stages: the creation of the sound data
base, and the transformation of the written text into speech (Text To Speech TTS). This transformation is
done firstly by a Phonetic Orthographical Transcription (POT) of any written Standard Arabic text with
the aim of transforming it into his corresponding phonetics sequence, and secondly by the generation of
the voice signal which corresponds to the chain transcribed. We spread out the different of conception of
the system, as well as the results obtained compared to others works studied to realize TTS based on
Standard Arabic.
Speech to text conversion for visually impaired person using µ law compandingiosrjce
The paper represents the overall design and implementation of DSP based speech recognition and
text conversion system. Speech is usually taken as a preferred mode of operation for human being, This paper
represent voice oriented command for converting into text. We intended to compute the entire speech processing
in real time. This involves simultaneously accepting the input from the user and using software filters to analyse
the data. The comparison was then to be established by using correlation and µ law companding techniques. In
this paper, voice recognition is carried out using MATLAB. The voice command is a person independent. The
voice command is stored in the data base with the help of the function keys. The real time input speech received
is then processed in the speech recognition system where the required feature of the speech words are extracted,
filtered out and matched with the existing sample stored in the database. Then the required MATLAB processes
are done to convert the received data and into text form.
Wavelet Based Feature Extraction for the Indonesian CV Syllables SoundTELKOMNIKA JOURNAL
This paper proposes the combined methods of Wavelet Transform (WT) and Euclidean Distance
(ED) to estimate the expected value of the possibly feature vector of Indonesian syllables. This research
aims to find the best properties in effectiveness and efficiency on performing feature extraction of each
syllable sound to be applied in the speech recognition systems. This proposed approach which is the
state-of-the-art of the previous study consist of three main phase. In the first phase, the speech signal is
segmented and normalized. In the second phase, the signal is transformed into frequency domain by using
the WT. In the third phase, to estimate the expected feature vector, the ED algorithm is used. Th e result
shows the list of features of each syllables can be used for the next research, and some recommendations
on the most effective and efficient WT to be used in performing syllable sound recognition.
Identification of Sex of the Speaker With Reference To Bodo Vowels: A Compara...IJERA Editor
This work presents an application of Fundamental Frequency (Pitch), Linear Predictive Cepstral Coefficient
(LPCC) and Mel Frequency Cepstral Coefficient (MFCC) in identification of sex of the speaker in speech
recognition research. The aim of this article is to compare the performance of these three methods for
identification of sex of the speakers. A successful speech recognition system can help in non critical operations
such as presenting the driving route to the driver, dialing a phone number, light switch turn on/off, the coffee
machine on/off etc. apart from speaker verification-caste wise, community wise and locality wise including
identification of sex. Here an attempt has been made to identify the sex of Bodo speakers through vowel
utterance by following Pitch value, LPCC and MFCC techniques. It is found here that the feature vector
organization of LPCC coefficients provides a more promising way of speech-speaker recognition in case of
Bodo Language than that of Pitch and MFCC.
Performance Calculation of Speech Synthesis Methods for Hindi languageiosrjce
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
IMPROVING MYANMAR AUTOMATIC SPEECH RECOGNITION WITH OPTIMIZATION OF CONVOLUTI...ijnlc
Researchers of many nations have developed automatic speech recognition (ASR) to show their national improvement in information and communication technology for their languages. This work intends to improve the ASR performance for Myanmar language by changing different Convolutional Neural Network (CNN) hyperparameters such as number of feature maps and pooling size. CNN has the abilities of reducing in spectral variations and modeling spectral correlations that exist in the signal due to the locality and pooling operation. Therefore, the impact of the hyperparameters on CNN accuracy in ASR tasks is investigated. A 42-hr-data set is used as training data and the ASR performance was evaluated on two open
test sets: web news and recorded data. As Myanmar language is a syllable-timed language, ASR based on syllable was built and compared with ASR based on word. As the result, it gained 16.7% word error rate (WER) and 11.5% syllable error rate (SER) on TestSet1. And it also achieved 21.83% WER and 15.76% SER on TestSet2.
SMATalk: Standard Malay Text to Speech Talk SystemCSCJournals
This paper presents a rule-based text- to- speech (TTS) Synthesis System for Standard Malay, namely SMaTTS. The proposed system using sinusoidal method and some pre- recorded wave files in generating speech for the system. The use of phone database significantly decreases the amount of computer memory space used, thus making the system very light and embeddable. The overall system was comprised of two phases the Natural Language Processing (NLP) that consisted of the high-level processing of text analysis, phonetic analysis, text normalization and morphophonemic module. The module was designed specially for SM to overcome few problems in defining the rules for SM orthography system before it can be passed to the DSP module. The second phase is the Digital Signal Processing (DSP) which operated on the low-level process of the speech waveform generation. A developed an intelligible and adequately natural sounding formant-based speech synthesis system with a light and user-friendly Graphical User Interface (GUI) is introduced. A Standard Malay Language (SM) phoneme set and an inclusive set of phone database have been constructed carefully for this phone-based speech synthesizer. By applying the generative phonology, a comprehensive letter-to-sound (LTS) rules and a pronunciation lexicon have been invented for SMaTTS. As for the evaluation tests, a set of Diagnostic Rhyme Test (DRT) word list was compiled and several experiments have been performed to evaluate the quality of the synthesized speech by analyzing the Mean Opinion Score (MOS) obtained. The overall performance of the system as well as the room for improvements was thoroughly discussed.
An expert system for automatic reading of a text written in standard arabicijnlc
In this work we present our expert system of Automatic reading or speech synthesis based on a text
written in Standard Arabic, our work is carried out in two great stages: the creation of the sound data
base, and the transformation of the written text into speech (Text To Speech TTS). This transformation is
done firstly by a Phonetic Orthographical Transcription (POT) of any written Standard Arabic text with
the aim of transforming it into his corresponding phonetics sequence, and secondly by the generation of
the voice signal which corresponds to the chain transcribed. We spread out the different of conception of
the system, as well as the results obtained compared to others works studied to realize TTS based on
Standard Arabic.
Speech to text conversion for visually impaired person using µ law compandingiosrjce
The paper represents the overall design and implementation of DSP based speech recognition and
text conversion system. Speech is usually taken as a preferred mode of operation for human being, This paper
represent voice oriented command for converting into text. We intended to compute the entire speech processing
in real time. This involves simultaneously accepting the input from the user and using software filters to analyse
the data. The comparison was then to be established by using correlation and µ law companding techniques. In
this paper, voice recognition is carried out using MATLAB. The voice command is a person independent. The
voice command is stored in the data base with the help of the function keys. The real time input speech received
is then processed in the speech recognition system where the required feature of the speech words are extracted,
filtered out and matched with the existing sample stored in the database. Then the required MATLAB processes
are done to convert the received data and into text form.
Wavelet Based Feature Extraction for the Indonesian CV Syllables SoundTELKOMNIKA JOURNAL
This paper proposes the combined methods of Wavelet Transform (WT) and Euclidean Distance
(ED) to estimate the expected value of the possibly feature vector of Indonesian syllables. This research
aims to find the best properties in effectiveness and efficiency on performing feature extraction of each
syllable sound to be applied in the speech recognition systems. This proposed approach which is the
state-of-the-art of the previous study consist of three main phase. In the first phase, the speech signal is
segmented and normalized. In the second phase, the signal is transformed into frequency domain by using
the WT. In the third phase, to estimate the expected feature vector, the ED algorithm is used. Th e result
shows the list of features of each syllables can be used for the next research, and some recommendations
on the most effective and efficient WT to be used in performing syllable sound recognition.
Identification of Sex of the Speaker With Reference To Bodo Vowels: A Compara...IJERA Editor
This work presents an application of Fundamental Frequency (Pitch), Linear Predictive Cepstral Coefficient
(LPCC) and Mel Frequency Cepstral Coefficient (MFCC) in identification of sex of the speaker in speech
recognition research. The aim of this article is to compare the performance of these three methods for
identification of sex of the speakers. A successful speech recognition system can help in non critical operations
such as presenting the driving route to the driver, dialing a phone number, light switch turn on/off, the coffee
machine on/off etc. apart from speaker verification-caste wise, community wise and locality wise including
identification of sex. Here an attempt has been made to identify the sex of Bodo speakers through vowel
utterance by following Pitch value, LPCC and MFCC techniques. It is found here that the feature vector
organization of LPCC coefficients provides a more promising way of speech-speaker recognition in case of
Bodo Language than that of Pitch and MFCC.
Approach To Build A Marathi Text-To-Speech System Using Concatenative Synthes...IJERA Editor
Marathi is one of the oldest languages in India. This research paper describes the development of Marathi Textto-
Speech System (TTS). In Marathi TTS the input is Marathi text in Unicode. The voices are sampled from real
recorded speech. The objective of a text to speech system is to convert an arbitrary text into its corresponding
spoken waveform. Speech synthesis is a process of building machinery that can generate human-like speech
from any text input to imitate human speakers. Text processing and speech generation are two main components
of a text to speech system. To build a natural sounding speech synthesis system, it is essential that text
processing component produce an appropriate sequence of phonemic units. Generation of sequence of phonetic
units for a given standard word is referred to as letter to phoneme rule or text to phoneme rule. The
complexity of these rules and their derivation depends upon the nature of the language. The quality of a speech
synthesizer is judged by its closeness to the natural human voice and understandability. In this research paper we
described an approach to build a Marathi TTS system using concatenative synthesis method with syllable as a
basic unit of concatenation.
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.
FORMANT ANALYSIS OF BANGLA VOWEL FOR AUTOMATIC SPEECH RECOGNITIONsipij
To provide new technological benefits to the mass people, nowadays, regional and local language
recognition draws attention to the researchers. Similarly to other languages, Bangla speech recognition
scheme is demandable. A formant is considered as the resonance frequency of vocal tract. Formant
frequencies play an important role for the purpose of automatic speech recognition, due to its noise robust
characteristics. In this paper, Bangla vowels are investigated to acquire formant frequencies and its
corresponding bandwidth from continuous Bangla sentences, which are considered as potential parameters
for wide voice applications. For the purpose of formant analysis, cepstrum based formant estimation and
Linear Predictive Coding (LPC) techniques are used. In order to acquire formant characteristics, enrich
continuous sentences and widely available Bangla language corpus namely “SHRUTI” is considered.
Intensive experimentation is carried out to determine formant characteristics (frequency and bandwidth) of
Bangla vowels for both male and female speakers. Finally, vowel recognition accuracy of Bangla language
is reported considering first three formants
Comparison of Feature Extraction MFCC and LPC in Automatic Speech Recognition...TELKOMNIKA JOURNAL
Speech recognition can be defined as the process of converting voice signals into the ranks of the
word, by applying a specific algorithm that is implemented in a computer program. The research of speech
recognition in Indonesia is relatively limited. This paper has studied methods of feature extraction which is
the best among the Linear Predictive Coding (LPC) and Mel Frequency Cepstral Coefficients (MFCC) for
speech recognition in Indonesian language. This is important because the method can produce a high
accuracy for a particular language does not necessarily produce the same accuracy for other languages,
considering every language has different characteristics. Thus this research hopefully can help further
accelerate the use of automatic speech recognition for Indonesian language. There are two main
processes in speech recognition, feature extraction and recognition. The method used for comparison
feature extraction in this study is the LPC and MFCC, while the method of recognition using Hidden
Markov Model (HMM). The test results showed that the MFCC method is better than LPC in Indonesian
language speech recognition.
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.
ANALYSIS OF SPEECH UNDER STRESS USING LINEAR TECHNIQUES AND NON-LINEAR TECHNI...cscpconf
Analysis of speech for recognition of stress is important for identification of emotional state of
person. This can be done using ‘Linear Techniques’, which has different parameters like pitch,
vocal tract spectrum, formant frequencies, Duration, MFCC etc. which are used for extraction
of features from speech. TEO-CB-Auto-Env is the method which is non-linear method of
features extraction. Analysis is done using TU-Berlin (Technical University of Berlin) German
database. Here emotion recognition is done for different emotions like neutral, happy, disgust,
sad, boredom and anger. Emotion recognition is used in lie detector, database access systems, and in military for recognition of soldiers’ emotion identification during the war.
Classification improvement of spoken arabic language based on radial basis fu...IJECEIAES
The important task in the computer interaction is the languages recognition and classification. In the Arab world, there is a persistent need for the Arabic spoken language recognition To help those who have lost the upper parties in doing what they want through speech computer interaction. While, the Arabic automatic speech recognition (AASR) did not receive the desired attention from the researchers. In this paper, the Radial Basis Function(RBF) is used for the improvement of the Arabic spoken language letter. The recognition and classification process are based on three steps; these are; preprocessing, feature extraction and classification (Recognition). The Arabic Language Letters (ALL) recognition is done by using the combination between the statistical features and the Temporal Radial Basis Function for different letter situation and noisy condition. The recognition percent are from 90% - 99.375% has been gained with independent speaker, where these results are over-perform the earlier works by nearly 2.045%. The simulati.on has been made by using Matlab 2015b.
CURVELET BASED SPEECH RECOGNITION SYSTEM IN NOISY ENVIRONMENT: A STATISTICAL ...ijcsit
Speech processing is considered as crucial and an intensive field of research in the growth of robust and efficient speech recognition system. But the accuracy for speech recognition still focuses for variation of context, speaker’s variability, and environment conditions. In this paper, we stated curvelet based Feature Extraction (CFE) method for speech recognition in noisy environment and the input speech signal is decomposed into different frequency channels using the characteristics of curvelet transform for reduce the computational complication and the feature vector size successfully and they have better accuracy, varying window size because of which they are suitable for non –stationary signals. For better word classification and recognition, discrete hidden markov model can be used and as they consider time distribution of
speech signals. The HMM classification method attained the maximum accuracy in term of identification rate for informal with 80.1%, scientific phrases with 86%, and control with 63.8 % detection rates. The objective of this study is to characterize the feature extraction methods and classification phage in speech
recognition system. The various approaches available for developing speech recognition system are compared along with their merits and demerits. The statistical results shows that signal recognition accuracy will be increased by using discrete Curvelet transforms over conventional methods.
An effective evaluation study of objective measures using spectral subtractiv...eSAT Journals
Abstract
Unwanted noises have a negative influence over communication because it disturbs the conversation and make the communication impossible. Speech enhancement algorithms are used for improving the quality and intelligibility or to reduce listener fatigues. Assessment of speech quality can be done by using either subjective listening test or objective quality measure. Evaluation of several objective measures with the speech processed by enhancement algorithms has been performed but these having limitations to assess original speech signal. This paper represents the study of speech quality measures and compute the values used for regression analyses of the objective measures evaluation study using spectral subtraction algorithm based enhanced speech signal.
Keywords: MOS, ITU-T (P.835), SNRseg, log- likelihood ratio and itakura-saito.
IJRET : International Journal of Research in Engineering and Technology is an international peer reviewed, online journal published by eSAT Publishing House for the enhancement of research in various disciplines of Engineering and Technology. The aim and scope of the journal is to provide an academic medium and an important reference for the advancement and dissemination of research results that support high-level learning, teaching and research in the fields of Engineering and Technology. We bring together Scientists, Academician, Field Engineers, Scholars and Students of related fields of Engineering and Technology.
IMPROVING MYANMAR AUTOMATIC SPEECH RECOGNITION WITH OPTIMIZATION OF CONVOLUTI...kevig
Researchers of many nations have developed automatic speech recognition (ASR) to show their national
improvement in information and communication technology for their languages. This work intends to
improve the ASR performance for Myanmar language by changing different Convolutional Neural Network
(CNN) hyperparameters such as number of feature maps and pooling size. CNN has the abilities of
reducing in spectral variations and modeling spectral correlations that exist in the signal due to the locality
and pooling operation. Therefore, the impact of the hyperparameters on CNN accuracy in ASR tasks is
investigated. A 42-hr-data set is used as training data and the ASR performance was evaluated on two open
test sets: web news and recorded data. As Myanmar language is a syllable-timed language, ASR based on
syllable was built and compared with ASR based on word. As the result, it gained 16.7% word error rate
(WER) and 11.5% syllable error rate (SER) on TestSet1. And it also achieved 21.83% WER and 15.76%
SER on TestSet2.
MULTILINGUAL SPEECH IDENTIFICATION USING ARTIFICIAL NEURAL NETWORKijitcs
Speech technology is an emerging technology and automatic speech recognition has made advances in recent years. Many researches has been performed for many foreign and regional languages. But at present the multilingual speech processing technology has been attracting for research purpose. This paper tries to propose a methodology for developing a bilingual speech identification system for Assamese and English language based on artificial neural network.
SYLLABLE-BASED SPEECH RECOGNITION SYSTEM FOR MYANMARijcseit
This proposed system is syllable-based Myanmar speech recognition system. There are three stages: Feature Extraction, Phone Recognition and Decoding. In feature extraction, the system transforms the input speech waveform into a sequence of acoustic feature vectors, each vector representing the information in a small time window of the signal. And then the likelihood of the observation of feature vectors given linguistic units (words, phones, subparts of phones) is computed in the phone recognition stage. Finally, the decoding stage takes the Acoustic Model (AM), which consists of this sequence of acoustic likelihoods, plus an phonetic dictionary of word pronunciations, combined with the Language Model (LM). The system will produce the most likely sequence of words as the output. The system creates the language model for Myanmar by using syllable segmentation and syllable based n-gram method.
SYLLABLE-BASED SPEECH RECOGNITION SYSTEM FOR MYANMARijcseit
This proposed system is syllable-based Myanmar speech recognition system. There are three stages:
Feature Extraction, Phone Recognition and Decoding. In feature extraction, the system transforms the
input speech waveform into a sequence of acoustic feature vectors, each vector representing the
information in a small time window of the signal. And then the likelihood of the observation of feature
vectors given linguistic units (words, phones, subparts of phones) is computed in the phone recognition
stage. Finally, the decoding stage takes the Acoustic Model (AM), which consists of this sequence of
acoustic likelihoods, plus an phonetic dictionary of word pronunciations, combined with the Language
Model (LM). The system will produce the most likely sequence of words as the output. The system creates
the language model for Myanmar by using syllable segmentation and syllable based n-gram method.
SYLLABLE-BASED SPEECH RECOGNITION SYSTEM FOR MYANMARijcseit
This proposed system is syllable-based Myanmar speech recognition system. There are three stages:
Feature Extraction, Phone Recognition and Decoding. In feature extraction, the system transforms the
input speech waveform into a sequence of acoustic feature vectors, each vector representing the
information in a small time window of the signal. And then the likelihood of the observation of feature
vectors given linguistic units (words, phones, subparts of phones) is computed in the phone recognition
stage. Finally, the decoding stage takes the Acoustic Model (AM), which consists of this sequence of
acoustic likelihoods, plus an phonetic dictionary of word pronunciations, combined with the Language
Model (LM). The system will produce the most likely sequence of words as the output. The system creates
the language model for Myanmar by using syllable segmentation and syllable based n-gram method.
SYLLABLE-BASED SPEECH RECOGNITION SYSTEM FOR MYANMARijcseit
This proposed system is syllable-based Myanmar speech recognition system. There are three stages:
Feature Extraction, Phone Recognition and Decoding. In feature extraction, the system transforms the
input speech waveform into a sequence of acoustic feature vectors, each vector representing the
information in a small time window of the signal. And then the likelihood of the observation of feature
vectors given linguistic units (words, phones, subparts of phones) is computed in the phone recognition
stage. Finally, the decoding stage takes the Acoustic Model (AM), which consists of this sequence of
acoustic likelihoods, plus an phonetic dictionary of word pronunciations, combined with the Language
Model (LM). The system will produce the most likely sequence of words as the output. The system creates
the language model for Myanmar by using syllable segmentation and syllable based n-gram method.
Effect of Dynamic Time Warping on Alignment of Phrases and Phonemeskevig
Speech synthesis and recognition are the basic techniques used for man-machine communication. This type
of communication is valuable when our hands and eyes are busy in some other task such as driving a
vehicle, performing surgery, or firing weapons at the enemy. Dynamic time warping (DTW) is mostly used
for aligning two given multidimensional sequences. It finds an optimal match between the given sequences.
The distance between the aligned sequences should be relatively lesser as compared to unaligned
sequences. The improvement in the alignment may be estimated from the corresponding distances. This
technique has applications in speech recognition, speech synthesis, and speaker transformation. The
objective of this research is to investigate the amount of improvement in the alignment corresponding to the
sentence based and phoneme based manually aligned phrases. The speech signals in the form of twenty five
phrases were recorded from each of six speakers (3 males and 3 females). The recorded material was
segmented manually and aligned at sentence and phoneme level. The aligned sentences of different speaker
pairs were analyzed using HNM and the HNM parameters were further aligned at frame level using DTW.
Mahalanobis distances were computed for each pair of sentences. The investigations have shown more than
20 % reduction in the average Mahalanobis distances.
Approach To Build A Marathi Text-To-Speech System Using Concatenative Synthes...IJERA Editor
Marathi is one of the oldest languages in India. This research paper describes the development of Marathi Textto-
Speech System (TTS). In Marathi TTS the input is Marathi text in Unicode. The voices are sampled from real
recorded speech. The objective of a text to speech system is to convert an arbitrary text into its corresponding
spoken waveform. Speech synthesis is a process of building machinery that can generate human-like speech
from any text input to imitate human speakers. Text processing and speech generation are two main components
of a text to speech system. To build a natural sounding speech synthesis system, it is essential that text
processing component produce an appropriate sequence of phonemic units. Generation of sequence of phonetic
units for a given standard word is referred to as letter to phoneme rule or text to phoneme rule. The
complexity of these rules and their derivation depends upon the nature of the language. The quality of a speech
synthesizer is judged by its closeness to the natural human voice and understandability. In this research paper we
described an approach to build a Marathi TTS system using concatenative synthesis method with syllable as a
basic unit of concatenation.
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.
FORMANT ANALYSIS OF BANGLA VOWEL FOR AUTOMATIC SPEECH RECOGNITIONsipij
To provide new technological benefits to the mass people, nowadays, regional and local language
recognition draws attention to the researchers. Similarly to other languages, Bangla speech recognition
scheme is demandable. A formant is considered as the resonance frequency of vocal tract. Formant
frequencies play an important role for the purpose of automatic speech recognition, due to its noise robust
characteristics. In this paper, Bangla vowels are investigated to acquire formant frequencies and its
corresponding bandwidth from continuous Bangla sentences, which are considered as potential parameters
for wide voice applications. For the purpose of formant analysis, cepstrum based formant estimation and
Linear Predictive Coding (LPC) techniques are used. In order to acquire formant characteristics, enrich
continuous sentences and widely available Bangla language corpus namely “SHRUTI” is considered.
Intensive experimentation is carried out to determine formant characteristics (frequency and bandwidth) of
Bangla vowels for both male and female speakers. Finally, vowel recognition accuracy of Bangla language
is reported considering first three formants
Comparison of Feature Extraction MFCC and LPC in Automatic Speech Recognition...TELKOMNIKA JOURNAL
Speech recognition can be defined as the process of converting voice signals into the ranks of the
word, by applying a specific algorithm that is implemented in a computer program. The research of speech
recognition in Indonesia is relatively limited. This paper has studied methods of feature extraction which is
the best among the Linear Predictive Coding (LPC) and Mel Frequency Cepstral Coefficients (MFCC) for
speech recognition in Indonesian language. This is important because the method can produce a high
accuracy for a particular language does not necessarily produce the same accuracy for other languages,
considering every language has different characteristics. Thus this research hopefully can help further
accelerate the use of automatic speech recognition for Indonesian language. There are two main
processes in speech recognition, feature extraction and recognition. The method used for comparison
feature extraction in this study is the LPC and MFCC, while the method of recognition using Hidden
Markov Model (HMM). The test results showed that the MFCC method is better than LPC in Indonesian
language speech recognition.
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.
ANALYSIS OF SPEECH UNDER STRESS USING LINEAR TECHNIQUES AND NON-LINEAR TECHNI...cscpconf
Analysis of speech for recognition of stress is important for identification of emotional state of
person. This can be done using ‘Linear Techniques’, which has different parameters like pitch,
vocal tract spectrum, formant frequencies, Duration, MFCC etc. which are used for extraction
of features from speech. TEO-CB-Auto-Env is the method which is non-linear method of
features extraction. Analysis is done using TU-Berlin (Technical University of Berlin) German
database. Here emotion recognition is done for different emotions like neutral, happy, disgust,
sad, boredom and anger. Emotion recognition is used in lie detector, database access systems, and in military for recognition of soldiers’ emotion identification during the war.
Classification improvement of spoken arabic language based on radial basis fu...IJECEIAES
The important task in the computer interaction is the languages recognition and classification. In the Arab world, there is a persistent need for the Arabic spoken language recognition To help those who have lost the upper parties in doing what they want through speech computer interaction. While, the Arabic automatic speech recognition (AASR) did not receive the desired attention from the researchers. In this paper, the Radial Basis Function(RBF) is used for the improvement of the Arabic spoken language letter. The recognition and classification process are based on three steps; these are; preprocessing, feature extraction and classification (Recognition). The Arabic Language Letters (ALL) recognition is done by using the combination between the statistical features and the Temporal Radial Basis Function for different letter situation and noisy condition. The recognition percent are from 90% - 99.375% has been gained with independent speaker, where these results are over-perform the earlier works by nearly 2.045%. The simulati.on has been made by using Matlab 2015b.
CURVELET BASED SPEECH RECOGNITION SYSTEM IN NOISY ENVIRONMENT: A STATISTICAL ...ijcsit
Speech processing is considered as crucial and an intensive field of research in the growth of robust and efficient speech recognition system. But the accuracy for speech recognition still focuses for variation of context, speaker’s variability, and environment conditions. In this paper, we stated curvelet based Feature Extraction (CFE) method for speech recognition in noisy environment and the input speech signal is decomposed into different frequency channels using the characteristics of curvelet transform for reduce the computational complication and the feature vector size successfully and they have better accuracy, varying window size because of which they are suitable for non –stationary signals. For better word classification and recognition, discrete hidden markov model can be used and as they consider time distribution of
speech signals. The HMM classification method attained the maximum accuracy in term of identification rate for informal with 80.1%, scientific phrases with 86%, and control with 63.8 % detection rates. The objective of this study is to characterize the feature extraction methods and classification phage in speech
recognition system. The various approaches available for developing speech recognition system are compared along with their merits and demerits. The statistical results shows that signal recognition accuracy will be increased by using discrete Curvelet transforms over conventional methods.
An effective evaluation study of objective measures using spectral subtractiv...eSAT Journals
Abstract
Unwanted noises have a negative influence over communication because it disturbs the conversation and make the communication impossible. Speech enhancement algorithms are used for improving the quality and intelligibility or to reduce listener fatigues. Assessment of speech quality can be done by using either subjective listening test or objective quality measure. Evaluation of several objective measures with the speech processed by enhancement algorithms has been performed but these having limitations to assess original speech signal. This paper represents the study of speech quality measures and compute the values used for regression analyses of the objective measures evaluation study using spectral subtraction algorithm based enhanced speech signal.
Keywords: MOS, ITU-T (P.835), SNRseg, log- likelihood ratio and itakura-saito.
IJRET : International Journal of Research in Engineering and Technology is an international peer reviewed, online journal published by eSAT Publishing House for the enhancement of research in various disciplines of Engineering and Technology. The aim and scope of the journal is to provide an academic medium and an important reference for the advancement and dissemination of research results that support high-level learning, teaching and research in the fields of Engineering and Technology. We bring together Scientists, Academician, Field Engineers, Scholars and Students of related fields of Engineering and Technology.
IMPROVING MYANMAR AUTOMATIC SPEECH RECOGNITION WITH OPTIMIZATION OF CONVOLUTI...kevig
Researchers of many nations have developed automatic speech recognition (ASR) to show their national
improvement in information and communication technology for their languages. This work intends to
improve the ASR performance for Myanmar language by changing different Convolutional Neural Network
(CNN) hyperparameters such as number of feature maps and pooling size. CNN has the abilities of
reducing in spectral variations and modeling spectral correlations that exist in the signal due to the locality
and pooling operation. Therefore, the impact of the hyperparameters on CNN accuracy in ASR tasks is
investigated. A 42-hr-data set is used as training data and the ASR performance was evaluated on two open
test sets: web news and recorded data. As Myanmar language is a syllable-timed language, ASR based on
syllable was built and compared with ASR based on word. As the result, it gained 16.7% word error rate
(WER) and 11.5% syllable error rate (SER) on TestSet1. And it also achieved 21.83% WER and 15.76%
SER on TestSet2.
MULTILINGUAL SPEECH IDENTIFICATION USING ARTIFICIAL NEURAL NETWORKijitcs
Speech technology is an emerging technology and automatic speech recognition has made advances in recent years. Many researches has been performed for many foreign and regional languages. But at present the multilingual speech processing technology has been attracting for research purpose. This paper tries to propose a methodology for developing a bilingual speech identification system for Assamese and English language based on artificial neural network.
SYLLABLE-BASED SPEECH RECOGNITION SYSTEM FOR MYANMARijcseit
This proposed system is syllable-based Myanmar speech recognition system. There are three stages: Feature Extraction, Phone Recognition and Decoding. In feature extraction, the system transforms the input speech waveform into a sequence of acoustic feature vectors, each vector representing the information in a small time window of the signal. And then the likelihood of the observation of feature vectors given linguistic units (words, phones, subparts of phones) is computed in the phone recognition stage. Finally, the decoding stage takes the Acoustic Model (AM), which consists of this sequence of acoustic likelihoods, plus an phonetic dictionary of word pronunciations, combined with the Language Model (LM). The system will produce the most likely sequence of words as the output. The system creates the language model for Myanmar by using syllable segmentation and syllable based n-gram method.
SYLLABLE-BASED SPEECH RECOGNITION SYSTEM FOR MYANMARijcseit
This proposed system is syllable-based Myanmar speech recognition system. There are three stages:
Feature Extraction, Phone Recognition and Decoding. In feature extraction, the system transforms the
input speech waveform into a sequence of acoustic feature vectors, each vector representing the
information in a small time window of the signal. And then the likelihood of the observation of feature
vectors given linguistic units (words, phones, subparts of phones) is computed in the phone recognition
stage. Finally, the decoding stage takes the Acoustic Model (AM), which consists of this sequence of
acoustic likelihoods, plus an phonetic dictionary of word pronunciations, combined with the Language
Model (LM). The system will produce the most likely sequence of words as the output. The system creates
the language model for Myanmar by using syllable segmentation and syllable based n-gram method.
SYLLABLE-BASED SPEECH RECOGNITION SYSTEM FOR MYANMARijcseit
This proposed system is syllable-based Myanmar speech recognition system. There are three stages:
Feature Extraction, Phone Recognition and Decoding. In feature extraction, the system transforms the
input speech waveform into a sequence of acoustic feature vectors, each vector representing the
information in a small time window of the signal. And then the likelihood of the observation of feature
vectors given linguistic units (words, phones, subparts of phones) is computed in the phone recognition
stage. Finally, the decoding stage takes the Acoustic Model (AM), which consists of this sequence of
acoustic likelihoods, plus an phonetic dictionary of word pronunciations, combined with the Language
Model (LM). The system will produce the most likely sequence of words as the output. The system creates
the language model for Myanmar by using syllable segmentation and syllable based n-gram method.
SYLLABLE-BASED SPEECH RECOGNITION SYSTEM FOR MYANMARijcseit
This proposed system is syllable-based Myanmar speech recognition system. There are three stages:
Feature Extraction, Phone Recognition and Decoding. In feature extraction, the system transforms the
input speech waveform into a sequence of acoustic feature vectors, each vector representing the
information in a small time window of the signal. And then the likelihood of the observation of feature
vectors given linguistic units (words, phones, subparts of phones) is computed in the phone recognition
stage. Finally, the decoding stage takes the Acoustic Model (AM), which consists of this sequence of
acoustic likelihoods, plus an phonetic dictionary of word pronunciations, combined with the Language
Model (LM). The system will produce the most likely sequence of words as the output. The system creates
the language model for Myanmar by using syllable segmentation and syllable based n-gram method.
Effect of Dynamic Time Warping on Alignment of Phrases and Phonemeskevig
Speech synthesis and recognition are the basic techniques used for man-machine communication. This type
of communication is valuable when our hands and eyes are busy in some other task such as driving a
vehicle, performing surgery, or firing weapons at the enemy. Dynamic time warping (DTW) is mostly used
for aligning two given multidimensional sequences. It finds an optimal match between the given sequences.
The distance between the aligned sequences should be relatively lesser as compared to unaligned
sequences. The improvement in the alignment may be estimated from the corresponding distances. This
technique has applications in speech recognition, speech synthesis, and speaker transformation. The
objective of this research is to investigate the amount of improvement in the alignment corresponding to the
sentence based and phoneme based manually aligned phrases. The speech signals in the form of twenty five
phrases were recorded from each of six speakers (3 males and 3 females). The recorded material was
segmented manually and aligned at sentence and phoneme level. The aligned sentences of different speaker
pairs were analyzed using HNM and the HNM parameters were further aligned at frame level using DTW.
Mahalanobis distances were computed for each pair of sentences. The investigations have shown more than
20 % reduction in the average Mahalanobis distances.
EFFECT OF DYNAMIC TIME WARPING ON ALIGNMENT OF PHRASES AND PHONEMESkevig
Speech synthesis and recognition are the basic techniques used for man-machine communication. This type
of communication is valuable when our hands and eyes are busy in some other task such as driving a
vehicle, performing surgery, or firing weapons at the enemy. Dynamic time warping (DTW) is mostly used
for aligning two given multidimensional sequences. It finds an optimal match between the given sequences.
The distance between the aligned sequences should be relatively lesser as compared to unaligned
sequences. The improvement in the alignment may be estimated from the corresponding distances. This
technique has applications in speech recognition, speech synthesis, and speaker transformation. The
objective of this research is to investigate the amount of improvement in the alignment corresponding to the
sentence based and phoneme based manually aligned phrases. The speech signals in the form of twenty five
phrases were recorded from each of six speakers (3 males and 3 females). The recorded material was
segmented manually and aligned at sentence and phoneme level. The aligned sentences of different speaker
pairs were analyzed using HNM and the HNM parameters were further aligned at frame level using DTW.
Mahalanobis distances were computed for each pair of sentences. The investigations have shown more than
20 % reduction in the average Mahalanobis distances.
Emotional telugu speech signals classification based on k nn classifiereSAT Publishing House
IJRET : International Journal of Research in Engineering and Technology is an international peer reviewed, online journal published by eSAT Publishing House for the enhancement of research in various disciplines of Engineering and Technology. The aim and scope of the journal is to provide an academic medium and an important reference for the advancement and dissemination of research results that support high-level learning, teaching and research in the fields of Engineering and Technology. We bring together Scientists, Academician, Field Engineers, Scholars and Students of related fields of Engineering and Technology.
Emotional telugu speech signals classification based on k nn classifiereSAT Journals
Abstract Speech processing is the study of speech signals, and the methods used to process them. In application such as speech coding, speech synthesis, speech recognition and speaker recognition technology, speech processing is employed. In speech classification, the computation of prosody effects from speech signals plays a major role. In emotional speech signals pitch and frequency is a most important parameters. Normally, the pitch value of sad and happy speech signals has a great difference and the frequency value of happy is higher than sad speech. But, in some cases the frequency of happy speech is nearly similar to sad speech or frequency of sad speech is similar to happy speech. In such situation, it is difficult to recognize the exact speech signal. To reduce such drawbacks, in this paper we propose a Telugu speech emotion classification system with three features like Energy Entropy, Short Time Energy, Zero Crossing Rate and K-NN classifier for the classification. Features are extracted from the speech signals and given to the K-NN. The implementation result shows the effectiveness of proposed speech emotion classification system in classifying the Telugu speech signals based on their prosody effects. The performance of the proposed speech emotion classification system is evaluated by conducting cross validation on the Telugu speech database. Keywords: Emotion Classification, K-NN classifier, Energy Entropy, Short Time Energy, Zero Crossing Rate.
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.
Hindi digits recognition system on speech data collected in different natural...csandit
This paper presents a baseline digits speech recognizer for Hindi language. The recording environment is different for all speakers, since the data is collected in their respective homes. The different environment refers to vehicle horn noises in some road facing rooms, internal background noises in some rooms like opening doors, silence in some rooms etc. All these recordings are used for training acoustic model. The Acoustic Model is trained on 8 speakers’ audio data. The vocabulary size of the recognizer is 10 words. HTK toolkit is used for building
acoustic model and evaluating the recognition rate of the recognizer. The efficiency of the recognizer developed on recorded data, is shown at the end of the paper and possible directions for future research work are suggested.
Effect of MFCC Based Features for Speech Signal Alignmentskevig
The fundamental techniques used for man-machine communication include Speech synthesis, speech
recognition, and speech transformation. Feature extraction techniques provide a compressed
representation of the speech signals. The HNM analyses and synthesis provides high quality speech with
less number of parameters. Dynamic time warping is well known technique used for aligning two given
multidimensional sequences. It locates an optimal match between the given sequences. The improvement in
the alignment is estimated from the corresponding distances. The objective of this research is to investigate
the effect of dynamic time warping on phrases, words, and phonemes based alignments. The speech signals
in the form of twenty five phrases were recorded. The recorded material was segmented manually and
aligned at sentence, word, and phoneme level. The Mahalanobis distance (MD) was computed between the
aligned frames. The investigation has shown better alignment in case of HNM parametric domain. It has
been seen that effective speech alignment can be carried out even at phrase level
EFFECT OF MFCC BASED FEATURES FOR SPEECH SIGNAL ALIGNMENTSijnlc
The fundamental techniques used for man-machine communication include Speech synthesis, speech
recognition, and speech transformation. Feature extraction techniques provide a compressed
representation of the speech signals. The HNM analyses and synthesis provides high quality speech with
less number of parameters. Dynamic time warping is well known technique used for aligning two given
multidimensional sequences. It locates an optimal match between the given sequences. The improvement in
the alignment is estimated from the corresponding distances. The objective of this research is to investigate
the effect of dynamic time warping on phrases, words, and phonemes based alignments. The speech signals
in the form of twenty five phrases were recorded. The recorded material was segmented manually and
aligned at sentence, word, and phoneme level. The Mahalanobis distance (MD) was computed between the
aligned frames. The investigation has shown better alignment in case of HNM parametric domain. It has
been seen that effective speech alignment can be carried out even at phrase level.
Broad Phoneme Classification Using Signal Based Features ijsc
Speech is the most efficient and popular means of human communication Speech is produced as a sequence of phonemes. Phoneme recognition is the first step performed by automatic speech recognition system. The state-of-the-art recognizers use mel-frequency cepstral coefficients (MFCC) features derived through short time analysis, for which the recognition accuracy is limited. Instead of this, here broad phoneme classification is achieved using features derived directly from the speech at the signal level itself. Broad phoneme classes include vowels, nasals, fricatives, stops, approximants and silence. The features identified useful for broad phoneme classification are voiced/unvoiced decision, zero crossing rate (ZCR), short time energy, most dominant frequency, energy in most dominant frequency, spectral flatness measure and first three formants. Features derived from short time frames of training speech are used to train a multilayer feedforward neural network based classifier with manually marked class label as output and classification accuracy is then tested. Later this broad phoneme classifier is used for broad syllable structure prediction which is useful for applications such as automatic speech recognition and automatic language identification.
Advanced Computing: An International Journal (ACIJ) is a peer-reviewed, open access peer-reviewed journal that publishes articles which contribute new results in all areas of the advanced computing. The journal focuses on all technical and practical aspects of high performance computing, green computing, pervasive computing, cloud computing etc. The goal of this journal is to bring together researchers and a practitioners from academia and industry to focus on understanding advances in computing and establishing new collaborations in these areas.
Authors are solicited to contribute to the journal by submitting articles that illustrate research results, projects, surveying works and industrial experiences that describe significant advances in the areas of computing.
Call for Papers - Advanced Computing An International Journal (ACIJ) (2).pdfacijjournal
Submit your Research Papers!!!
Advanced Computing: An International Journal ( ACIJ )
ISSN: 2229 -6727 [Online] ; 2229 - 726X [Print]
Webpage URL: http://airccse.org/journal/acij/acij.html
Submission URL: http://coneco2009.com/submissions/imagination/home.html
Submission Deadline : April 08, 2023
Here's where you can reach us : acijjournal@yahoo.com or acij@aircconline
Advanced Computing: An International Journal (ACIJ
)
is a bi monthly open access peer-reviewed journal that publishes articles which contribute new results in all areas of the advancedcomputing. The journal focuses on all technical and practical aspects of high performancecomputing, green computing, pervasive computing, cloud computing etc. The goal of this journalis to bring together researchers anda practitioners from academia and industry to focus onunderstanding advances in computing and establishing new collaborations in these areas
Submit your Research Papers!!!
Advanced Computing: An International Journal ( ACIJ )
ISSN: 2229 -6727 [Online] ; 2229 - 726X [Print]
Webpage URL: http://airccse.org/journal/acij/acij.html
Submission URL: http://coneco2009.com/submissions/imagination/home.html
Here's where you can reach us : acijjournal@yahoo.com or acij@aircconline.com
7thInternational Conference on Data Mining & Knowledge Management (DaKM 2022)acijjournal
7thInternational Conference on Data Mining & Knowledge Management (DaKM 2022)provides a forum for researchers who address this issue and to present their work in a peer-reviewed forum.
7thInternational Conference on Data Mining & Knowledge Management (DaKM 2022)acijjournal
7thInternational Conference on Data Mining & Knowledge Management (DaKM 2022)provides a forum for researchers who address this issue and to present their work in a peer-reviewed forum.
7thInternational Conference on Data Mining & Knowledge Management (DaKM 2022)acijjournal
7thInternational Conference on Data Mining & Knowledge Management (DaKM 2022)provides a forum for researchers who address this issue and to present their work in a peer-reviewed forum.
4thInternational Conference on Machine Learning & Applications (CMLA 2022)acijjournal
4thInternational Conference on Machine Learning & Applications (CMLA 2022)will provide an excellent international forum for sharing knowledge and results in theory, methodology and applications of on Machine Learning & Applications. The aim of the conference is to provide a platform to the researchers and practitioners from both academia as well as industry to meet and share cutting-edge development in the field.
7thInternational Conference on Data Mining & Knowledge Management (DaKM 2022)acijjournal
7thInternational Conference on Data Mining & Knowledge Management (DaKM 2022)provides a forum for researchers who address this issue and to present their work in a peer-reviewed forum.Authors are solicited to contribute to the conference by submitting articles that illustrate research results, projects, surveying works and industrial experiences that describe significant advances in the following areas, but are not limited to these topics only.
3rdInternational Conference on Natural Language Processingand Applications (N...acijjournal
3rdInternational Conference on Natural Language Processing and Applications (NLPA 2022)will provide an excellent international forum for sharing knowledge and results in theory, methodology and applications of Natural Language Computing and its applications. The Conference looks for significant contributions to all major fieldsof the Natural Language processing in theoretical and practical aspects.
4thInternational Conference on Machine Learning & Applications (CMLA 2022)acijjournal
4thInternational Conference on Machine Learning & Applications (CMLA 2022)will provide an excellent international forum for sharing knowledge and results in theory, methodology and applications of on Machine Learning & Applications. The aim of the conference is to provide a platform to the researchers and practitioners from both academia as well as industry to meet and share cutting-edge development in the field.
Graduate School Cyber Portfolio: The Innovative Menu For Sustainable Developmentacijjournal
In today’s milieu, new demands and trends emerge in the field of Education giving teachers of Higher Education Institutions (HEI’s) no choice but to be innovative to cope with the fast changing technology. To be naturally innovative, a graduate school teacher needs to be technologically and pedagogically competent. One of the ways to be on this level is by creating his cyber portfolio to support students’ eportfolio for lifelong learning. Cyber portfolio is an innovative menu for teachers who seek out strategies to integrate technology in their lessons. This paper presents a straightforward preparation on how to innovate a cyber portfolio that has its practical and breakthrough solution against expensive and inflexible vended software which often saddle many universities. Additionally, this cyber portfolio is free and it addresses the 21st century skills of graduate students blended with higher order thinking skills, multiple intelligence, technology and multimedia.
Genetic Algorithms and Programming - An Evolutionary Methodologyacijjournal
Genetic programming (GP) is an automated method for creating a working computer program from a high-level problem statement of a problem. Genetic programming starts from a high-level statement of “what needs to be done” and automatically creates a computer program to solve the problem. In artificial intelligence, genetic programming (GP) is an evolutionary algorithm-based methodology inspired by biological evolution to find computer programs that perform a user defined task. It is a specialization of genetic algorithms (GA) where each individual is a computer program. It is a machine learning technique used to optimize a population of computer programs according to a fitness span determined by a program's ability to perform a given computational task. This paper presents a idea of the various principles of genetic programming which includes, relative effectiveness of mutation, crossover, breeding computer programs and fitness test in genetic programming. The literature of traditional genetic algorithms contains related studies, but through GP, it saves time by freeing the human from having to design complex algorithms. Not only designing the algorithms but creating ones that give optimal solutions than traditional counterparts in noteworthy ways.
Data Transformation Technique for Protecting Private Information in Privacy P...acijjournal
Data mining is the process of extracting patterns from data. Data mining is seen as an increasingly important tool by modern business to transform data into an informational advantage. Data
Mining can be utilized in any organization that needs to find patterns or relationships in their data. A group of techniques that find relationships that have not previously been discovered. In many situations, the extracted patterns are highly private and it should not be disclosed. In order to maintain the secrecy of data,
there is in need of several techniques and algorithms for modifying the original data in order to limit the extraction of confidential patterns. There have been two types of privacy in data mining. The first type of privacy is that the data is altered so that the mining result will preserve certain privacy. The second type of privacy is that the data is manipulated so that the mining result is not affected or minimally affected. The aim of privacy preserving data mining researchers is to develop data mining techniques that could be
applied on data bases without violating the privacy of individuals. Many techniques for privacy preserving data mining have come up over the last decade. Some of them are statistical, cryptographic, randomization methods, k-anonymity model, l-diversity and etc. In this work, we propose a new perturbative masking technique known as data transformation technique can be used for protecting the sensitive information. An
experimental result shows that the proposed technique gives the better result compared with the existing technique.
Advanced Computing: An International Journal (ACIJ) acijjournal
Advanced Computing: An International Journal (ACIJ) is a bi monthly open access peer-reviewed journal that publishes articles which contribute new results in all areas of the advanced computing. The journal focuses on all technical and practical aspects of high performance computing, green computing, pervasive computing, cloud computing etc. The goal of this journal is to bring together researchers and practitioners from academia and industry to focus on understanding advances in computing and establishing new collaborations in these areas.
E-Maintenance: Impact Over Industrial Processes, Its Dimensions & Principlesacijjournal
During the course of the industrial 4.0 era, companies have been exponentially developed and have
digitized almost the whole business system to stick to their performance targets and to keep or to even
enlarge their market share. Maintenance function has obviously followed the trend as it’s considered one
of the most important processes in every enterprise as it impacts a group of the most critical performance
indicators such as: cost, reliability, availability, safety and productivity. E-maintenance emerged in early
2000 and now is a common term in maintenance literature representing the digitalized side of maintenance
whereby assets are monitored and controlled over the internet. According to literature, e-maintenance has
a remarkable impact on maintenance KPIs and aims at ambitious objectives like zero-downtime.
10th International Conference on Software Engineering and Applications (SEAPP...acijjournal
10th International Conference on Software Engineering and Applications (SEAPP 2021) will provide an excellent international forum for sharing knowledge and results in theory, methodology and applications of Software Engineering and Applications. The goal of this Conference is to bring together researchers and practitioners from academia and industry to focus on understanding Modern software engineering concepts and establishing new collaborations in these areas.
10th International conference on Parallel, Distributed Computing and Applicat...acijjournal
10th International conference on Parallel, Distributed Computing and Applications (IPDCA 2021) will provide an excellent international forum for sharing knowledge and results in theory, methodology and applications of Parallel, Distributed Computing. Original papers are invited on Algorithms and Applications, computer Networks, Cyber trust and security, Wireless networks and mobile Computing and Bioinformatics. The aim of the conference is to provide a platform to the researchers and practitioners from both academia as well as industry to meet and share cutting-edge development in the field.
DETECTION OF FORGERY AND FABRICATION IN PASSPORTS AND VISAS USING CRYPTOGRAPH...acijjournal
In this paper, we present a novel solution to detect forgery and fabrication in passports and visas using
cryptography and QR codes. The solution requires that the passport and visa issuing authorities obtain a
cryptographic key pair and publish their public key on their website. Further they are required to encrypt
the passport or visa information with their private key, encode the ciphertext in a QR code and print it on
the passport or visa they issue to the applicant.
The issuing authorities are also required to create a mobile or desktop QR code scanning app and place it
for download on their website or Google Play Store and iPhone App Store. Any individual or immigration
authority that needs to check the passport or visa for forgery and fabrication can scan its QR code, which
will decrypt the ciphertext encoded in the QR code using the public key stored in the app memory and
displays the passport or visa information on the app screen. The details on the app screen can be
compared with the actual details printed on the passport or visa. Any mismatch between the two is a clear
indication of forgery or fabrication.
Discussed the need for a universal desktop and mobile app that can be used by immigration authorities and
consulates all over the world to enable fast checking of passports and visas at ports of entry for forgery
and fabrication.
Detection of Forgery and Fabrication in Passports and Visas Using Cryptograph...acijjournal
In this paper, wepresenta novel solution to detect forgery and fabrication in passports and visas using cryptography and QR codes. The solution requires that the passport and visa issuing authorities obtain a cryptographic key pair and publish their public key on their website. Further they are required to encrypt the passport or visa information with their private key, encode the ciphertext in a QR code and print it on the passport or visa they issue to the applicant.
The issuing authorities are also required to create a mobile or desktop QR code scanning app and place it for download on their website or Google Play Store and iPhone App Store. Any individual or immigration authority that needs to check the passport or visa for forgery and fabrication can scan its QR code, which will decrypt the ciphertext encoded in the QR code using the public key stored in the app memory and displays the passport or visa information on the app screen. The details on the app screen can be compared with the actual details printed on the passport or visa. Any mismatch between the two is a clear indication of forgery or fabrication.
Discussed the need for a universal desktop and mobile app that can be used by immigration authorities and consulates all over the world to enable fast checking of passports and visas at ports of entry for forgery and fabrication.
CHINA’S GEO-ECONOMIC OUTREACH IN CENTRAL ASIAN COUNTRIES AND FUTURE PROSPECTjpsjournal1
The rivalry between prominent international actors for dominance over Central Asia's hydrocarbon
reserves and the ancient silk trade route, along with China's diplomatic endeavours in the area, has been
referred to as the "New Great Game." This research centres on the power struggle, considering
geopolitical, geostrategic, and geoeconomic variables. Topics including trade, political hegemony, oil
politics, and conventional and nontraditional security are all explored and explained by the researcher.
Using Mackinder's Heartland, Spykman Rimland, and Hegemonic Stability theories, examines China's role
in Central Asia. This study adheres to the empirical epistemological method and has taken care of
objectivity. This study analyze primary and secondary research documents critically to elaborate role of
china’s geo economic outreach in central Asian countries and its future prospect. China is thriving in trade,
pipeline politics, and winning states, according to this study, thanks to important instruments like the
Shanghai Cooperation Organisation and the Belt and Road Economic Initiative. According to this study,
China is seeing significant success in commerce, pipeline politics, and gaining influence on other
governments. This success may be attributed to the effective utilisation of key tools such as the Shanghai
Cooperation Organisation and the Belt and Road Economic Initiative.
Harnessing WebAssembly for Real-time Stateless Streaming PipelinesChristina Lin
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SPEAKER VERIFICATION USING ACOUSTIC AND PROSODIC FEATURES
1. Advanced Computing: An International Journal ( ACIJ ), Vol.4, No.1, January 2013
DOI : 10.5121/acij.2013.4105 45
SPEAKER VERIFICATION USING ACOUSTIC AND
PROSODIC FEATURES
Utpal Bhattacharjee1
and Kshirod Sarmah2
Department of Computer Science and Engineering, Rajiv Gandhi University, Rono
Hills, Doimukh, Arunachal Pradesh, India, PIN – 791112
1
utpalbhattacharjee@rediffmail.com
2
kshirodsarmah@gmail.com
ABSTRACT
In this paper we report the experiment carried out on recently collected speaker recognition database
namely Arunachali Language Speech Database (ALS-DB)to make a comparative study on the
performance of acoustic and prosodic features for speaker verification task.The speech database consists
of speech data recorded from 200 speakers with Arunachali languages of North-East India as mother
tongue. The collected database is evaluated using Gaussian mixture model-Universal Background Model
(GMM-UBM) based speaker verification system. The acoustic feature considered in the present study is
Mel-Frequency Cepstral Coefficients (MFCC) along with its derivatives.The performance of the system
has been evaluated for both acoustic feature and prosodic feature individually as well as in
combination.It has been observed that acoustic feature, when considered individually, provide better
performance compared to prosodic features. However, if prosodic features are combined with acoustic
feature, performance of the system outperforms both the systems where the features are considered
individually. There is a nearly 5% improvement in recognition accuracy with respect to the system where
acoustic features are considered individually and nearly 20% improvement with respect to the system
where only prosodic features are considered.
KEYWORDS
Speaker Verification, Multilingual,GMM-UBM,MFCC, Prosodic Feature
1. INTRODUCTION
Automatic Speaker Recognition (ASR)refers to recognizing persons from their voice. The sound
of each speaker is not identical because their vocal tract shapes, larynx sizes and other parts of
their voice production organs are different. ASR System can be divided into either Automatic
Speaker Verification(ASV) or Automatic Speaker Identification (ASI) systems [1,2, 3].Speaker
verification aims to verify whether an input speech corresponds to the claimed identity. Speaker
identification aims to identify an input speech by selecting one model from a set of enrolled
speaker models: in some cases, speaker verification will follow speaker identification in order to
validate the identification result [4].Speaker Verification is the task of determining whether a
person is who he or she claims to be (a yes/ no decision).Since it is generally assumed that
imposter (falsely claimed speaker) are not known to the system, it is also referred to as an Open-
Set task [5].
The speaker verification system aims to verify whether an input speech corresponds to the
claimed identity or not. A security system based on this ability has great potential in several
application domains. Speaker verification systems are typically distinguished into two
categories – text-dependent and text-independent [6]. In text-dependent system, a predetermined
group of words or sentences is used to enroll the speaker to the system and those words or
sentences are used to verify the speaker. Text-dependent system use an explicit verification
protocol, usually combined with pass phrases or Personal Identification Number (PIN) as an
2. Advanced Computing: An International Journal ( ACIJ ), Vol.4, No.1, January 2013
46
additional level of security. In text-independent system, no constraints are placed on what can
be said by the speaker. It is an implicit verification process where the verification is done while
the user is performing some other tasks like talking with the customer care executive or
registering acomplain.
The state-of-art speaker verification system use either adaptive Gaussian mixture model (GMM)
[7] with universal background model (UBM) or support vector machine (SVM) over GMM
super-vector [8].
Mel-frequency Cepstral coefficients (MFCCs) are most commonly used feature vector for
speaker verification system. Supra-segmental features like – prosody, speaking style are also
combined with the cepstral feature to improve the performance[9].
Prosody plays a key role in the perception of human speech. The information contained in
prosodic features is partly different from the information contained in cepstral features.
Therefore, more and more researchers from the speech recognition area are showing interests in
prosodic features. Generally, prosody means “the structure that organizessound”. Pitch (tone),
Energy (loudness) and normalized duration (rhythm) are the main components of prosody for a
speaker. Prosody can vary from speaker to speaker and relies on long-term information of
speech.
Very often, prosodic features are extracted with larger frame size than acoustical features since
prosodic features exist over a long speech segment such as syllables. The pitch and energy-
contours change slowly compared to the spectrum, which implies that the variation can be
captured over a long speech segment [10].
The rest of the paper is organized as follows: Section–2 describes the details of the speaker
recognition database. Section–3 details the speaker verification system. The experimental setup,
data used in the experiments and result obtained are described in Section 4. The paper is
concluded in Section–5.
2. SPEAKER RECOGNITION DATABASE
In this section we describe the recently collected Arunachali Language Speech Database (ALS-
DB).Arunachal Pradesh of North East India is one of the linguistically richest and most diverse
regions in all of Asia, being home to at least thirty and possibly as many as fifty distinct
languages in addition to innumerable dialects and subdialects thereof [11]. The vast majority of
languages indigenous to modern-day Arunachal Pradesh belong to the Tibeto-Burman language
family. The majority of these in turn belong to a single branch of Tibeto-Burman, namely Tani.
Almost all Tani languages are indigenous to central Arunachal Pradesh while a handful of Tani
languages are also spoken in Tibet. Tani languages are noticeably characterized by an overall
relative uniformity, suggesting relatively recent origin and dispersal within their present-day
area of concentration. Most Tani languages are mutually intelligible with at least one other Tani
language, meaning that the area constitutes a dialect chain. In addition to these non-Indo-
European languages, the Indo-European languages Assamese, Bengali, English, Nepali and
especially Hindi are making strong inroads into Arunachal Pradesh primarily as a result of the
primary education system in which classes are generally taught by immigrant teachers from
Hindi-speaking parts of northern India. Because of the linguistic diversity of the region, English
is the only official language recognized in the state.
To study the impact of language variability on speaker recognition task, ALS-DB is collected in
multilingual environment. Each speaker is recorded for three different languages – English,
Hindi and a local language, which belongs to any one of the four major Arunachali languages -
3. Advanced Computing: An International Journal ( ACIJ ), Vol.4, No.1, January 2013
47
Adi, Nyishi, Galo and Apatani. Each recording is of 4-5 minutes duration. Speech data were
recorded in parallel across four recording devices, which are listed in table -1.
Table 1: Device type and recording specifications
Device
Sl. No
Device Type Sampling
Rate
File Format
Device 1 Table mounted microphone 16 kHz wav
Device 2 Headset microphone 16 kHz wav
Device 3 Laptop microphone 16 kHz wav
Device 4 Portable Voice Recorder 44.1 kHz mp3
The speakers are recorded for reading style of conversation. The speech data collection was
done in laboratory environment with air conditioner, server and other equipments switched on.
The speech data was contributed by 112 male and 88 female informants chosen from the age
group 20-50 years. During recording, the subject was asked to read a story from the school book
of duration 4-5 minutes in each language for twice and the second reading was considered for
recording. Each informant participates in four recording sessions and there is a gap of at least
one week between two sessions.
3. SPEAKER VERIFICATION SYSTEM
In this works, aSpeaker Verification system has been developed using Gaussian Mixture Model
with Universal Background model (GMM-UBM) based modeling approach. A 39-dimensional
feature vector was used, made up of 13 mel-frequency cepstral coefficient (MFCC) and their
first order and second order derivatives. The first order derivatives were approximated over
three samples and similarly for second order derivatives.The coefficients were extracted from a
speech sampled at 16 KHz with 16 bits/sample resolution. A pre-emphasis filterܪሺݖሻ = 1 −
0.97ݖିଵ
has been applied before framing. The pre-emphasized speech signal is segmented into
frame of 20 microseconds with frame frequency 100 Hz. Each frame is multiplied by a
Hamming window. From the windowed frame, FFT has been computed and the magnitude
spectrum is filtered with a bank of 22 triangular filters spaced on Mel-scale. The log-
compressed filter outputs are converted to cepstral coefficients by DCT. The 0th
cepstral
coefficient is not used in the cepstral feature vector since it corresponds to the energy of the
whole frame [12], and only next 12 MFCC coefficients have been used. To capture the time
varying nature of the speech signal, the MFCC coefficients were combined with its first order
and second derivatives, we get a 39-dimensional feature vector.
Cepstral Mean Subtraction (CMS) has been applied on all features to reduce the effect of
channel mismatch. In this approach we apply Cepstral Variance Normalization (CVN) which
forces the feature vectors to follow a unit variance distribution in feature level solution to get
more robustness results.
The prosodic features typically include pitch, intensity and normalized duration of the syllable.
However, as a limitation of combination, the features have to be frame based and therefore only
pitch and intensity are selected to represent the prosodic information in the present study. Pitch
and intensity are static features as they are calculated frame by frame. They only represent the
exact value of the current frame. In order to incorporate more temporal information their 1st
order and 2nd
order derivatives has also been included.
The Gaussian mixture model with 1024 Gaussian components has been used for both the UBM
and speaker model. The UBM was created by training the speaker model with 50 male and 50
4. Advanced Computing: An International Journal ( ACIJ ), Vol.4, No.1, January 2013
48
female speaker’s data with 512 Gaussian components each male and female model with
Expectation Maximization (EM) algorithm. Finally UBM model is created by pulling the both
male and female models and finding the average of all these models [13]. The speaker models
were created by adapting only the mean parameters of the UBM using maximum a posteriori
(MAP) approach with the speaker specific data [8].
The detection error trade-off (DET) curve has been plotted using log likelihood ratio between
the claimed model and the UBM and the equal error rate (EER) obtained from the DET curve
has been used as a measure for the performance of the speaker verification system. Another
measurement Minimum DCF values has also been evaluated.
4. EXPERIMENTS AND RESULTS
All the experiments reported in this paper are carried out using the database ASL-DB described
in section 2. An energy based silence detector is used to identify and discard the silence frames
prior to feature extraction. Only data from the headset microphone has been considered in the
present study. All the four available sessions were considered for the experiments. Each speaker
model was trained using one complete session. The test sequences were extracted from the next
three sessions. The training set consists of speech data of length 120 seconds per speaker. The
test set consists of speech data of length 15 seconds, 30 seconds and 45 seconds. The test set
contains more than 3500 test segments of varying length and each test segment will be evaluated
against 11 hypothesized speakers of the same sex as segment speaker [14].
In this experiment single language (English, Hindi, and a Local language) has been considered
for training the system and each language has been considered separately for testing the system.
Training sample of length 120 seconds from a single session has been considered for training
the system and the other three sessions have been considered for testing the system.
Figure–1(a),(b) and (c) shows the DET curves for Speaker Verification system obtained for the
three languages in the speech database. The result of the experiments has been summarized in
table–1.
Figure 1(a). DET curves for the speaker verification system using the Local language as training
and testing language
5. Advanced Computing: An International Journal ( ACIJ ), Vol.4, No.1, January 2013
49
Figure 1(b). DET curves for the speaker verification system using the Hindi language as training
and testing language
Figure 1(c). DET curves for the speaker verification system using the English language as
training and testing language
6. Advanced Computing: An International Journal ( ACIJ ), Vol.4, No.1, January 2013
50
Table 2.EER andMin DCF values for speaker verification system for training with one language
and testing with each language
Training &
Testing
Language
Feature Vectors ERR% Recognition
Rate%
Minimum
DCF Value
Local MFCC + ∆MFCC+ ∆∆MFCC 6.81 93.19 0.0991
PROSODIC 21.50 79.50 0.4045
MFCC + ∆MFCC+ ∆∆MFCC+
PROSODIC
4.55 95.45 0.0823
Hindi MFCC + ∆MFCC+ ∆∆MFCC 6.81 93.19 0.0968
PROSODIC 25.00 75.00 0.4438
MFCC + ∆MFCC+ ∆∆MFCC+
PROSODIC
4.55 95.45 0.0927
English MFCC + ∆MFCC+ ∆∆MFCC 9.09 90.91 0.1195
PROSODIC 26.50 73.50 0.4632
MFCC + ∆MFCC+ ∆∆MFCC+
PROSODIC
4.55 95.45 0.0823
5. CONCLUSIONS
The experiments reported in the above section established the fact that MFCC features along
with its time derivatives may be considered as an efficient parameterization of the speech signal
for speaker verification task. It has been observed that when MFCC with its 1st
and 2nd
order
derivatives has been considered as feature vector, it gives a recognition accuracy of around
92.43%. Performance of another feature vector namely prosody has also been evaluated in the
present study. It has been observed that when the prosodic features are considered as feature
vector for the speaker verification system, it gives a recognition accuracy of 76%, which is
nearly 16% below the recognition accuracy MFCC features vector. However, it has been
observed that when both MFCC and prosodic features are combined, a recognition accuracy of
95.45% has been achieved. Thus, it may be conclude that MFCC features when combined with
prosodic features, the performance of the system improves marginally, in the present study by
3%, without increasing the complexity of the system. Another interesting observation made in
the present study is that when English languages has been considered for training and testing the
system, a recognition accuracy of 90.19% has been achieved with MFCC features whereas
under same experimental condition the recognition accuracy for Local and Hindi languages is
93.19%. The relatively poor performance in case of English can be explained in the context of
linguistic scenario of Arunachal Pradesh. The people of Arunachal Pradesh of India, specially
the educated section use Hindi in their day-to-day conversation even with their family members.
Therefore, Hindi is as fluent to them as their native language and its articulation is relatively
robust to context like their native language. However, English being the non-native language
undergoes major deviation in the articulation with context.
ACKNOWLEDGEMENT
This work has been supported by the ongoing project grant No. 12(12)/2009-ESD sponsored by
the Department of Information Technology, Government of India.
7. Advanced Computing: An International Journal ( ACIJ ), Vol.4, No.1, January 2013
51
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