This document describes a speaker independent Malayalam word recognition system using support vector machines. It extracts mel frequency cepstral coefficients (MFCCs) as features and compares the performance of linear and radial basis function kernels in the SVM classifier. An overall accuracy of 91.8% was achieved on a database containing isolated word utterances from different speakers, demonstrating the effectiveness of the MFCC features and RBF kernel for speaker independent speech recognition.
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.
Developing a hands-free interface to operate a Computer using voice commandMohammad Liton Hossain
The main focus of this study is to help a handicap person to operate a computer by voice command. It can be used to operate the entire computer functions on the user’s voice commands. It makes use of the Speech Recognition technology that allows the computer system to identify and recognize words spoken by a human using a microphone. This Software will be able to recognize spoken words and enable user to interact with the computer. This interaction includes user giving commands to his computer which will then respond by performing several tasks, actions or operations depending on the commands they gave. For Example: Opening /closing a file in computer, YouTube automation using voice command, Google search using voice command, make a note using voice command, calculation by calculator using voice command etc.
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.
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.
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.
Developing a hands-free interface to operate a Computer using voice commandMohammad Liton Hossain
The main focus of this study is to help a handicap person to operate a computer by voice command. It can be used to operate the entire computer functions on the user’s voice commands. It makes use of the Speech Recognition technology that allows the computer system to identify and recognize words spoken by a human using a microphone. This Software will be able to recognize spoken words and enable user to interact with the computer. This interaction includes user giving commands to his computer which will then respond by performing several tasks, actions or operations depending on the commands they gave. For Example: Opening /closing a file in computer, YouTube automation using voice command, Google search using voice command, make a note using voice command, calculation by calculator using voice command etc.
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.
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.
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.
SPEAKER VERIFICATION USING ACOUSTIC AND PROSODIC FEATURESacijjournal
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.
Evaluation of Hidden Markov Model based Marathi Text-ToSpeech Synthesis SystemIJERA Editor
The objective of this paper is to evaluate the quality of HMM based Marathi TTS system. The main advantage of HMM technique is its ability to allow the variation in voice easily. The output speeches produced in this method have greater impact on emotion, style and intonation. The naturalness and intelligibility are the two important parameters to decide the quality of synthetic speech. Depending on the parameters specified the results of synthetic speech are categorized into 4 categories: natural speech, high quality synthetic speech, low quality synthetic speech and moderate quality synthetic speech. The results are obtained by using CT, DRT and MOS test.
International Journal of Engineering Research and Applications (IJERA) is an open access online peer reviewed international journal that publishes research and review articles in the fields of Computer Science, Neural Networks, Electrical Engineering, Software Engineering, Information Technology, Mechanical Engineering, Chemical Engineering, Plastic Engineering, Food Technology, Textile Engineering, Nano Technology & science, Power Electronics, Electronics & Communication Engineering, Computational mathematics, Image processing, Civil Engineering, Structural Engineering, Environmental Engineering, VLSI Testing & Low Power VLSI Design etc.
Speech Recognized Automation System Using Speaker Identification through Wire...IOSR Journals
This paper discusses the methodology for a project named “Speech Recognized Automation System
using Speaker Identification through wireless communication”. This project gives the design of Automation
system using wireless communication and speaker recognition using Matlab code. Straightforward
programming interface of Matlab makes it an ideal tool for speech analysis in project. This automation system
is useful for home appliances as well as in industry. This paper discusses the overall design of a wireless
automation system which is built and implemented. The speech recognition centers on recognition of speech
commands stored in data base of Matlab and it is matched with incoming voice command of speaker. Mel
Frequency Cepstral Coefficient (MFCC) algorithm is used to recognize the speech of speaker and to extract
features of speech. It uses low-power RF ZigBee transceiver wireless communication modules which are
relatively cheap. This automation system is intended to control lights, fans and other electrical appliances in a
home or office using speech commands like Light, Fan etc. Further, if security is not big issue then Speech
processor is used to control the appliances without speaker identification
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
An optimized approach to voice translation on mobile phoneseSAT Journals
Abstract Current voice translation tools and services use natural language understanding and natural language processing to convert words. However, these parsing methods concentrate more on capturing keywords and translating them, completely neglecting the considerable amount of processing time involved. In this paper, we are suggesting techniques that can optimize the processing time thereby increasing the throughput of voice translation services. Techniques like template matching, indexing frequently used words using probability search and session-based cache can considerably enhance processing times. More so, these factors become all the more important when we need to achieve real-time translation on mobile phones. Keywords:-Optimized voice translation, mobile client, speech recognition, language interpretation, template matching, probability search, session- based cache.
Approach of Syllable Based Unit Selection Text- To-Speech Synthesis System fo...iosrjce
IOSR journal of VLSI and Signal Processing (IOSRJVSP) is a double blind peer reviewed International Journal that publishes articles which contribute new results in all areas of VLSI Design & Signal Processing. The goal of this journal is to bring together researchers and practitioners from academia and industry to focus on advanced VLSI Design & Signal Processing concepts and establishing new collaborations in these areas.Design and realization of microelectronic systems using VLSI/ULSI technologies require close collaboration among scientists and engineers in the fields of systems architecture, logic and circuit design, chips and wafer fabrication, packaging, testing and systems applications. Generation of specifications, design and verification must be performed at all abstraction levels, including the system, register-transfer, logic, circuit, transistor and process levels.
Assistive Examination System for Visually ImpairedEditor IJCATR
This paper presents a design of voice enabled examination system which can be used by the visually challenged students.
The system uses Text-to-Speech (TTS) and Speech-to-Text (STT) technology. The text-to-speech and speech-to-text web based
academic testing software would provide an interaction for blind students to enhance their educational experiences by providing them
with a tool to give the exams. This system will aid the differently-abled to appear for online tests and enable them to come at par with
the other students. This system can also be used by students with learning disabilities or by people who wish to take the examination in
a combined auditory and visual way.
Utterance Based Speaker Identification Using ANNIJCSEA Journal
In this paper we present the implementation of speaker identification system using artificial neural network with digital signal processing. The system is designed to work with the text-dependent speaker identification for Bangla Speech. The utterances of speakers are recorded for specific Bangla words using an audio wave recorder. The speech features are acquired by the digital signal processing technique. The identification of speaker using frequency domain data is performed using back propagation algorithm. Hamming window and Blackman-Harris window are used to investigate better speaker identification performance. Endpoint detection of speech is developed in order to achieve high accuracy of the system.
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.
SPEAKER VERIFICATION USING ACOUSTIC AND PROSODIC FEATURESacijjournal
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.
Evaluation of Hidden Markov Model based Marathi Text-ToSpeech Synthesis SystemIJERA Editor
The objective of this paper is to evaluate the quality of HMM based Marathi TTS system. The main advantage of HMM technique is its ability to allow the variation in voice easily. The output speeches produced in this method have greater impact on emotion, style and intonation. The naturalness and intelligibility are the two important parameters to decide the quality of synthetic speech. Depending on the parameters specified the results of synthetic speech are categorized into 4 categories: natural speech, high quality synthetic speech, low quality synthetic speech and moderate quality synthetic speech. The results are obtained by using CT, DRT and MOS test.
International Journal of Engineering Research and Applications (IJERA) is an open access online peer reviewed international journal that publishes research and review articles in the fields of Computer Science, Neural Networks, Electrical Engineering, Software Engineering, Information Technology, Mechanical Engineering, Chemical Engineering, Plastic Engineering, Food Technology, Textile Engineering, Nano Technology & science, Power Electronics, Electronics & Communication Engineering, Computational mathematics, Image processing, Civil Engineering, Structural Engineering, Environmental Engineering, VLSI Testing & Low Power VLSI Design etc.
Speech Recognized Automation System Using Speaker Identification through Wire...IOSR Journals
This paper discusses the methodology for a project named “Speech Recognized Automation System
using Speaker Identification through wireless communication”. This project gives the design of Automation
system using wireless communication and speaker recognition using Matlab code. Straightforward
programming interface of Matlab makes it an ideal tool for speech analysis in project. This automation system
is useful for home appliances as well as in industry. This paper discusses the overall design of a wireless
automation system which is built and implemented. The speech recognition centers on recognition of speech
commands stored in data base of Matlab and it is matched with incoming voice command of speaker. Mel
Frequency Cepstral Coefficient (MFCC) algorithm is used to recognize the speech of speaker and to extract
features of speech. It uses low-power RF ZigBee transceiver wireless communication modules which are
relatively cheap. This automation system is intended to control lights, fans and other electrical appliances in a
home or office using speech commands like Light, Fan etc. Further, if security is not big issue then Speech
processor is used to control the appliances without speaker identification
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
An optimized approach to voice translation on mobile phoneseSAT Journals
Abstract Current voice translation tools and services use natural language understanding and natural language processing to convert words. However, these parsing methods concentrate more on capturing keywords and translating them, completely neglecting the considerable amount of processing time involved. In this paper, we are suggesting techniques that can optimize the processing time thereby increasing the throughput of voice translation services. Techniques like template matching, indexing frequently used words using probability search and session-based cache can considerably enhance processing times. More so, these factors become all the more important when we need to achieve real-time translation on mobile phones. Keywords:-Optimized voice translation, mobile client, speech recognition, language interpretation, template matching, probability search, session- based cache.
Approach of Syllable Based Unit Selection Text- To-Speech Synthesis System fo...iosrjce
IOSR journal of VLSI and Signal Processing (IOSRJVSP) is a double blind peer reviewed International Journal that publishes articles which contribute new results in all areas of VLSI Design & Signal Processing. The goal of this journal is to bring together researchers and practitioners from academia and industry to focus on advanced VLSI Design & Signal Processing concepts and establishing new collaborations in these areas.Design and realization of microelectronic systems using VLSI/ULSI technologies require close collaboration among scientists and engineers in the fields of systems architecture, logic and circuit design, chips and wafer fabrication, packaging, testing and systems applications. Generation of specifications, design and verification must be performed at all abstraction levels, including the system, register-transfer, logic, circuit, transistor and process levels.
Assistive Examination System for Visually ImpairedEditor IJCATR
This paper presents a design of voice enabled examination system which can be used by the visually challenged students.
The system uses Text-to-Speech (TTS) and Speech-to-Text (STT) technology. The text-to-speech and speech-to-text web based
academic testing software would provide an interaction for blind students to enhance their educational experiences by providing them
with a tool to give the exams. This system will aid the differently-abled to appear for online tests and enable them to come at par with
the other students. This system can also be used by students with learning disabilities or by people who wish to take the examination in
a combined auditory and visual way.
Utterance Based Speaker Identification Using ANNIJCSEA Journal
In this paper we present the implementation of speaker identification system using artificial neural network with digital signal processing. The system is designed to work with the text-dependent speaker identification for Bangla Speech. The utterances of speakers are recorded for specific Bangla words using an audio wave recorder. The speech features are acquired by the digital signal processing technique. The identification of speaker using frequency domain data is performed using back propagation algorithm. Hamming window and Blackman-Harris window are used to investigate better speaker identification performance. Endpoint detection of speech is developed in order to achieve high accuracy of the system.
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.
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.
ACHIEVING SECURITY VIA SPEECH RECOGNITIONijistjournal
Speech is one of the essential sources of the conversation between human beings. We as humans speak and listen to each other in human-human interface. People have tried to develop systems that can listen and prepare a speech as persons do so naturally. This paper presents a brief survey on Speech recognition, allow people to compose documents and control their computers with their voice. In other words, the process of enabling a machine (like a computer) to identify and respond to the sounds produced in human speech. ASR can be treated as the independent, computer-driven script of spoken language into readable text in real time. The Speech Recognition system requires careful attention to the following issues: Meaning of various types of speeches, speech representation, feature extraction techniques, speech classifiers, and database and performance evaluation. This paper helps in understanding the technique along with their pros and cons. A comparative study of different technique is done as per stages.
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.
Performance of different classifiers in speech recognitioneSAT Journals
Abstract Speech is the most natural means of communication among human beings and speech processing and recognition are intensive areas of research for the last five decades. Since speech recognition is a pattern recognition problem, classification is an important part of any speech recognition system. In this work, a speech recognition system is developed for recognizing speaker independent spoken digits in Malayalam. Voice signals are sampled directly from the microphone. The proposed method is implemented for 1000 speakers uttering 10 digits each. Since the speech signals are affected by background noise, the signals are tuned by removing the noise from it using wavelet denoising method based on Soft Thresholding. Here, the features from the signals are extracted using Discrete Wavelet Transforms (DWT) because they are well suitable for processing non-stationary signals like speech. This is due to their multi- resolutional, multi-scale analysis characteristics. Speech recognition is a multiclass classification problem. So, the feature vector set obtained are classified using three classifiers namely, Artificial Neural Networks (ANN), Support Vector Machines (SVM) and Naive Bayes classifiers which are capable of handling multiclasses. During classification stage, the input feature vector data is trained using information relating to known patterns and then they are tested using the test data set. The performances of all these classifiers are evaluated based on recognition accuracy. All the three methods produced good recognition accuracy. DWT and ANN produced a recognition accuracy of 89%, SVM and DWT combination produced an accuracy of 86.6% and Naive Bayes and DWT combination produced an accuracy of 83.5%. ANN is found to be better among the three methods. Index Terms: Speech Recognition, Soft Thresholding, Discrete Wavelet Transforms, Artificial Neural Networks, Support Vector Machines and Naive Bayes Classifier.
On Developing an Automatic Speech Recognition System for Commonly used Englis...rahulmonikasharma
Speech is one of the easiest and the fastest way to communicate. Recognition of speech by computer for various languages is a challenging task. The accuracy of Automatic speech recognition system (ASR) remains one of the key challenges, even after years of research. Accuracy varies due to speaker and language variability, vocabulary size and noise. Also, due to the design of speech recognition that is based on issues like- speech database, feature extraction techniques and performance evaluation. This paper aims to describe the development of a speaker-independent isolated automatic speech recognition system for Indian English language. The acoustic model is build using Carnegie Mellon University (CMU) Sphinx tools. The corpus used is based on Most Commonly used English words in everyday life. Speech database includes the recordings of 76 Punjabi Speakers (north-west Indian English accent). After testing, the system obtained an accuracy of 85.20 %, when trained using 128 GMMs (Gaussian Mixture Models).
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.
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.
A Survey on Speech Recognition with Language Specificationijtsrd
As a cross disciplinary, speech recognition is entirely based on the speech as the survey object. Speech recognition allows the machine to convert the speech signal into text or commands via the process of identification and understanding. Speech recognition involves in various fields of physiology, psychology, linguistics, computer science and signal processing, and is even related to the person’s body language, and its goal is to achieve natural language communication between man and machine. The speech recognition technology is gradually becoming the key technology of the IT man machine interface. This paper describes the development of speech recognition technology and its basic principles, methods, reviewed the classification of speech recognition systems, speech recognition approaches and voice recognition technology, analyzed the problems faced by the speech recognition. Dr. Preeti Savant | Lakshmi Sandhya H "A Survey on Speech Recognition with Language Specification" Published in International Journal of Trend in Scientific Research and Development (ijtsrd), ISSN: 2456-6470, Volume-6 | Issue-3 , April 2022, URL: https://www.ijtsrd.com/papers/ijtsrd49370.pdf Paper URL: https://www.ijtsrd.com/computer-science/speech-recognition/49370/a-survey-on-speech-recognition-with-language-specification/dr-preeti-savant
June 3, 2024 Anti-Semitism Letter Sent to MIT President Kornbluth and MIT Cor...Levi Shapiro
Letter from the Congress of the United States regarding Anti-Semitism sent June 3rd to MIT President Sally Kornbluth, MIT Corp Chair, Mark Gorenberg
Dear Dr. Kornbluth and Mr. Gorenberg,
The US House of Representatives is deeply concerned by ongoing and pervasive acts of antisemitic
harassment and intimidation at the Massachusetts Institute of Technology (MIT). Failing to act decisively to ensure a safe learning environment for all students would be a grave dereliction of your responsibilities as President of MIT and Chair of the MIT Corporation.
This Congress will not stand idly by and allow an environment hostile to Jewish students to persist. The House believes that your institution is in violation of Title VI of the Civil Rights Act, and the inability or
unwillingness to rectify this violation through action requires accountability.
Postsecondary education is a unique opportunity for students to learn and have their ideas and beliefs challenged. However, universities receiving hundreds of millions of federal funds annually have denied
students that opportunity and have been hijacked to become venues for the promotion of terrorism, antisemitic harassment and intimidation, unlawful encampments, and in some cases, assaults and riots.
The House of Representatives will not countenance the use of federal funds to indoctrinate students into hateful, antisemitic, anti-American supporters of terrorism. Investigations into campus antisemitism by the Committee on Education and the Workforce and the Committee on Ways and Means have been expanded into a Congress-wide probe across all relevant jurisdictions to address this national crisis. The undersigned Committees will conduct oversight into the use of federal funds at MIT and its learning environment under authorities granted to each Committee.
• The Committee on Education and the Workforce has been investigating your institution since December 7, 2023. The Committee has broad jurisdiction over postsecondary education, including its compliance with Title VI of the Civil Rights Act, campus safety concerns over disruptions to the learning environment, and the awarding of federal student aid under the Higher Education Act.
• The Committee on Oversight and Accountability is investigating the sources of funding and other support flowing to groups espousing pro-Hamas propaganda and engaged in antisemitic harassment and intimidation of students. The Committee on Oversight and Accountability is the principal oversight committee of the US House of Representatives and has broad authority to investigate “any matter” at “any time” under House Rule X.
• The Committee on Ways and Means has been investigating several universities since November 15, 2023, when the Committee held a hearing entitled From Ivory Towers to Dark Corners: Investigating the Nexus Between Antisemitism, Tax-Exempt Universities, and Terror Financing. The Committee followed the hearing with letters to those institutions on January 10, 202
Welcome to TechSoup New Member Orientation and Q&A (May 2024).pdfTechSoup
In this webinar you will learn how your organization can access TechSoup's wide variety of product discount and donation programs. From hardware to software, we'll give you a tour of the tools available to help your nonprofit with productivity, collaboration, financial management, donor tracking, security, and more.
Model Attribute Check Company Auto PropertyCeline George
In Odoo, the multi-company feature allows you to manage multiple companies within a single Odoo database instance. Each company can have its own configurations while still sharing common resources such as products, customers, and suppliers.
Read| The latest issue of The Challenger is here! We are thrilled to announce that our school paper has qualified for the NATIONAL SCHOOLS PRESS CONFERENCE (NSPC) 2024. Thank you for your unwavering support and trust. Dive into the stories that made us stand out!
A Strategic Approach: GenAI in EducationPeter Windle
Artificial Intelligence (AI) technologies such as Generative AI, Image Generators and Large Language Models have had a dramatic impact on teaching, learning and assessment over the past 18 months. The most immediate threat AI posed was to Academic Integrity with Higher Education Institutes (HEIs) focusing their efforts on combating the use of GenAI in assessment. Guidelines were developed for staff and students, policies put in place too. Innovative educators have forged paths in the use of Generative AI for teaching, learning and assessments leading to pockets of transformation springing up across HEIs, often with little or no top-down guidance, support or direction.
This Gasta posits a strategic approach to integrating AI into HEIs to prepare staff, students and the curriculum for an evolving world and workplace. We will highlight the advantages of working with these technologies beyond the realm of teaching, learning and assessment by considering prompt engineering skills, industry impact, curriculum changes, and the need for staff upskilling. In contrast, not engaging strategically with Generative AI poses risks, including falling behind peers, missed opportunities and failing to ensure our graduates remain employable. The rapid evolution of AI technologies necessitates a proactive and strategic approach if we are to remain relevant.
The French Revolution, which began in 1789, was a period of radical social and political upheaval in France. It marked the decline of absolute monarchies, the rise of secular and democratic republics, and the eventual rise of Napoleon Bonaparte. This revolutionary period is crucial in understanding the transition from feudalism to modernity in Europe.
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The Roman Empire A Historical Colossus.pdfkaushalkr1407
The Roman Empire, a vast and enduring power, stands as one of history's most remarkable civilizations, leaving an indelible imprint on the world. It emerged from the Roman Republic, transitioning into an imperial powerhouse under the leadership of Augustus Caesar in 27 BCE. This transformation marked the beginning of an era defined by unprecedented territorial expansion, architectural marvels, and profound cultural influence.
The empire's roots lie in the city of Rome, founded, according to legend, by Romulus in 753 BCE. Over centuries, Rome evolved from a small settlement to a formidable republic, characterized by a complex political system with elected officials and checks on power. However, internal strife, class conflicts, and military ambitions paved the way for the end of the Republic. Julius Caesar’s dictatorship and subsequent assassination in 44 BCE created a power vacuum, leading to a civil war. Octavian, later Augustus, emerged victorious, heralding the Roman Empire’s birth.
Under Augustus, the empire experienced the Pax Romana, a 200-year period of relative peace and stability. Augustus reformed the military, established efficient administrative systems, and initiated grand construction projects. The empire's borders expanded, encompassing territories from Britain to Egypt and from Spain to the Euphrates. Roman legions, renowned for their discipline and engineering prowess, secured and maintained these vast territories, building roads, fortifications, and cities that facilitated control and integration.
The Roman Empire’s society was hierarchical, with a rigid class system. At the top were the patricians, wealthy elites who held significant political power. Below them were the plebeians, free citizens with limited political influence, and the vast numbers of slaves who formed the backbone of the economy. The family unit was central, governed by the paterfamilias, the male head who held absolute authority.
Culturally, the Romans were eclectic, absorbing and adapting elements from the civilizations they encountered, particularly the Greeks. Roman art, literature, and philosophy reflected this synthesis, creating a rich cultural tapestry. Latin, the Roman language, became the lingua franca of the Western world, influencing numerous modern languages.
Roman architecture and engineering achievements were monumental. They perfected the arch, vault, and dome, constructing enduring structures like the Colosseum, Pantheon, and aqueducts. These engineering marvels not only showcased Roman ingenuity but also served practical purposes, from public entertainment to water supply.