This paper is aimed to implement Classification of Language Speech Recognition System by using feature extraction and classification. It is an Automatic language Speech Recognition system. This system is a software architecture which outputs digits from the input speech signals. The system is emphasized on Speaker Dependent Isolated Word Recognition System. To implement this system, a good quality microphone is required to record the speech signals. This system contains two main modules feature extraction and feature matching. Feature extraction is the process of extracting a small amount of data from the voice signal that can later be used to represent each speech signal. Feature matching involves the actual procedure to identify the unknown speech signal by comparing extracted features from the voice input of a set of known speech signals and the decision making process. In this system, the Mel frequency Cepstrum Coefficient MFCC is used for feature extraction and Vector Quantization VQ which uses the LBG algorithm is used for feature matching. Khin May Yee | Moh Moh Khaing | Thu Zar Aung "Classification of Language Speech Recognition System" Published in International Journal of Trend in Scientific Research and Development (ijtsrd), ISSN: 2456-6470, Volume-3 | Issue-5 , August 2019, URL: https://www.ijtsrd.com/papers/ijtsrd26546.pdfPaper URL: https://www.ijtsrd.com/computer-science/speech-recognition/26546/classification-of-language-speech-recognition-system/khin-may-yee
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 backpropagation 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.
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.
Financial Transactions in ATM Machines using Speech SignalsIJERA Editor
Speech is the natural and simplest way of communication and Speech Recognition is a fascinating application of Digital Signal Processing which has many real-world applications. In this paper, a speech recognition system is developed for Automated Teller Machines (ATMs) using Wavelet Packet Decomposition (WPD) and Artificial Neural Networks (ANN). Speech signals are one-dimensional and are random in nature. ATM machines communicate with the customers using the stored speech samples and the user communicates with the machine using spoken digits. Daubechies wavelets are employed here. A multilayer neural network trained with back propagation training algorithm is used for classification purpose. The proposed method is implemented for 500 speakers uttering 10 spoken digits in English. The experimental results show good recognition accuracy of 87.38% and the efficiency of combining these two techniques
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.
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.
Real Time Speaker Identification System – Design, Implementation and ValidationIDES Editor
This paper presents design, implementation and
validation of a PC based Prototype speaker recognition and
verification system. This system is organized to receive speech
signal, find the features of speech signal, and recognize and
verify a person using voice as the biometric. The system is
implemented to capture the speech signal from microphone
and to compare it with the stored data base using filter-bank
based closed-set speaker verification system. At first, the
identification of the voice signals is done using an algorithm
developed in MATLAB. Next, a PC based prototype system is
developed and is validated in real time. Several tests were made
on different sets of voice signals, and measured the performance
and the speed of the proposed system in real environment. The
result confirmed the use of proposed system for various real
time applications.
This paper is aimed to implement a robust speaker identification system. It is a software architecture which identifies the current talker out of a set of speakers. The system is emphasized on text-dependent speaker identification system. It contains three main modules: endpoint detection, feature extraction and feature matching. The additional module, endpoint detection, removes unwanted signal and background noise from the input speech signal before subsequent processing. In the proposed system, Short-Term Energy analysis is used for endpoint detection. Mel-frequency Cepstrum Coefficients (MFCC) is applied for feature extraction to extract a small amount of data from the voice signal that can later be used to represent each speaker. For feature matching, Vector Quantization (VQ) approach using Linde, Buzo and Gray (LBG) clustering algorithm is proposed because it can reduce the amount of data and complexity. The experimental study shows that the proposed system is more robust than using the original system and it is faster in computation than the existing one. To implement this system MATLAB is used for programming. Zaw Win Aung"A Robust Speaker Identification System" Published in International Journal of Trend in Scientific Research and Development (ijtsrd), ISSN: 2456-6470, Volume-2 | Issue-5 , August 2018, URL: http://www.ijtsrd.com/papers/ijtsrd18274.pdf http://www.ijtsrd.com/other-scientific-research-area/other/18274/a-robust-speaker-identification-system/zaw-win-aung
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 backpropagation 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.
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.
Financial Transactions in ATM Machines using Speech SignalsIJERA Editor
Speech is the natural and simplest way of communication and Speech Recognition is a fascinating application of Digital Signal Processing which has many real-world applications. In this paper, a speech recognition system is developed for Automated Teller Machines (ATMs) using Wavelet Packet Decomposition (WPD) and Artificial Neural Networks (ANN). Speech signals are one-dimensional and are random in nature. ATM machines communicate with the customers using the stored speech samples and the user communicates with the machine using spoken digits. Daubechies wavelets are employed here. A multilayer neural network trained with back propagation training algorithm is used for classification purpose. The proposed method is implemented for 500 speakers uttering 10 spoken digits in English. The experimental results show good recognition accuracy of 87.38% and the efficiency of combining these two techniques
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.
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.
Real Time Speaker Identification System – Design, Implementation and ValidationIDES Editor
This paper presents design, implementation and
validation of a PC based Prototype speaker recognition and
verification system. This system is organized to receive speech
signal, find the features of speech signal, and recognize and
verify a person using voice as the biometric. The system is
implemented to capture the speech signal from microphone
and to compare it with the stored data base using filter-bank
based closed-set speaker verification system. At first, the
identification of the voice signals is done using an algorithm
developed in MATLAB. Next, a PC based prototype system is
developed and is validated in real time. Several tests were made
on different sets of voice signals, and measured the performance
and the speed of the proposed system in real environment. The
result confirmed the use of proposed system for various real
time applications.
This paper is aimed to implement a robust speaker identification system. It is a software architecture which identifies the current talker out of a set of speakers. The system is emphasized on text-dependent speaker identification system. It contains three main modules: endpoint detection, feature extraction and feature matching. The additional module, endpoint detection, removes unwanted signal and background noise from the input speech signal before subsequent processing. In the proposed system, Short-Term Energy analysis is used for endpoint detection. Mel-frequency Cepstrum Coefficients (MFCC) is applied for feature extraction to extract a small amount of data from the voice signal that can later be used to represent each speaker. For feature matching, Vector Quantization (VQ) approach using Linde, Buzo and Gray (LBG) clustering algorithm is proposed because it can reduce the amount of data and complexity. The experimental study shows that the proposed system is more robust than using the original system and it is faster in computation than the existing one. To implement this system MATLAB is used for programming. Zaw Win Aung"A Robust Speaker Identification System" Published in International Journal of Trend in Scientific Research and Development (ijtsrd), ISSN: 2456-6470, Volume-2 | Issue-5 , August 2018, URL: http://www.ijtsrd.com/papers/ijtsrd18274.pdf http://www.ijtsrd.com/other-scientific-research-area/other/18274/a-robust-speaker-identification-system/zaw-win-aung
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 Emotion Recognition is a recent research topic in the Human Computer Interaction (HCI) field. The need has risen for a more natural communication interface between humans and computer, as computers have become an integral part of our lives. A lot of work currently going on to improve the interaction between humans and computers. To achieve this goal, a computer would have to be able to distinguish its present situation and respond differently depending on that observation. Part of this process involves understanding a user‟s emotional state. To make the human computer interaction more natural, the objective is that computer should be able to recognize emotional states in the same as human does. The efficiency of emotion recognition system depends on type of features extracted and classifier used for detection of emotions. The proposed system aims at identification of basic emotional states such as anger, joy, neutral and sadness from human speech. While classifying different emotions, features like MFCC (Mel Frequency Cepstral Coefficient) and Energy is used. In this paper, Standard Emotional Database i.e. English Database is used which gives the satisfactory detection of emotions than recorded samples of emotions. This methodology describes and compares the performances of Learning Vector Quantization Neural Network (LVQ NN), Multiclass Support Vector Machine (SVM) and their combination for emotion recognition.
ADAPTIVE WATERMARKING TECHNIQUE FOR SPEECH SIGNAL AUTHENTICATION ijcsit
Biometrics data recently has become a major role in determining the identity of the person. With such
importance for the use of biometrics data, there are many attacks that threaten the security and integrity of
biometrics data itself. Therefore, it becomes necessary to protect the originality of biometrics data against
manipulation and fraud. This paper presents an authentication technique to achieve the authenticity of
speech signals based on adaptive watermarking technique. The basic idea is depends on extracting the
speech features from the speech signal initially and then using these features as a watermark. The
watermark information embeds into the same speech signal. The short time energy technique is used to
identifying the suitable positions for embedding the watermark in order to avoid the regions that used in
the speech recognition system. After exclusion the important areas that used in speech recognition the
Genetic Algorithm (GA) is used to generate random locations to hide the watermark information in an
intelligent manner. The experimental results have achieved high efficiency in establishing the authenticity
of speech signal and the process of embedding
In the present-day communications speech signals get contaminated due to
various sorts of noises that degrade the speech quality and adversely impacts
speech recognition performance. To overcome these issues, a novel approach
for speech enhancement using Modified Wiener filtering is developed and
power spectrum computation is applied for degraded signal to obtain the
noise characteristics from a noisy spectrum. In next phase, MMSE technique
is applied where Gaussian distribution of each signal i.e. original and noisy
signal is analyzed. The Gaussian distribution provides spectrum estimation
and spectral coefficient parameters which can be used for probabilistic model
formulation. Moreover, a-priori-SNR computation is also incorporated for
coefficient updation and noise presence estimation which operates similar to
the conventional VAD. However, conventional VAD scheme is based on the
hard threshold which is not capable to derive satisfactory performance and a
soft-decision based threshold is developed for improving the performance of
speech enhancement. An extensive simulation study is carried out using
MATLAB simulation tool on NOIZEUS speech database and a comparative
study is presented where proposed approach is proved better in comparison
with existing technique.
We propose a model for carrying out deep learning based multimodal sentiment analysis. The MOUD dataset is taken for experimentation purposes. We developed two parallel text based and audio basedmodels and further, fused these heterogeneous feature maps taken from intermediate layers to complete thearchitecture. Performance measures–Accuracy, precision, recall and F1-score–are observed to outperformthe existing models.
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.
Audio Features Based Steganography Detection in WAV Fileijtsrd
Audio signals containing secret information or not is a security issue addressed in the context of steganalysis. ThRainfalle conceptual ide lies in the difference of the distribution of various statistical distance measures between the cover audio signals and stego audio signals. The aim of the propose system is to analyze the audio signal which have the presence of information hiding behavior or not. Mel frequency ceptral coefficient, zero crossing rate, spectral flux and short time energy features of audio signal are extracted, and combine these features with the features extracted from the modified version that is generated by randomly modifying with significant bits. Moreover, the extracted features are detected or classified with a support vector machine in this propose system. Experimental result show that the propose method performs well in steganalysis of the audio stegnograms that are produced by using S tools4. Khin Myo Kyi "Audio Features Based Steganography Detection in WAV File" Published in International Journal of Trend in Scientific Research and Development (ijtsrd), ISSN: 2456-6470, Volume-3 | Issue-5 , August 2019, URL: https://www.ijtsrd.com/papers/ijtsrd26807.pdf Paper URL: https://www.ijtsrd.com/computer-science/other/26807/audio-features-based-steganography-detection-in-wav-file/khin-myo-kyi
Text independent speaker identification system using average pitch and forman...ijitjournal
The aim of this paper is to design a closed-set text-independent Speaker Identification system using average
pitch and speech features from formant analysis. The speech features represented by the speech signal are
potentially characterized by formant analysis (Power Spectral Density). In this paper we have designed two
methods: one for average pitch estimation based on Autocorrelation and other for formant analysis. The
average pitches of speech signals are calculated and employed with formant analysis. From the performance
comparison of the proposed method with some of the existing methods, it is evident that the designed
speaker identification system with the proposed method is superior to others.
Voice Recognition System using Template MatchingIJORCS
It is easy for human to recognize familiar voice but using computer programs to identify a voice when compared with others is a herculean task. This is due to the problem that is encountered when developing the algorithm to recognize human voice. It is impossible to say a word the same way in two different occasions. Human speech analysis by computer gives different interpretation based on varying speed of speech delivery. This research paper gives detail description of the process behind implementation of an effective voice recognition algorithm. The algorithm utilize discrete Fourier transform to compare the frequency spectra of two voice samples because it remained unchanged as speech is slightly varied. Chebyshev inequality is then used to determine whether the two voices came from the same person. The algorithm is implemented and tested using MATLAB.
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.
On the use of voice activity detection in speech emotion recognitionjournalBEEI
Emotion recognition through speech has many potential applications, however the challenge comes from achieving a high emotion recognition while using limited resources or interference such as noise. In this paper we have explored the possibility of improving speech emotion recognition by utilizing the voice activity detection (VAD) concept. The emotional voice data from the Berlin Emotion Database (EMO-DB) and a custom-made database LQ Audio Dataset are firstly preprocessed by VAD before feature extraction. The features are then passed to the deep neural network for classification. In this paper, we have chosen MFCC to be the sole determinant feature. From the results obtained using VAD and without, we have found that the VAD improved the recognition rate of 5 emotions (happy, angry, sad, fear, and neutral) by 3.7% when recognizing clean signals, while the effect of using VAD when training a network with both clean and noisy signals improved our previous results by 50%.
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.
The paper presents a k-means based semi-supervised clustering approach for
recognizing and classifying P300 signals for BCI Speller System. P300 signals are proved to
be the most suitable Event Related Potential (ERP) signal, used to develop the BCI systems.
Due to non-stationary nature of ERP signals, the wavelet transform is the best analysis tool
for extracting informative features from P300 signals. The focus of the research is on semi-
supervised clustering as supervised clustering approach need large amount of labeled data
for training, which is a tedious task. Hence works for small-labeled datasets to train
classifiers. On the other hand, unsupervised clustering works when no prior information is
available i.e. totally unlabeled data. Thus leads to low level of performance. The in-between
solution is to use semi-supervised clustering, which uses a few labeled with large unlabeled
data causes less trouble and time. The authors have selected and defined adhoc features and
assumed the Clusters for small datasets. This motivates us to propose a novel approach that
discovers the features embedded in P300 (EEG) signals, using an k-means based semi-
supervised cluster classification using ensemble SVM
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.
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 backpropagation 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.
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 Emotion Recognition is a recent research topic in the Human Computer Interaction (HCI) field. The need has risen for a more natural communication interface between humans and computer, as computers have become an integral part of our lives. A lot of work currently going on to improve the interaction between humans and computers. To achieve this goal, a computer would have to be able to distinguish its present situation and respond differently depending on that observation. Part of this process involves understanding a user‟s emotional state. To make the human computer interaction more natural, the objective is that computer should be able to recognize emotional states in the same as human does. The efficiency of emotion recognition system depends on type of features extracted and classifier used for detection of emotions. The proposed system aims at identification of basic emotional states such as anger, joy, neutral and sadness from human speech. While classifying different emotions, features like MFCC (Mel Frequency Cepstral Coefficient) and Energy is used. In this paper, Standard Emotional Database i.e. English Database is used which gives the satisfactory detection of emotions than recorded samples of emotions. This methodology describes and compares the performances of Learning Vector Quantization Neural Network (LVQ NN), Multiclass Support Vector Machine (SVM) and their combination for emotion recognition.
ADAPTIVE WATERMARKING TECHNIQUE FOR SPEECH SIGNAL AUTHENTICATION ijcsit
Biometrics data recently has become a major role in determining the identity of the person. With such
importance for the use of biometrics data, there are many attacks that threaten the security and integrity of
biometrics data itself. Therefore, it becomes necessary to protect the originality of biometrics data against
manipulation and fraud. This paper presents an authentication technique to achieve the authenticity of
speech signals based on adaptive watermarking technique. The basic idea is depends on extracting the
speech features from the speech signal initially and then using these features as a watermark. The
watermark information embeds into the same speech signal. The short time energy technique is used to
identifying the suitable positions for embedding the watermark in order to avoid the regions that used in
the speech recognition system. After exclusion the important areas that used in speech recognition the
Genetic Algorithm (GA) is used to generate random locations to hide the watermark information in an
intelligent manner. The experimental results have achieved high efficiency in establishing the authenticity
of speech signal and the process of embedding
In the present-day communications speech signals get contaminated due to
various sorts of noises that degrade the speech quality and adversely impacts
speech recognition performance. To overcome these issues, a novel approach
for speech enhancement using Modified Wiener filtering is developed and
power spectrum computation is applied for degraded signal to obtain the
noise characteristics from a noisy spectrum. In next phase, MMSE technique
is applied where Gaussian distribution of each signal i.e. original and noisy
signal is analyzed. The Gaussian distribution provides spectrum estimation
and spectral coefficient parameters which can be used for probabilistic model
formulation. Moreover, a-priori-SNR computation is also incorporated for
coefficient updation and noise presence estimation which operates similar to
the conventional VAD. However, conventional VAD scheme is based on the
hard threshold which is not capable to derive satisfactory performance and a
soft-decision based threshold is developed for improving the performance of
speech enhancement. An extensive simulation study is carried out using
MATLAB simulation tool on NOIZEUS speech database and a comparative
study is presented where proposed approach is proved better in comparison
with existing technique.
We propose a model for carrying out deep learning based multimodal sentiment analysis. The MOUD dataset is taken for experimentation purposes. We developed two parallel text based and audio basedmodels and further, fused these heterogeneous feature maps taken from intermediate layers to complete thearchitecture. Performance measures–Accuracy, precision, recall and F1-score–are observed to outperformthe existing models.
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.
Audio Features Based Steganography Detection in WAV Fileijtsrd
Audio signals containing secret information or not is a security issue addressed in the context of steganalysis. ThRainfalle conceptual ide lies in the difference of the distribution of various statistical distance measures between the cover audio signals and stego audio signals. The aim of the propose system is to analyze the audio signal which have the presence of information hiding behavior or not. Mel frequency ceptral coefficient, zero crossing rate, spectral flux and short time energy features of audio signal are extracted, and combine these features with the features extracted from the modified version that is generated by randomly modifying with significant bits. Moreover, the extracted features are detected or classified with a support vector machine in this propose system. Experimental result show that the propose method performs well in steganalysis of the audio stegnograms that are produced by using S tools4. Khin Myo Kyi "Audio Features Based Steganography Detection in WAV File" Published in International Journal of Trend in Scientific Research and Development (ijtsrd), ISSN: 2456-6470, Volume-3 | Issue-5 , August 2019, URL: https://www.ijtsrd.com/papers/ijtsrd26807.pdf Paper URL: https://www.ijtsrd.com/computer-science/other/26807/audio-features-based-steganography-detection-in-wav-file/khin-myo-kyi
Text independent speaker identification system using average pitch and forman...ijitjournal
The aim of this paper is to design a closed-set text-independent Speaker Identification system using average
pitch and speech features from formant analysis. The speech features represented by the speech signal are
potentially characterized by formant analysis (Power Spectral Density). In this paper we have designed two
methods: one for average pitch estimation based on Autocorrelation and other for formant analysis. The
average pitches of speech signals are calculated and employed with formant analysis. From the performance
comparison of the proposed method with some of the existing methods, it is evident that the designed
speaker identification system with the proposed method is superior to others.
Voice Recognition System using Template MatchingIJORCS
It is easy for human to recognize familiar voice but using computer programs to identify a voice when compared with others is a herculean task. This is due to the problem that is encountered when developing the algorithm to recognize human voice. It is impossible to say a word the same way in two different occasions. Human speech analysis by computer gives different interpretation based on varying speed of speech delivery. This research paper gives detail description of the process behind implementation of an effective voice recognition algorithm. The algorithm utilize discrete Fourier transform to compare the frequency spectra of two voice samples because it remained unchanged as speech is slightly varied. Chebyshev inequality is then used to determine whether the two voices came from the same person. The algorithm is implemented and tested using MATLAB.
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.
On the use of voice activity detection in speech emotion recognitionjournalBEEI
Emotion recognition through speech has many potential applications, however the challenge comes from achieving a high emotion recognition while using limited resources or interference such as noise. In this paper we have explored the possibility of improving speech emotion recognition by utilizing the voice activity detection (VAD) concept. The emotional voice data from the Berlin Emotion Database (EMO-DB) and a custom-made database LQ Audio Dataset are firstly preprocessed by VAD before feature extraction. The features are then passed to the deep neural network for classification. In this paper, we have chosen MFCC to be the sole determinant feature. From the results obtained using VAD and without, we have found that the VAD improved the recognition rate of 5 emotions (happy, angry, sad, fear, and neutral) by 3.7% when recognizing clean signals, while the effect of using VAD when training a network with both clean and noisy signals improved our previous results by 50%.
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.
The paper presents a k-means based semi-supervised clustering approach for
recognizing and classifying P300 signals for BCI Speller System. P300 signals are proved to
be the most suitable Event Related Potential (ERP) signal, used to develop the BCI systems.
Due to non-stationary nature of ERP signals, the wavelet transform is the best analysis tool
for extracting informative features from P300 signals. The focus of the research is on semi-
supervised clustering as supervised clustering approach need large amount of labeled data
for training, which is a tedious task. Hence works for small-labeled datasets to train
classifiers. On the other hand, unsupervised clustering works when no prior information is
available i.e. totally unlabeled data. Thus leads to low level of performance. The in-between
solution is to use semi-supervised clustering, which uses a few labeled with large unlabeled
data causes less trouble and time. The authors have selected and defined adhoc features and
assumed the Clusters for small datasets. This motivates us to propose a novel approach that
discovers the features embedded in P300 (EEG) signals, using an k-means based semi-
supervised cluster classification using ensemble SVM
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.
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 backpropagation 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.
Speaker Recognition System using MFCC and Vector Quantization Approachijsrd.com
This paper presents an approach to speaker recognition using frequency spectral information with Mel frequency for the improvement of speech feature representation in a Vector Quantization codebook based recognition approach. The Mel frequency approach extracts the features of the speech signal to get the training and testing vectors. The VQ Codebook approach uses training vectors to form clusters and recognize accurately with the help of LBG algorithm.
GENDER RECOGNITION SYSTEM USING SPEECH SIGNALIJCSEIT Journal
In this paper, a system, developed for speech encoding, analysis, synthesis and gender identification is
presented. A typical gender recognition system can be divided into front-end system and back-end system.
The task of the front-end system is to extract the gender related information from a speech signal and
represents it by a set of vectors called feature. Features like power spectrum density, frequency at
maximum power carry speaker information. The feature is extracted using First Fourier Transform (FFT)
algorithm. The task of the back-end system (also called classifier) is to create a gender model to recognize
the gender from his/her speech signal in recognition phase. This paper also presents the digital processing
of a speech signals (pronounced “A” and “B”) which are taken from 10 persons, 5 of them are Male and
the rest of them are Female. Power Spectrum Estimation of the signal is examined .The frequency at
maximum power of the English Phonemes is extracted from the estimated power spectrum. The system uses
threshold technique as identification tool. The recognition accuracy of this system is 80% on average.
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.
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.
Forensic and Automatic Speaker Recognition System IJECEIAES
Current Automatic Speaker Recognition (ASR) System has emerged as an important medium of confirmation of identity in many businesses, ecommerce applications, forensics and law enforcement as well. Specialists trained in criminological recognition can play out this undertaking far superior by looking at an arrangement of acoustic, prosodic, and semantic attributes which has been referred to as structured listening. An algorithmbased system has been developed in the recognition of forensic speakers by physics scientists and forensic linguists to reduce the probability of a contextual bias or pre-centric understanding of a reference model with the validity of an unknown audio s ample and any suspicious individual. Many researchers are continuing to develop automatic algorithms in signal processing and machine learning so that improving performance can effectively introduce the speaker’s identity, where the automatic system performs equally with the human audience. In this paper, I examine the literature about the identification of speakers by machines and humans, emphasizing the key technical speaker pattern emerging for the automatic technology in the last decade. I focus on many aspects of automatic speaker recognition (ASR) systems, including speaker-specific features, speaker models, standard assessment data sets, and performance metrics.
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.
‘Six Sigma Technique’ A Journey Through its Implementationijtsrd
The manufacturing industries all over the world are facing tough challenges for growth, development and sustainability in today’s competitive environment. They have to achieve apex position by adapting with the global competitive environment by delivering goods and services at low cost, prime quality and better price to increase wealth and consumer satisfaction. Cost Management ensures profit, growth and sustainability of the business with implementation of Continuous Improvement Technique like Six Sigma. This leads to optimize Business performance. The method drives for customer satisfaction, low variation, reduction in waste and cycle time resulting into a competitive advantage over other industries which did not implement it. The main objective of this paper ‘Six Sigma Technique A Journey Through Its Implementation’ is to conceptualize the effectiveness of Six Sigma Technique through the journey of its implementation. Aditi Sunilkumar Ghosalkar "‘Six Sigma Technique’: A Journey Through its Implementation" Published in International Journal of Trend in Scientific Research and Development (ijtsrd), ISSN: 2456-6470, Volume-8 | Issue-1 , February 2024, URL: https://www.ijtsrd.com/papers/ijtsrd64546.pdf Paper Url: https://www.ijtsrd.com/other-scientific-research-area/other/64546/‘six-sigma-technique’-a-journey-through-its-implementation/aditi-sunilkumar-ghosalkar
Edge Computing in Space Enhancing Data Processing and Communication for Space...ijtsrd
Edge computing, a paradigm that involves processing data closer to its source, has gained significant attention for its potential to revolutionize data processing and communication in space missions. With the increasing complexity and data volume generated by modern space missions, traditional centralized computing approaches face challenges related to latency, bandwidth, and security. Edge computing in space, involving on board processing and analysis of data, offers promising solutions to these challenges. This paper explores the concept of edge computing in space, its benefits, applications, and future prospects in enhancing space missions. Manish Verma "Edge Computing in Space: Enhancing Data Processing and Communication for Space Missions" Published in International Journal of Trend in Scientific Research and Development (ijtsrd), ISSN: 2456-6470, Volume-8 | Issue-1 , February 2024, URL: https://www.ijtsrd.com/papers/ijtsrd64541.pdf Paper Url: https://www.ijtsrd.com/computer-science/artificial-intelligence/64541/edge-computing-in-space-enhancing-data-processing-and-communication-for-space-missions/manish-verma
Dynamics of Communal Politics in 21st Century India Challenges and Prospectsijtsrd
Communal politics in India has evolved through centuries, weaving a complex tapestry shaped by historical legacies, colonial influences, and contemporary socio political transformations. This research comprehensively examines the dynamics of communal politics in 21st century India, emphasizing its historical roots, socio political dynamics, economic implications, challenges, and prospects for mitigation. The historical perspective unravels the intricate interplay of religious identities and power dynamics from ancient civilizations to the impact of colonial rule, providing insights into the evolution of communalism. The socio political dynamics section delves into the contemporary manifestations, exploring the roles of identity politics, socio economic disparities, and globalization. The economic implications section highlights how communal politics intersects with economic issues, perpetuating disparities and influencing resource allocation. Challenges posed by communal politics are scrutinized, revealing multifaceted issues ranging from social fragmentation to threats against democratic values. The prospects for mitigation present a multifaceted approach, incorporating policy interventions, community engagement, and educational initiatives. The paper conducts a comparative analysis with international examples, identifying common patterns such as identity politics and economic disparities. It also examines unique challenges, emphasizing Indias diverse religious landscape, historical legacy, and secular framework. Lessons for effective strategies are drawn from international experiences, offering insights into inclusive policies, interfaith dialogue, media regulation, and global cooperation. By scrutinizing historical epochs, contemporary dynamics, economic implications, and international comparisons, this research provides a comprehensive understanding of communal politics in India. The proposed strategies for mitigation underscore the importance of a holistic approach to foster social harmony, inclusivity, and democratic values. Rose Hossain "Dynamics of Communal Politics in 21st Century India: Challenges and Prospects" Published in International Journal of Trend in Scientific Research and Development (ijtsrd), ISSN: 2456-6470, Volume-8 | Issue-1 , February 2024, URL: https://www.ijtsrd.com/papers/ijtsrd64528.pdf Paper Url: https://www.ijtsrd.com/humanities-and-the-arts/history/64528/dynamics-of-communal-politics-in-21st-century-india-challenges-and-prospects/rose-hossain
Assess Perspective and Knowledge of Healthcare Providers Towards Elehealth in...ijtsrd
Background and Objective Telehealth has become a well known tool for the delivery of health care in Saudi Arabia, and the perspective and knowledge of healthcare providers are influential in the implementation, adoption and advancement of the method. This systematic review was conducted to examine the current literature base regarding telehealth and the related healthcare professional perspective and knowledge in the Kingdom of Saudi Arabia. Materials and Methods This systematic review was conducted by searching 7 databases including, MEDLINE, CINHAL, Web of Science, Scopus, PubMed, PsycINFO, and ProQuest Central. Studies on healthcare practitioners telehealth knowledge and perspectives published in English in Saudi Arabia from 2000 to 2023 were included. Boland directed this comprehensive review. The researchers examined each connected study using the AXIS tool, which evaluates cross sectional systematic reviews. Narrative synthesis was used to summarise and convey the data. Results Out of 1840 search results, 10 studies were included. Positive outlook and limited knowledge among providers were seen across trials. Healthcare professionals like telehealth for its ability to improve quality, access, and delivery, save time and money, and be successful. Age, gender, occupation, and work experience also affect health workers knowledge. In Saudi Arabia, healthcare professionals face inadequate expert assistance, patient privacy, internet connection concerns, lack of training courses, lack of telehealth understanding, and high costs while performing telemedicine. Conclusions Healthcare practitioners telehealth perceptions and knowledge were examined in this systematic study. Its collection of concerned experts different personal attitudes and expertise would help enhance telehealths implementation in Saudi Arabia, develop its healthcare delivery alternative, and eliminate frequent problems. Badriah Mousa I Mulayhi | Dr. Jomin George | Judy Jenkins "Assess Perspective and Knowledge of Healthcare Providers Towards Elehealth in Saudi Arabia: A Systematic Review" Published in International Journal of Trend in Scientific Research and Development (ijtsrd), ISSN: 2456-6470, Volume-8 | Issue-1 , February 2024, URL: https://www.ijtsrd.com/papers/ijtsrd64535.pdf Paper Url: https://www.ijtsrd.com/medicine/other/64535/assess-perspective-and-knowledge-of-healthcare-providers-towards-elehealth-in-saudi-arabia-a-systematic-review/badriah-mousa-i-mulayhi
The Impact of Digital Media on the Decentralization of Power and the Erosion ...ijtsrd
The impact of digital media on the distribution of power and the weakening of traditional gatekeepers has gained considerable attention in recent years. The adoption of digital technologies and the internet has resulted in declining influence and power for traditional gatekeepers such as publishing houses and news organizations. Simultaneously, digital media has facilitated the emergence of new voices and players in the media industry. Digital medias impact on power decentralization and gatekeeper erosion is visible in several ways. One significant aspect is the democratization of information, which enables anyone with an internet connection to publish and share content globally, leading to citizen journalism and bypassing traditional gatekeepers. Another aspect is the disruption of conventional media industry business models, as traditional organizations struggle to adjust to the decrease in advertising revenue and the rise of digital platforms. Alternative business models, such as subscription models and crowdfunding, have become more prevalent, leading to the emergence of new players. Overall, the impact of digital media on the distribution of power and the weakening of traditional gatekeepers has brought about significant changes in the media landscape and the way information is shared. Further research is required to fully comprehend the implications of these changes and their impact on society. Dr. Kusum Lata "The Impact of Digital Media on the Decentralization of Power and the Erosion of Traditional Gatekeepers" Published in International Journal of Trend in Scientific Research and Development (ijtsrd), ISSN: 2456-6470, Volume-8 | Issue-1 , February 2024, URL: https://www.ijtsrd.com/papers/ijtsrd64544.pdf Paper Url: https://www.ijtsrd.com/humanities-and-the-arts/political-science/64544/the-impact-of-digital-media-on-the-decentralization-of-power-and-the-erosion-of-traditional-gatekeepers/dr-kusum-lata
Online Voices, Offline Impact Ambedkars Ideals and Socio Political Inclusion ...ijtsrd
This research investigates the nexus between online discussions on Dr. B.R. Ambedkars ideals and their impact on social inclusion among college students in Gurugram, Haryana. Surveying 240 students from 12 government colleges, findings indicate that 65 actively engage in online discussions, with 80 demonstrating moderate to high awareness of Ambedkars ideals. Statistically significant correlations reveal that higher online engagement correlates with increased awareness p 0.05 and perceived social inclusion. Variations across colleges and a notable effect of college type on perceived social inclusion highlight the influence of contextual factors. Furthermore, the intersectional analysis underscores nuanced differences based on gender, caste, and socio economic status. Dr. Kusum Lata "Online Voices, Offline Impact: Ambedkar's Ideals and Socio-Political Inclusion - A Study of Gurugram District" Published in International Journal of Trend in Scientific Research and Development (ijtsrd), ISSN: 2456-6470, Volume-8 | Issue-1 , February 2024, URL: https://www.ijtsrd.com/papers/ijtsrd64543.pdf Paper Url: https://www.ijtsrd.com/humanities-and-the-arts/political-science/64543/online-voices-offline-impact-ambedkars-ideals-and-sociopolitical-inclusion--a-study-of-gurugram-district/dr-kusum-lata
Problems and Challenges of Agro Entreprenurship A Studyijtsrd
Noting calls for contextualizing Agro entrepreneurs problems and challenges of the agro entrepreneurs and for greater attention to the Role of entrepreneurs in agro entrepreneurship research, we conduct a systematic literature review of extent research in agriculture entrepreneurship to overcome the study objectives of complications of agro entrepreneurs through various factors, Development of agriculture products is a key factor for the overall economic growth of agro entrepreneurs Agro Entrepreneurs produces firsthand large scale employment, utilizes the labor and natural resources, This research outlines the problems of Weather and Soil Erosions, Market price fluctuation, stimulates labor cost problems, reduces concentration of Price volatility, Dependency on Intermediaries, induces Limited Bargaining Power, and Storage and Transportation Costs. This paper mainly devoted to highlight Problems and challenges faced for the sustainable of Agro Entrepreneurs in India. Vinay Prasad B "Problems and Challenges of Agro Entreprenurship - A Study" Published in International Journal of Trend in Scientific Research and Development (ijtsrd), ISSN: 2456-6470, Volume-8 | Issue-1 , February 2024, URL: https://www.ijtsrd.com/papers/ijtsrd64540.pdf Paper Url: https://www.ijtsrd.com/other-scientific-research-area/other/64540/problems-and-challenges-of-agro-entreprenurship--a-study/vinay-prasad-b
Comparative Analysis of Total Corporate Disclosure of Selected IT Companies o...ijtsrd
Disclosure is a process through which a business enterprise communicates with external parties. A corporate disclosure is communication of financial and non financial information of the activities of a business enterprise to the interested entities. Corporate disclosure is done through publishing annual reports. So corporate disclosure through annual reports plays a vital role in the life of all the companies and provides valuable information to investors. The basic objectives of corporate disclosure is to give a true and fair view of companies to the parties related either directly or indirectly like owner, government, creditors, shareholders etc. in the companies act, provisions have been made about mandatory and voluntary disclosure. The IT sector in India is rapidly growing, the trend to invest in the IT sector is rising and employment opportunities in IT sectors are also increasing. Therefore the IT sector is expected to have fair, full and adequate disclosure of all information. Unfair and incomplete disclosure may adversely affect the entire economy. A research study on disclosure practices of IT companies could play an important role in this regard. Hence, the present research study has been done to study and review comparative analysis of total corporate disclosure of selected IT companies of India and to put forward overall findings and suggestions with a view to increase disclosure score of these companies. The researcher hopes that the present research study will be helpful to all selected Companies for improving level of corporate disclosure through annual reports as well as the government, creditors, investors, all business organizations and upcoming researcher for comparative analyses of level of corporate disclosure with special reference to selected IT companies. Dr. Vaibhavi D. Thaker "Comparative Analysis of Total Corporate Disclosure of Selected IT Companies of India" Published in International Journal of Trend in Scientific Research and Development (ijtsrd), ISSN: 2456-6470, Volume-8 | Issue-1 , February 2024, URL: https://www.ijtsrd.com/papers/ijtsrd64539.pdf Paper Url: https://www.ijtsrd.com/other-scientific-research-area/other/64539/comparative-analysis-of-total-corporate-disclosure-of-selected-it-companies-of-india/dr-vaibhavi-d-thaker
The Impact of Educational Background and Professional Training on Human Right...ijtsrd
This study investigated the impact of educational background and professional training on human rights awareness among secondary school teachers in the Marathwada region of Maharashtra, India. The key findings reveal that higher levels of education, particularly a master’s degree, and fields of study related to education, humanities, or social sciences are associated with greater human rights awareness among teachers. Additionally, both pre service teacher training and in service professional development programs focused on human rights education significantly enhance teacher’s knowledge, skills, and competencies in promoting human rights principles in their classrooms. Baig Ameer Bee Mirza Abdul Aziz | Dr. Syed Azaz Ali Amjad Ali "The Impact of Educational Background and Professional Training on Human Rights Awareness among Secondary School Teachers" Published in International Journal of Trend in Scientific Research and Development (ijtsrd), ISSN: 2456-6470, Volume-8 | Issue-1 , February 2024, URL: https://www.ijtsrd.com/papers/ijtsrd64529.pdf Paper Url: https://www.ijtsrd.com/humanities-and-the-arts/education/64529/the-impact-of-educational-background-and-professional-training-on-human-rights-awareness-among-secondary-school-teachers/baig-ameer-bee-mirza-abdul-aziz
A Study on the Effective Teaching Learning Process in English Curriculum at t...ijtsrd
“One Language sets you in a corridor for life. Two languages open every door along the way” Frank Smith English as a foreign language or as a second language has been ruling in India since the period of Lord Macaulay. But the question is how much we teach or learn English properly in our culture. Is there any scope to use English as a language rather than a subject How much we learn or teach English without any interference of mother language specially in the classroom teaching learning scenario in West Bengal By considering all these issues the researcher has attempted in this article to focus on the effective teaching learning process comparing to other traditional strategies in the field of English curriculum at the secondary level to investigate whether they fulfill the present teaching learning requirements or not by examining the validity of the present curriculum of English. The purpose of this study is to focus on the effectiveness of the systematic, scientific, sequential and logical transaction of the course between the teachers and the learners in the perspective of the 5Es programme that is engage, explore, explain, extend and evaluate. Sanchali Mondal | Santinath Sarkar "A Study on the Effective Teaching Learning Process in English Curriculum at the Secondary Level of West Bengal" Published in International Journal of Trend in Scientific Research and Development (ijtsrd), ISSN: 2456-6470, Volume-8 | Issue-1 , February 2024, URL: https://www.ijtsrd.com/papers/ijtsrd62412.pdf Paper Url: https://www.ijtsrd.com/humanities-and-the-arts/education/62412/a-study-on-the-effective-teaching-learning-process-in-english-curriculum-at-the-secondary-level-of-west-bengal/sanchali-mondal
The Role of Mentoring and Its Influence on the Effectiveness of the Teaching ...ijtsrd
This paper reports on a study which was conducted to investigate the role of mentoring and its influence on the effectiveness of the teaching of Physics in secondary schools in the South West Region of Cameroon. The study adopted the convergent parallel mixed methods design, focusing on respondents in secondary schools in the South West Region of Cameroon. Both quantitative and qualitative data were collected, analysed separately, and the results were compared to see if the findings confirm or disconfirm each other. The quantitative analysis found that majority of the respondents 72 of Physics teachers affirmed that they had more experienced colleagues as mentors to help build their confidence, improve their teaching, and help them improve their effectiveness and efficiency in guiding learners’ achievements. Only 28 of the respondents disagreed with these statements. With majority respondents 72 agreeing with the statements, it implies that in most secondary schools, experienced Physics teachers act as mentors to build teachers’ confidence in teaching and improving students’ learning. The interview qualitative data analysis summarized how secondary school Principals use meetings with mentors and mentees to promote mentorship in the school milieu. This has helped strengthen teachers’ classroom practices in secondary schools in the South West Region of Cameroon. With the results confirming each other, the study recommends that mentoring should focus on helping teachers employ social interactions and instructional practices feedback and clarity in teaching that have direct measurable impact on students’ learning achievements. Andrew Ngeim Sumba | Frederick Ebot Ashu | Peter Agborbechem Tambi "The Role of Mentoring and Its Influence on the Effectiveness of the Teaching of Physics in Secondary Schools in the South West Region of Cameroon" Published in International Journal of Trend in Scientific Research and Development (ijtsrd), ISSN: 2456-6470, Volume-8 | Issue-1 , February 2024, URL: https://www.ijtsrd.com/papers/ijtsrd64524.pdf Paper Url: https://www.ijtsrd.com/management/management-development/64524/the-role-of-mentoring-and-its-influence-on-the-effectiveness-of-the-teaching-of-physics-in-secondary-schools-in-the-south-west-region-of-cameroon/andrew-ngeim-sumba
Design Simulation and Hardware Construction of an Arduino Microcontroller Bas...ijtsrd
This study primarily focuses on the design of a high side buck converter using an Arduino microcontroller. The converter is specifically intended for use in DC DC applications, particularly in standalone solar PV systems where the PV output voltage exceeds the load or battery voltage. To evaluate the performance of the converter, simulation experiments are conducted using Proteus Software. These simulations provide insights into the input and output voltages, currents, powers, and efficiency under different state of charge SoC conditions of a 12V,70Ah rechargeable lead acid battery. Additionally, the hardware design of the converter is implemented, and practical data is collected through operation, monitoring, and recording. By comparing the simulation results with the practical results, the efficiency and performance of the designed converter are assessed. The findings indicate that while the buck converter is suitable for practical use in standalone PV systems, its efficiency is compromised due to a lower output current. Chan Myae Aung | Dr. Ei Mon "Design Simulation and Hardware Construction of an Arduino-Microcontroller Based DC-DC High-Side Buck Converter for Standalone PV System" Published in International Journal of Trend in Scientific Research and Development (ijtsrd), ISSN: 2456-6470, Volume-8 | Issue-1 , February 2024, URL: https://www.ijtsrd.com/papers/ijtsrd64518.pdf Paper Url: https://www.ijtsrd.com/engineering/mechanical-engineering/64518/design-simulation-and-hardware-construction-of-an-arduinomicrocontroller-based-dcdc-highside-buck-converter-for-standalone-pv-system/chan-myae-aung
Sustainable Energy by Paul A. Adekunte | Matthew N. O. Sadiku | Janet O. Sadikuijtsrd
Energy becomes sustainable if it meets the needs of the present without compromising the ability of future generations to meet their own needs. Some of the definitions of sustainable energy include the considerations of environmental aspects such as greenhouse gas emissions, social, and economic aspects such as energy poverty. Generally far more sustainable than fossil fuel are renewable energy sources such as wind, hydroelectric power, solar, and geothermal energy sources. Worthy of note is that some renewable energy projects, like the clearing of forests to produce biofuels, can cause severe environmental damage. The sustainability of nuclear power which is a low carbon source is highly debated because of concerns about radioactive waste, nuclear proliferation, and accidents. The switching from coal to natural gas has environmental benefits, including a lower climate impact, but could lead to delay in switching to more sustainable options. “Carbon capture and storage” can be built into power plants to remove the carbon dioxide CO2 emissions, but this technology is expensive and has rarely been implemented. Leading non renewable energy sources around the world is fossil fuels, coal, petroleum, and natural gas. Nuclear energy is usually considered another non renewable energy source, although nuclear energy itself is a renewable energy source, but the material used in nuclear power plants is not. The paper addresses the issue of sustainable energy, its attendant benefits to the future generation, and humanity in general. Paul A. Adekunte | Matthew N. O. Sadiku | Janet O. Sadiku "Sustainable Energy" Published in International Journal of Trend in Scientific Research and Development (ijtsrd), ISSN: 2456-6470, Volume-8 | Issue-1 , February 2024, URL: https://www.ijtsrd.com/papers/ijtsrd64534.pdf Paper Url: https://www.ijtsrd.com/engineering/electrical-engineering/64534/sustainable-energy/paul-a-adekunte
Concepts for Sudan Survey Act Implementations Executive Regulations and Stand...ijtsrd
This paper aims to outline the executive regulations, survey standards, and specifications required for the implementation of the Sudan Survey Act, and for regulating and organizing all surveying work activities in Sudan. The act has been discussed for more than 5 years. The Land Survey Act was initiated by the Sudan Survey Authority and all official legislations were headed by the Sudan Ministry of Justice till it was issued in 2022. The paper presents conceptual guidelines to be used for the Survey Act implementation and to regulate the survey work practice, standardizing the field surveys, processing, quality control, procedures, and the processes related to survey work carried out by the stakeholders and relevant authorities in Sudan. The conceptual guidelines are meant to improve the quality and harmonization of geospatial data and to aid decision making processes as well as geospatial information systems. The established comprehensive executive regulations will govern and regulate the implementation of the Sudan Survey Geomatics Act in all surveying and mapping practices undertaken by the Sudan Survey Authority SSA and state local survey departments for public or private sector organizations. The targeted standards and specifications include the reference frame, projection, coordinate systems, and the guidelines and specifications that must be followed in the field of survey work, processes, and mapping products. In the last few decades, there has been a growing awareness of the importance of geomatics activities and measurements on the Earths surface in space and time, together with observing and mapping the changes. In such cases, data must be captured promptly, standardized, and obtained with more accuracy and specified in much detail. The paper will also highlight the current situation in Sudan, the degree to which survey standards are used, the problems encountered, and the errors that arise from not using the standards and survey specifications. Kamal A. A. Sami "Concepts for Sudan Survey Act Implementations - Executive Regulations and Standards" Published in International Journal of Trend in Scientific Research and Development (ijtsrd), ISSN: 2456-6470, Volume-8 | Issue-1 , February 2024, URL: https://www.ijtsrd.com/papers/ijtsrd63484.pdf Paper Url: https://www.ijtsrd.com/engineering/civil-engineering/63484/concepts-for-sudan-survey-act-implementations--executive-regulations-and-standards/kamal-a-a-sami
Towards the Implementation of the Sudan Interpolated Geoid Model Khartoum Sta...ijtsrd
The discussions between ellipsoid and geoid have invoked many researchers during the recent decades, especially during the GNSS technology era, which had witnessed a great deal of development but still geoid undulation requires more investigations. To figure out a solution for Sudans local geoid, this research has tried to intake the possibility of determining the geoid model by following two approaches, gravimetric and geometrical geoid model determination, by making use of GNSS leveling benchmarks at Khartoum state. The Benchmarks are well distributed in the study area, in which, the horizontal coordinates and the height above the ellipsoid have been observed by GNSS while orthometric heights were carried out using precise leveling. The Global Geopotential Model GGM represented in EGM2008 has been exploited to figure out the geoid undulation at the benchmarks in the study area. This is followed by a fitting process, that has been done to suit the geoid undulation data which has been computed using GNSS leveling data and geoid undulation inspired by the EGM2008. Two geoid surfaces were created after the fitting process to ensure that they are identical and both of them could be counted for getting the same geoid undulation with an acceptable accuracy. In this respect, statistical operation played an important role in ensuring the consistency and integrity of the model by applying cross validation techniques splitting the data into training and testing datasets for building the geoid model and testing its eligibility. The geometrical solution for geoid undulation computation has been utilized by applying straightforward equations that facilitate the calculation of the geoid undulation directly through applying statistical techniques for the GNSS leveling data of the study area to get the common equation parameters values that could be utilized to calculate geoid undulation of any position in the study area within the claimed accuracy. Both systems were checked and proved eligible to be used within the study area with acceptable accuracy which may contribute to solving the geoid undulation problem in the Khartoum area, and be further generalized to determine the geoid model over the entire country, and this could be considered in the future, for regional and continental geoid model. Ahmed M. A. Mohammed. | Kamal A. A. Sami "Towards the Implementation of the Sudan Interpolated Geoid Model (Khartoum State Case Study)" Published in International Journal of Trend in Scientific Research and Development (ijtsrd), ISSN: 2456-6470, Volume-8 | Issue-1 , February 2024, URL: https://www.ijtsrd.com/papers/ijtsrd63483.pdf Paper Url: https://www.ijtsrd.com/engineering/civil-engineering/63483/towards-the-implementation-of-the-sudan-interpolated-geoid-model-khartoum-state-case-study/ahmed-m-a-mohammed
Activating Geospatial Information for Sudans Sustainable Investment Mapijtsrd
Sudan is witnessing an acceleration in the processes of development and transformation in the performance of government institutions to raise the productivity and investment efficiency of the government sector. The development plans and investment opportunities have focused on achieving national goals in various sectors. This paper aims to illuminate the path to the future and provide geospatial data and information to develop the investment climate and environment for all sized businesses, and to bridge the development gap between the Sudan states. The Sudan Survey Authority SSA is the main advisor to the Sudan Government in conducting surveying, mappings, designing, and developing systems related to geospatial data and information. In recent years, SSA made a strategic partnership with the Ministry of Investment to activate Geospatial Information for Sudans Sustainable Investment and in particular, for the preparation and implementation of the Sudan investment map, based on the directives and objectives of the Ministry of Investment MI in Sudan. This paper comes within the framework of activating the efforts of the Ministry of Investment to develop technical investment services by applying techniques adopted by the Ministry and its strategic partners for advancing investment processes in the country. Kamal A. A. Sami "Activating Geospatial Information for Sudan's Sustainable Investment Map" Published in International Journal of Trend in Scientific Research and Development (ijtsrd), ISSN: 2456-6470, Volume-8 | Issue-1 , February 2024, URL: https://www.ijtsrd.com/papers/ijtsrd63482.pdf Paper Url: https://www.ijtsrd.com/engineering/information-technology/63482/activating-geospatial-information-for-sudans-sustainable-investment-map/kamal-a-a-sami
Educational Unity Embracing Diversity for a Stronger Societyijtsrd
In a rapidly changing global landscape, the importance of education as a unifying force cannot be overstated. This paper explores the crucial role of educational unity in fostering a stronger and more inclusive society through the embrace of diversity. By examining the benefits of diverse learning environments, the paper aims to highlight the positive impact on societal strength. The discussion encompasses various dimensions, from curriculum design to classroom dynamics, and emphasizes the need for educational institutions to become catalysts for unity in diversity. It highlights the need for a paradigm shift in educational policies, curricula, and pedagogical approaches to ensure that they are reflective of the diverse fabric of society. This paper also addresses the challenges associated with implementing inclusive educational practices and offers practical strategies for overcoming barriers. It advocates for collaborative efforts between educational institutions, policymakers, and communities to create a supportive ecosystem that promotes diversity and unity. Mr. Amit Adhikari | Madhumita Teli | Gopal Adhikari "Educational Unity: Embracing Diversity for a Stronger Society" Published in International Journal of Trend in Scientific Research and Development (ijtsrd), ISSN: 2456-6470, Volume-8 | Issue-1 , February 2024, URL: https://www.ijtsrd.com/papers/ijtsrd64525.pdf Paper Url: https://www.ijtsrd.com/humanities-and-the-arts/education/64525/educational-unity-embracing-diversity-for-a-stronger-society/mr-amit-adhikari
Integration of Indian Indigenous Knowledge System in Management Prospects and...ijtsrd
The diversity of indigenous knowledge systems in India is vast and can vary significantly between different communities and regions. Preserving and respecting these knowledge systems is crucial for maintaining cultural heritage, promoting sustainable practices, and fostering cross cultural understanding. In this paper, an overview of the prospects and challenges associated with incorporating Indian indigenous knowledge into management is explored. It is found that IIKS helps in management in many areas like sustainable development, tourism, food security, natural resource management, cultural preservation and innovation, etc. However, IIKS integration with management faces some challenges in the form of a lack of documentation, cultural sensitivity, language barriers legal framework, etc. Savita Lathwal "Integration of Indian Indigenous Knowledge System in Management: Prospects and Challenges" Published in International Journal of Trend in Scientific Research and Development (ijtsrd), ISSN: 2456-6470, Volume-8 | Issue-1 , February 2024, URL: https://www.ijtsrd.com/papers/ijtsrd63500.pdf Paper Url: https://www.ijtsrd.com/management/accounting-and-finance/63500/integration-of-indian-indigenous-knowledge-system-in-management-prospects-and-challenges/savita-lathwal
DeepMask Transforming Face Mask Identification for Better Pandemic Control in...ijtsrd
The COVID 19 pandemic has highlighted the crucial need of preventive measures, with widespread use of face masks being a key method for slowing the viruss spread. This research investigates face mask identification using deep learning as a technological solution to be reducing the risk of coronavirus transmission. The proposed method uses state of the art convolutional neural networks CNNs and transfer learning to automatically recognize persons who are not wearing masks in a variety of circumstances. We discuss how this strategy improves public health and safety by providing an efficient manner of enforcing mask wearing standards. The report also discusses the obstacles, ethical concerns, and prospective applications of face mask detection systems in the ongoing fight against the pandemic. Dilip Kumar Sharma | Aaditya Yadav "DeepMask: Transforming Face Mask Identification for Better Pandemic Control in the COVID-19 Era" Published in International Journal of Trend in Scientific Research and Development (ijtsrd), ISSN: 2456-6470, Volume-8 | Issue-1 , February 2024, URL: https://www.ijtsrd.com/papers/ijtsrd64522.pdf Paper Url: https://www.ijtsrd.com/engineering/electronics-and-communication-engineering/64522/deepmask-transforming-face-mask-identification-for-better-pandemic-control-in-the-covid19-era/dilip-kumar-sharma
Streamlining Data Collection eCRF Design and Machine Learningijtsrd
Efficient and accurate data collection is paramount in clinical trials, and the design of Electronic Case Report Forms eCRFs plays a pivotal role in streamlining this process. This paper explores the integration of machine learning techniques in the design and implementation of eCRFs to enhance data collection efficiency. We delve into the synergies between eCRF design principles and machine learning algorithms, aiming to optimize data quality, reduce errors, and expedite the overall data collection process. The application of machine learning in eCRF design brings forth innovative approaches to data validation, anomaly detection, and real time adaptability. This paper discusses the benefits, challenges, and future prospects of leveraging machine learning in eCRF design for streamlined and advanced data collection in clinical trials. Dhanalakshmi D | Vijaya Lakshmi Kannareddy "Streamlining Data Collection: eCRF Design and Machine Learning" Published in International Journal of Trend in Scientific Research and Development (ijtsrd), ISSN: 2456-6470, Volume-8 | Issue-1 , February 2024, URL: https://www.ijtsrd.com/papers/ijtsrd63515.pdf Paper Url: https://www.ijtsrd.com/biological-science/biotechnology/63515/streamlining-data-collection-ecrf-design-and-machine-learning/dhanalakshmi-d
Macroeconomics- Movie Location
This will be used as part of your Personal Professional Portfolio once graded.
Objective:
Prepare a presentation or a paper using research, basic comparative analysis, data organization and application of economic information. You will make an informed assessment of an economic climate outside of the United States to accomplish an entertainment industry objective.
Unit 8 - Information and Communication Technology (Paper I).pdfThiyagu K
This slides describes the basic concepts of ICT, basics of Email, Emerging Technology and Digital Initiatives in Education. This presentations aligns with the UGC Paper I syllabus.
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!
Introduction to AI for Nonprofits with Tapp NetworkTechSoup
Dive into the world of AI! Experts Jon Hill and Tareq Monaur will guide you through AI's role in enhancing nonprofit websites and basic marketing strategies, making it easy to understand and apply.
This presentation includes basic of PCOS their pathology and treatment and also Ayurveda correlation of PCOS and Ayurvedic line of treatment mentioned in classics.
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.
2024.06.01 Introducing a competency framework for languag learning materials ...Sandy Millin
http://sandymillin.wordpress.com/iateflwebinar2024
Published classroom materials form the basis of syllabuses, drive teacher professional development, and have a potentially huge influence on learners, teachers and education systems. All teachers also create their own materials, whether a few sentences on a blackboard, a highly-structured fully-realised online course, or anything in between. Despite this, the knowledge and skills needed to create effective language learning materials are rarely part of teacher training, and are mostly learnt by trial and error.
Knowledge and skills frameworks, generally called competency frameworks, for ELT teachers, trainers and managers have existed for a few years now. However, until I created one for my MA dissertation, there wasn’t one drawing together what we need to know and do to be able to effectively produce language learning materials.
This webinar will introduce you to my framework, highlighting the key competencies I identified from my research. It will also show how anybody involved in language teaching (any language, not just English!), teacher training, managing schools or developing language learning materials can benefit from using the framework.
A review of the growth of the Israel Genealogy Research Association Database Collection for the last 12 months. Our collection is now passed the 3 million mark and still growing. See which archives have contributed the most. See the different types of records we have, and which years have had records added. You can also see what we have for the future.
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important and usefulinformationfor identifyingspeakers.In
thecurrent speech recognition systems, it is very rarely used
as it cannot be reliably extracted, and is notalwayspresentin
thespeech signal. An attempt is made to utilize this pitch and
voicing information for speaker identification.RanD.Zilca[2]
studied Text-Independent.F.K. Soong, A. E.RosenbergandB.
H. Juang [3] were described as a vector quantization
approach to speaker recognition. A. E. Rosenberg and F. K.
Soong [4] expressed recent research in automatic speaker
recognition.
Text-Independent Speaker Verification was described by
Gintaras Barisevicius.In thissystem,theapplicationsfortext-
independent verification system are vast: starting with
telephone service and endingup withhandlingbankaccount.
SPEAKER IDENTIFICATION STEPS
The most important partsofaspeakerrecognitionsystemare
the feature extraction and classification methods. The aim of
the feature extraction step is to strip unnecessary
information from the sensor data and convert the properties
of thesignal, which are important for the pattern recognition
task to a format that simples the distinction of the classes.
Usually, the featureextractionprocessreducesthedimension
of the data in order to avoid the curse of dimensionality. The
goal of the classification step is to estimate the general
extension of the classes within feature space from the
training set.
A. SPEECH PARAMETERIZATION (FEATURE
EXTRACTION)
Feature extraction: Before identifying any voices or training
person to be identified by the system, the voice signal must
be processed to extractimportant speech characteristics,the
amount of data used for comparisons is greatly reduced and
thus, less computation and less time is needed for
comparisons. The steps used in feature extraction are frame
blocking, windowing, Fast Fourier transform,Mel-frequency
wrapping and cepstrum Figure 1. Frame blocking and
windowing: In this step, the signal is put into frames (each
256 samples long). This corresponds to about sound per
frame. Each frame is then put through a hamming window.
Windowing is done to avoid problems due to truncation of
the signal. The hamming window has the form:
1Nn0,
1N
n2
cos46.054.0)n(w
(1)
Wher N=256 is the length of the frame.
The system of home meter reading is composed of a control
terminal in distance, GPRSmoduleandusermeteringmodule
is shown in Figure 2.
1
0
/2
1,...,2,1,0,
N
k
Njkn
kn NnexX
(2)
Where, j is the imaginary unit, , i.e. j = 1 .
Mel-frequency wrapping: In the fourth step, psychophysical
studies have shown that human perception of the frequency
contents of sounds for speech signals does notfollow alinear
scale. Thus for each tone with an actual frequency, f,
measured in Hz, a subjective pitch is measured on a scale
called the ‘Mel’ scale. The meal-frequency scale is linear
frequency spacing below 1000 Hz and a logarithmic spacing
above1000 Hz. As a reference point, the pitchofa1kHztone,
40 dB above the perceptual hearing threshold, is defined as
1000 meals. Therefore it can be used the following
approximate formula to compute the models for a given
frequency, f, in Hz:
Mel( f ) = 2595*log10(1+ f / 700) (3)
One approach to simulating the subjective spectrum is to use
a filter bank, one filter for each desired Mel-frequency
component. That filter bank has a triangular bandpass
frequency response, and the spacing, as well as the
bandwidth, is determined by a constant meal-frequency
interval. The modified spectrum of S(w) thus consists of the
output power of these filters when S(w) is the input. The
number of melspectrum coefficients, K, is typicallychosen as
20. This filter bank is applied in the frequency domain,
therefore it simply amounts to taking those triangle-shaped
windows on the spectrum.Ausefulwayofthinkingaboutthis
mel-wrappingfilter bank is to view each filter as a histogram
bin (where bins have overlap) in the frequency domain.
Cepstrum: In this final step, it converts the log mel spectrum
back to time. The result is called the Mel frequency cepstrum
coefficients (MFCC). The cepstral representation of the
speech spectrum provides a good representation of the local
spectral properties of the signal for the given frame analysis.
Because the mel spectrum coefficients (and so their
logarithm) are real numbers, it can be converted to the time
domain using the Discrete Cosine Transform (DCT).
Thereforeif it denotesthosemelpowerspectrumcoefficients
that are the result of the last step are ,,..,2,1,
~
KkSk the
MFCC’s, nc~
, can be calculated as follow:
k
knSC kn
2
1
cos)
~
(log
~
, (4)
n=1,2,…,K
Note that it excludes the first component, 0c~
, from the DCT
since it represents the mean value of the input signal which
carried little speaker-specific information.
PATTERN MATCHING AND CLASSIFICATION
Speaker identification is basically a pattern classification
problem preceded by a feature extraction stage [5]. Given a
sequence of feature vectors representing the given test
utterance, it is the job of the classifier to find out which
speaker has produced this utterance[6]. In ordertocarryout
this task, the acoustic models are constructed for each of the
speakers from its training data.
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In the classification stage, the sequenceoffeaturevectors
representing the test utterance is compared with each
acoustic model to produce a similarity measure that relates
the test utterance with each speaker. Using this measure, the
speaker identification system recognizes the identity of the
speaker.
There exists a lot of modelforclassification:Templatemodels
and statistical models. TemplatesmodelsusedDynamicTime
Wrapping (DTW) and Vector Quantization (VQ) models.
Statistical models include a Gaussian Mixture Model (GMM),
Hidden MarkovModels (HMM) andArtificialNeuralNetwork
(ANN) [7,8]. Vector quantization methods wereoutlinedand
an example of such classification was displayed.
1. Dynamic Time Wrapping (DTW)
In this paper, Vector Quantization (VQ) method was used
for classification so DynamicTime Wrapping (DTW) method
didn’t discuss in detail. The main idea of this approachis that
training template T consistingofNT framesandtestutterance
R consisting of NR frames, the Dynamic Time Wrapping
model is able to find the function m=w(n) , which maps the
time axis n of T to time axis m of R.
Thus, the system makes the comparisonbetweenthetestand
training data of the speaker evaluating the distance between
them and makes the decision whether in favor of the user
identify or not identify [8].
2. Vector Quantization (VQ)
Vector Quantization (VQ) method could be to use all the
feature vectors of a given speaker occurring in the training
data to form this speaker’s model. However, this is not
practical as thereare too manyfeaturevectorsinthetraining
data for each speaker. Therefore, a method of reducing the
number of training vectors is required.
It is possible to reduce the training data by using a Vector
Quantization (VQ) codebook consisting of a small number of
highly representative vectors that efficiently represent the
speaker-specific characteristics [2,9].NotethattheVQ-based
classifiers were popular in earlier days for text-dependent
speaker recognition.
There is a well-known algorithm,namely,Linde,Buzoand
Gray(LBG) algorithm, for clusteringasetofLtrainingvectors
onto a set of M codebook vectors. Each feature vector is the
sequence X is compared with all the stored codeword in the
codebook andthecodewordwiththeminimumdistancefrom
the feature vectors is selected as a proposed command.
For each codebook, adistancemeasureiscomputed,andthe
command with the lowest distance is chosen.
d(x, y) = (5)
The search of the nearest vector is done exhaustively, by
finding the distance between the input vector X and each of
the codewords C1-CM from the codebook. The one with the
smallest distance is coded as the output command.
PROPOSED SYSTEM DESIGN
Figure 2 shows the basic structure of the speaker
identification system. It is two-phase in this system. Thefirst
phase is the training phase. In this phase, the voices are
recorded.
Figure2. The basic structure of the speaker
identification system
The recorded voices are then extracted. The features
extracted from the recorded voices are used to develop
models of the languages. The second phase in the system is
testing. In this phase, the entered voice is processed and
compared with the languages model to identifythelanguage.
In the training part, there are Pre-processing, Feature
Extraction and Speaker Modeling. To get the feature vectors
incomingvoice,pre-processingwillbeperformed.Forfeature
extraction, Mel-frequency Cepstral Coefficient (MFCC)
algorithm is used. For codebook construction, Vector
Quantization (VQ) algorithm is used. After extracting the
features, speaker models are built and then save to the
database. In the testing part, features of speech signals are
extracted and matched it to the speaker model in the
database. And then make a decision based on the minimum
distance between the input pattern and speaker model. To
decide, a threshold is used with each speaker. If the speaker
minimum distance if lower than this threshold, then the
language is identified. However, if the speaker minimum
distanceexceeds thisthreshold,thelanguageisnotidentified.
Figure 3 shows the flow chart of the testing process.
Figure3. Flowchart of the speech recognition system
for the testing phase
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TEST AND RESULTS
Firstly, the speech signal is recorded and saved as wave files for the person.
In this step, voice signal is saved as wave files in the traineddataset to match the input voice files from theuser. So, the user can
see the wave files which are saved in the train data set by clicking any language push button. If the user chooses the Myanmar
language button, the feature database display appears as shown in Figure 4.
Figure4. Feature database display of Myanmar language
Table 1 shows the parameters of code minimum and maximum values for each language. Thedataarecalculated as
average for each type of languages. The recording duration also depends on the test person ascend. The threshold value for
distance vector is defined by checking and testing many times according to the recording database.
TABLE I COMPARISON OF PARAMETERS
Language Type
Min Codebook
value
Max Codebook
value
Frequency
Average
Duration
Euclidean distance
(threshold)
English language -14.5934 13.2986 80-130 Hz 4 sec 2
Myanmar language -29.2414 7.8993 85-150 Hz 5 sec 2
Chinese language -18.1614 12.5227 100–190 Hz 4.5 sec 2
Recognition is to test the unknown speaker; the steps are open speech file, recognize the language, play wave
file and main as shown in Figure 5.
Figure5. Recognition dialog box
The open speech file button shows the wav files that are saved. If the user selects a wav file want to recognize, the
speech file is loaded as shown in Figure 6 and 7.
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Figure6. Open voice dialog box
Figure7. Loaded speech file dialog box
The recognize language button identifies the result as a text of language such as Myanmar, English and Chineseas
shown in Figure 8. The result was identified as a voice if the user has to click the play wav file button. The main button is going
back to Main Window of the Language Speech Recognition System based on Feature Extraction and Classification.
Figure8. Result screen of recognized languages as a text
The accuracy of the system is calculated with 25 train files for each language per person. Fivepersons arerecordedtoget
precise accuracy in each language. So, totally 125 files are recorded for each language. For testing, the untrained files are also
checked 4 files for each language per person. Different persons are also tested and the result is shown in table II. The known
trained persons haveachieved 100%accuracy. But the untrained persons have gotlow accuracydueto the ascentoftheirvoice
and the level of voice.
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The Chinese language is most difficult to recognize for untrained persons. The style of speaking and the wood frames are
different from other languages. The accuracy can also change according to the test person is whether the native speaker or not
TABLE II IDENTIFICATION ACCURACY OF THE SYSTEM
LANGUAGE TYPE No of train file No of the Test file % Correct for train file % Correct for Test file
English language 125 100 100 60
Myanmar language 125 100 100 54
Chinese language 125 100 100 40
CONCLUSION
The goal of this paper was to create a speaker
recognition system and apply it to the speech of an
unknown speaker. By investigating the extracted features of
the unknown speech and then compare them to the stored
extracted features for each different speaker in order to
identify the unknown speaker. To evaluate the performance
of theproposed system, the systemtrainedthe125wavefiles
of theMyanmar,EnglishandChineselanguageandtested100
wave file by untrained persons. The result is 100% accuracy
in the trained speaker verification system by using MFCC
(Mel Frequency Cepstral Coefficients). But the untrained
speaker could not get the good accuracy, it got around 50 %
accuracy depends on the speaker ascend. The function
‘melcepst’ is used to calculate the mel cepstrum of a signal.
The speaker was modeled using Vector Quantization (VQ). A
VQ codebook is generated by clustering the training feature
vectors of each speaker and then stored in the speaker
database. In this method, Linde, Buzo and Gray (LBG)
algorithm, for clustering a set of L training vectors onto a set
of M codebook vectors. In the recognition stage, a distortion
measure which based on the minimizing the distance was
used when matching an unknown speaker with the speaker
database. During this paper, we have found out that the VQ
based clustering approach provides us with the faster
speaker identification process.
ACKNOWLEDGMENT
The author would like to thank all teachers, from the
Department of Information Technology, Technological
(Taunggyi), who gives suggestions and advice for her
submission of the paper.
REFERENCES
[1] Markowitz, J. A and colleagues: “J. Markowitz,
Consultants”, (2003).
[2] Ran Zilca, D: “Text-Independent Speaker Verification
Using Utterance Level Scoring and Covariance
Modeling”, (2002).
[3] F. K. Soong, A. E. Rosenberg, and B.H. Juang, “A vector
quantization approach to speaker recognition,” AT & T
Journal, vol, 66, no.2, pp. 14-26, 1987.
[4] A. E Rosenberg, and F. K. Soong, “Recent research in
automatic speaker recognition,” in Advances in Speech
Signal Processing, S. Furui, M. Sondhi, Eds. New York:
Marcel Dekker Inc., pp. 701-737, 1992.
[5] L. R. Rabiner and R. W. Schafer, Digital Processing of
Speech Signals, New Jersey; Prentice-Hall, pp. 141-
161.pp. 314-322, pp. 476-485, 1978.