Identification of Sex of the Speaker With Reference To Bodo Vowels: A Compara...IJERA Editor
This work presents an application of Fundamental Frequency (Pitch), Linear Predictive Cepstral Coefficient
(LPCC) and Mel Frequency Cepstral Coefficient (MFCC) in identification of sex of the speaker in speech
recognition research. The aim of this article is to compare the performance of these three methods for
identification of sex of the speakers. A successful speech recognition system can help in non critical operations
such as presenting the driving route to the driver, dialing a phone number, light switch turn on/off, the coffee
machine on/off etc. apart from speaker verification-caste wise, community wise and locality wise including
identification of sex. Here an attempt has been made to identify the sex of Bodo speakers through vowel
utterance by following Pitch value, LPCC and MFCC techniques. It is found here that the feature vector
organization of LPCC coefficients provides a more promising way of speech-speaker recognition in case of
Bodo Language than that of Pitch and MFCC.
This document reviews techniques used in spoken-word recognition systems. It discusses popular feature extraction techniques like MFCC, LPC, DWT, WPD that are used to represent speech signals in a compact form before classification. Classification techniques discussed are ANN, HMM, DTW, and VQ. The document provides a brief overview of each technique and their advantages. It also presents the generalized workflow of a spoken-word recognition system including stages of speech acquisition, pre-emphasis, feature extraction, modeling, classification, and output of recognized text.
LPC Models and Different Speech Enhancement Techniques- A Reviewijiert bestjournal
Author has already published one review paper on the quality enhancement of a speech signal by minimizing the noise. This is a second paper of same series. In last two decades the researchers have taken continuous efforts to reduce the noise signal from the speech signal. Th is paper comments on,various study carried out and analysis propos als of the researchers for en hancement of the quality of speech signal. Various models,coding,speech quality improvement methods,speaker dependent codebooks,autocorrelation subtraction,speech restoration,producing speech at low bit rates,compression and enhancement are the vari ous aspects of speech enhancement. We have presented the review of all above mentioned technologies in this paper and also willing to examine few of the techniques in order to analyze the factors affecting them in upcoming paper of the series.
ppt-Piezoelectric Throat Microphone Based Voice Analysis.pptxlanimathew1
This document discusses the use of a piezoelectric throat microphone for voice analysis and speech recognition. It describes how throat microphones capture vocal fold vibrations directly using piezoelectric sensors. The document outlines a study where speech samples were collected from participants using a throat microphone and analyzed with Praat software to investigate the potential for analyzing voice disorders. Analysis of fundamental frequency estimation showed that throat microphones can provide accurate results due to directly picking up laryngeal vibrations.
FORMANT ANALYSIS OF BANGLA VOWEL FOR AUTOMATIC SPEECH RECOGNITIONsipij
To provide new technological benefits to the mass people, nowadays, regional and local language
recognition draws attention to the researchers. Similarly to other languages, Bangla speech recognition
scheme is demandable. A formant is considered as the resonance frequency of vocal tract. Formant
frequencies play an important role for the purpose of automatic speech recognition, due to its noise robust
characteristics. In this paper, Bangla vowels are investigated to acquire formant frequencies and its
corresponding bandwidth from continuous Bangla sentences, which are considered as potential parameters
for wide voice applications. For the purpose of formant analysis, cepstrum based formant estimation and
Linear Predictive Coding (LPC) techniques are used. In order to acquire formant characteristics, enrich
continuous sentences and widely available Bangla language corpus namely “SHRUTI” is considered.
Intensive experimentation is carried out to determine formant characteristics (frequency and bandwidth) of
Bangla vowels for both male and female speakers. Finally, vowel recognition accuracy of Bangla language
is reported considering first three formants
Effect of MFCC Based Features for Speech Signal Alignmentskevig
The fundamental techniques used for man-machine communication include Speech synthesis, speech
recognition, and speech transformation. Feature extraction techniques provide a compressed
representation of the speech signals. The HNM analyses and synthesis provides high quality speech with
less number of parameters. Dynamic time warping is well known technique used for aligning two given
multidimensional sequences. It locates an optimal match between the given sequences. The improvement in
the alignment is estimated from the corresponding distances. The objective of this research is to investigate
the effect of dynamic time warping on phrases, words, and phonemes based alignments. The speech signals
in the form of twenty five phrases were recorded. The recorded material was segmented manually and
aligned at sentence, word, and phoneme level. The Mahalanobis distance (MD) was computed between the
aligned frames. The investigation has shown better alignment in case of HNM parametric domain. It has
been seen that effective speech alignment can be carried out even at phrase level
EFFECT OF MFCC BASED FEATURES FOR SPEECH SIGNAL ALIGNMENTSijnlc
The fundamental techniques used for man-machine communication include Speech synthesis, speech
recognition, and speech transformation. Feature extraction techniques provide a compressed
representation of the speech signals. The HNM analyses and synthesis provides high quality speech with
less number of parameters. Dynamic time warping is well known technique used for aligning two given
multidimensional sequences. It locates an optimal match between the given sequences. The improvement in
the alignment is estimated from the corresponding distances. The objective of this research is to investigate
the effect of dynamic time warping on phrases, words, and phonemes based alignments. The speech signals
in the form of twenty five phrases were recorded. The recorded material was segmented manually and
aligned at sentence, word, and phoneme level. The Mahalanobis distance (MD) was computed between the
aligned frames. The investigation has shown better alignment in case of HNM parametric domain. It has
been seen that effective speech alignment can be carried out even at phrase level.
DATABASES, FEATURES, CLASSIFIERS AND CHALLENGES IN AUTOMATIC SPEECH RECOGNITI...IRJET Journal
This document provides a review of speech recognition technologies including databases, features, classifiers, and challenges. It summarizes 13 speech corpora in terms of languages covered, recording lengths, development status, and accessibility. It also outlines important feature extraction techniques for speech recognition like MFCCs, formants, fundamental frequency, and LPCCs. Classifiers discussed include HMM, GMM, DNN, and fusion techniques. Challenges covered are developing robust systems that can handle variability across speakers, environments, and languages.
Identification of Sex of the Speaker With Reference To Bodo Vowels: A Compara...IJERA Editor
This work presents an application of Fundamental Frequency (Pitch), Linear Predictive Cepstral Coefficient
(LPCC) and Mel Frequency Cepstral Coefficient (MFCC) in identification of sex of the speaker in speech
recognition research. The aim of this article is to compare the performance of these three methods for
identification of sex of the speakers. A successful speech recognition system can help in non critical operations
such as presenting the driving route to the driver, dialing a phone number, light switch turn on/off, the coffee
machine on/off etc. apart from speaker verification-caste wise, community wise and locality wise including
identification of sex. Here an attempt has been made to identify the sex of Bodo speakers through vowel
utterance by following Pitch value, LPCC and MFCC techniques. It is found here that the feature vector
organization of LPCC coefficients provides a more promising way of speech-speaker recognition in case of
Bodo Language than that of Pitch and MFCC.
This document reviews techniques used in spoken-word recognition systems. It discusses popular feature extraction techniques like MFCC, LPC, DWT, WPD that are used to represent speech signals in a compact form before classification. Classification techniques discussed are ANN, HMM, DTW, and VQ. The document provides a brief overview of each technique and their advantages. It also presents the generalized workflow of a spoken-word recognition system including stages of speech acquisition, pre-emphasis, feature extraction, modeling, classification, and output of recognized text.
LPC Models and Different Speech Enhancement Techniques- A Reviewijiert bestjournal
Author has already published one review paper on the quality enhancement of a speech signal by minimizing the noise. This is a second paper of same series. In last two decades the researchers have taken continuous efforts to reduce the noise signal from the speech signal. Th is paper comments on,various study carried out and analysis propos als of the researchers for en hancement of the quality of speech signal. Various models,coding,speech quality improvement methods,speaker dependent codebooks,autocorrelation subtraction,speech restoration,producing speech at low bit rates,compression and enhancement are the vari ous aspects of speech enhancement. We have presented the review of all above mentioned technologies in this paper and also willing to examine few of the techniques in order to analyze the factors affecting them in upcoming paper of the series.
ppt-Piezoelectric Throat Microphone Based Voice Analysis.pptxlanimathew1
This document discusses the use of a piezoelectric throat microphone for voice analysis and speech recognition. It describes how throat microphones capture vocal fold vibrations directly using piezoelectric sensors. The document outlines a study where speech samples were collected from participants using a throat microphone and analyzed with Praat software to investigate the potential for analyzing voice disorders. Analysis of fundamental frequency estimation showed that throat microphones can provide accurate results due to directly picking up laryngeal vibrations.
FORMANT ANALYSIS OF BANGLA VOWEL FOR AUTOMATIC SPEECH RECOGNITIONsipij
To provide new technological benefits to the mass people, nowadays, regional and local language
recognition draws attention to the researchers. Similarly to other languages, Bangla speech recognition
scheme is demandable. A formant is considered as the resonance frequency of vocal tract. Formant
frequencies play an important role for the purpose of automatic speech recognition, due to its noise robust
characteristics. In this paper, Bangla vowels are investigated to acquire formant frequencies and its
corresponding bandwidth from continuous Bangla sentences, which are considered as potential parameters
for wide voice applications. For the purpose of formant analysis, cepstrum based formant estimation and
Linear Predictive Coding (LPC) techniques are used. In order to acquire formant characteristics, enrich
continuous sentences and widely available Bangla language corpus namely “SHRUTI” is considered.
Intensive experimentation is carried out to determine formant characteristics (frequency and bandwidth) of
Bangla vowels for both male and female speakers. Finally, vowel recognition accuracy of Bangla language
is reported considering first three formants
Effect of MFCC Based Features for Speech Signal Alignmentskevig
The fundamental techniques used for man-machine communication include Speech synthesis, speech
recognition, and speech transformation. Feature extraction techniques provide a compressed
representation of the speech signals. The HNM analyses and synthesis provides high quality speech with
less number of parameters. Dynamic time warping is well known technique used for aligning two given
multidimensional sequences. It locates an optimal match between the given sequences. The improvement in
the alignment is estimated from the corresponding distances. The objective of this research is to investigate
the effect of dynamic time warping on phrases, words, and phonemes based alignments. The speech signals
in the form of twenty five phrases were recorded. The recorded material was segmented manually and
aligned at sentence, word, and phoneme level. The Mahalanobis distance (MD) was computed between the
aligned frames. The investigation has shown better alignment in case of HNM parametric domain. It has
been seen that effective speech alignment can be carried out even at phrase level
EFFECT OF MFCC BASED FEATURES FOR SPEECH SIGNAL ALIGNMENTSijnlc
The fundamental techniques used for man-machine communication include Speech synthesis, speech
recognition, and speech transformation. Feature extraction techniques provide a compressed
representation of the speech signals. The HNM analyses and synthesis provides high quality speech with
less number of parameters. Dynamic time warping is well known technique used for aligning two given
multidimensional sequences. It locates an optimal match between the given sequences. The improvement in
the alignment is estimated from the corresponding distances. The objective of this research is to investigate
the effect of dynamic time warping on phrases, words, and phonemes based alignments. The speech signals
in the form of twenty five phrases were recorded. The recorded material was segmented manually and
aligned at sentence, word, and phoneme level. The Mahalanobis distance (MD) was computed between the
aligned frames. The investigation has shown better alignment in case of HNM parametric domain. It has
been seen that effective speech alignment can be carried out even at phrase level.
DATABASES, FEATURES, CLASSIFIERS AND CHALLENGES IN AUTOMATIC SPEECH RECOGNITI...IRJET Journal
This document provides a review of speech recognition technologies including databases, features, classifiers, and challenges. It summarizes 13 speech corpora in terms of languages covered, recording lengths, development status, and accessibility. It also outlines important feature extraction techniques for speech recognition like MFCCs, formants, fundamental frequency, and LPCCs. Classifiers discussed include HMM, GMM, DNN, and fusion techniques. Challenges covered are developing robust systems that can handle variability across speakers, environments, and languages.
Performance estimation based recurrent-convolutional encoder decoder for spee...karthik annam
This document discusses a proposed Recurrent-Convolutional Encoder-Decoder (R-CED) network for speech enhancement. The R-CED network aims to overcome challenges with existing methods by estimating the a priori and posteriori signal-to-noise ratios to separate noise from speech. The R-CED consists of convolutional layers with increasing and decreasing numbers of filters to encode and decode features. Performance will be evaluated using metrics like PESQ, STOI, CER, MSE, SNR, and SDR. The proposed method aims to improve speech enhancement accuracy and recover enhanced speech quality compared to other techniques.
The document outlines a methodology for developing an automatic speech recognition system, including extracting features from speech signals, building acoustic models using tools like hidden Markov models and Gaussian mixture models, and recognizing speech patterns to convert spoken words to text. It discusses challenges in applying speech recognition to under-resourced tribal languages in India and the need to preserve indigenous languages and cultures. The proposed research aims to implement and evaluate speech recognition techniques for tribal languages to help document and promote endangered languages.
Emotional telugu speech signals classification based on k nn classifiereSAT Publishing House
IJRET : International Journal of Research in Engineering and Technology is an international peer reviewed, online journal published by eSAT Publishing House for the enhancement of research in various disciplines of Engineering and Technology. The aim and scope of the journal is to provide an academic medium and an important reference for the advancement and dissemination of research results that support high-level learning, teaching and research in the fields of Engineering and Technology. We bring together Scientists, Academician, Field Engineers, Scholars and Students of related fields of Engineering and Technology.
Emotional telugu speech signals classification based on k nn classifiereSAT Journals
Abstract Speech processing is the study of speech signals, and the methods used to process them. In application such as speech coding, speech synthesis, speech recognition and speaker recognition technology, speech processing is employed. In speech classification, the computation of prosody effects from speech signals plays a major role. In emotional speech signals pitch and frequency is a most important parameters. Normally, the pitch value of sad and happy speech signals has a great difference and the frequency value of happy is higher than sad speech. But, in some cases the frequency of happy speech is nearly similar to sad speech or frequency of sad speech is similar to happy speech. In such situation, it is difficult to recognize the exact speech signal. To reduce such drawbacks, in this paper we propose a Telugu speech emotion classification system with three features like Energy Entropy, Short Time Energy, Zero Crossing Rate and K-NN classifier for the classification. Features are extracted from the speech signals and given to the K-NN. The implementation result shows the effectiveness of proposed speech emotion classification system in classifying the Telugu speech signals based on their prosody effects. The performance of the proposed speech emotion classification system is evaluated by conducting cross validation on the Telugu speech database. Keywords: Emotion Classification, K-NN classifier, Energy Entropy, Short Time Energy, Zero Crossing Rate.
Gender voice recognition stands for an imperative research field in acoustics and speech processing as human voice shows very remarkable aspects. This study investigates speech signals to devise a gender classifier by speech analysis to forecast the gender of the speaker by investigating diverse parameters of the voice sample. A database has 2270 voice samples of celebrities, both male and female. Through Mel frequency cepstrum coefficient (MFCC), vector quantization (VQ), and machine learning algorithm (J 48), an accuracy of about 100% is achieved by the proposed classification technique based on data mining and Java script.
This document provides an overview of speech recognition systems and recent progress in the field. It discusses different types of speech recognition including isolated word, connected word, continuous speech, and spontaneous speech. Various techniques used in speech recognition are also summarized, such as simulated evolutionary computation, artificial neural networks, fuzzy logic, Kalman filters, and Hidden Markov Models. The document reviews several papers published between 2004-2012 that studied speech recognition methods including using dynamic spectral subband centroids, Kalman filters, biomimetic computing techniques, noise estimation, and modulation filtering. It concludes that Hidden Markov Models combined with MFCC features provide good recognition results and are suitable for large vocabulary, speaker-independent, continuous speech recognition.
This document provides an overview of speech recognition systems and recent progress in the field. It discusses different types of speech recognition including isolated word, connected word, continuous speech, and spontaneous speech. Various techniques used in speech recognition are also summarized, such as simulated evolutionary computation, artificial neural networks, fuzzy logic, Kalman filters, and Hidden Markov Models. The document reviews several papers published between 2004-2012 that studied speech recognition methods including using dynamic spectral subband centroids, Kalman filters, biomimetic computing techniques, noise estimation, and modulation filtering. It concludes that Hidden Markov Models combined with MFCC features provide good recognition results for large vocabulary, speaker-independent, continuous speech recognition.
Speech emotion recognition with light gradient boosting decision trees machineIJECEIAES
Speech emotion recognition aims to identify the emotion expressed in the speech by analyzing the audio signals. In this work, data augmentation is first performed on the audio samples to increase the number of samples for better model learning. The audio samples are comprehensively encoded as the frequency and temporal domain features. In the classification, a light gradient boosting machine is leveraged. The hyperparameter tuning of the light gradient boosting machine is performed to determine the optimal hyperparameter settings. As the speech emotion recognition datasets are imbalanced, the class weights are regulated to be inversely proportional to the sample distribution where minority classes are assigned higher class weights. The experimental results demonstrate that the proposed method outshines the state-of-the-art methods with 84.91% accuracy on the Berlin database of emotional speech (emo-DB) dataset, 67.72% on the Ryerson audio-visual database of emotional speech and song (RAVDESS) dataset, and 62.94% on the interactive emotional dyadic motion capture (IEMOCAP) dataset.
IMPROVING MYANMAR AUTOMATIC SPEECH RECOGNITION WITH OPTIMIZATION OF CONVOLUTI...ijnlc
Researchers of many nations have developed automatic speech recognition (ASR) to show their national improvement in information and communication technology for their languages. This work intends to improve the ASR performance for Myanmar language by changing different Convolutional Neural Network (CNN) hyperparameters such as number of feature maps and pooling size. CNN has the abilities of reducing in spectral variations and modeling spectral correlations that exist in the signal due to the locality and pooling operation. Therefore, the impact of the hyperparameters on CNN accuracy in ASR tasks is investigated. A 42-hr-data set is used as training data and the ASR performance was evaluated on two open
test sets: web news and recorded data. As Myanmar language is a syllable-timed language, ASR based on syllable was built and compared with ASR based on word. As the result, it gained 16.7% word error rate (WER) and 11.5% syllable error rate (SER) on TestSet1. And it also achieved 21.83% WER and 15.76% SER on TestSet2.
IMPROVING MYANMAR AUTOMATIC SPEECH RECOGNITION WITH OPTIMIZATION OF CONVOLUTI...kevig
Researchers of many nations have developed automatic speech recognition (ASR) to show their national
improvement in information and communication technology for their languages. This work intends to
improve the ASR performance for Myanmar language by changing different Convolutional Neural Network
(CNN) hyperparameters such as number of feature maps and pooling size. CNN has the abilities of
reducing in spectral variations and modeling spectral correlations that exist in the signal due to the locality
and pooling operation. Therefore, the impact of the hyperparameters on CNN accuracy in ASR tasks is
investigated. A 42-hr-data set is used as training data and the ASR performance was evaluated on two open
test sets: web news and recorded data. As Myanmar language is a syllable-timed language, ASR based on
syllable was built and compared with ASR based on word. As the result, it gained 16.7% word error rate
(WER) and 11.5% syllable error rate (SER) on TestSet1. And it also achieved 21.83% WER and 15.76%
SER on TestSet2.
Utterance Based Speaker Identification Using ANNIJCSEA Journal
This document summarizes a research paper on speaker identification using artificial neural networks. The paper presents a speaker identification system that uses digital signal processing and ANN techniques. Speech features are extracted from utterances using FFT and windowing. These features are used to train a multi-layer perceptron network to classify speakers. The system was tested on Bangla speech and achieved accurate identification of speakers from their utterances.
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.
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.
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 recognition techniques are one of the most important modern technologies. Many different systems have been developed in terms of methods used in the extraction of features and methods of classification. Voice recognition includes two areas: speech recognition and speaker recognition, where the research is confined to the field of speech recognition. The research presents a proposal to improve the performance of single word recognition systems by an algorithm that combines more than one of the techniques used in character extraction and modulation of the neural network to study the effects of recognition science and study the effect of noise on the proposed system. In this research four systems of speech recognition were studied, the first system adopted the MFCC algorithm to extract the features. The second system adopted the PLP algorithm, while the third system was based on combining the two previous algorithms in addition to the zero-passing rate. In the fourth system, the neural network used in the differentiation process was modified and the error ratio was determined. The impact of noise on these previous systems. The outcomes were looked at regarding the rate of recognizable proof and the season of preparing the neural network for every system independently, to get a rate of distinguishing proof and quiet up to 98% utilizing the proposed framework.
SPEECH RECOGNITION BY IMPROVING THE PERFORMANCE OF ALGORITHMS USED IN DISCRIM...ijcsit
This document discusses improving speech recognition performance through algorithms used for feature extraction and classification. It examines 4 systems: 1) MFCC extraction, 2) PLP extraction, 3) combining MFCC, PLP and zero-crossing rate, 4) modifying the neural network. System 3 achieved the highest recognition rate of 98% even with noise, outperforming the individual algorithms. Increasing the training samples to 500 further improved recognition ratio.
Accents of English have been investigated for many years both from the perspective of native and non-native speakers of the language. Various research results imply that non-native speakers of English language produce certain speech characteristics which are uncommon in native speakers’ speech. This is because non-native speakers do not produce the same tongue movement as native speakers. This paper presents an isolated English word recognition system devised with the speech of local Bangladeshi people, who are also non-native speakers of English language. Here, we have also noticed a different speech characteristic which is not available within the speech of native English speakers. Two acoustic features, ‘pitch’ and ‘formants’ have been utilized to develop the system. The system is speaker-independent and stands on Template based approach. The recognition method applied here is very simple and the recognition accuracy is also very satisfactory.
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.
Spoken language identification using i-vectors, x-vectors, PLDA and logistic ...journalBEEI
This document discusses spoken language identification using i-vectors and x-vectors for feature extraction, and PLDA and logistic regression for classification. It examines extracting features from Javanese, Sundanese, and Minangkabau languages, then classifying the languages using various parameters. The study finds that x-vector outperforms i-vector when using PLDA classification, except when using logistic regression, where i-vector performs better. It tunes parameters for i-vector UBM size, i-vector dimension, x-vector max frame size, and num repeats, reporting equal error rates to evaluate performance on test segments of 3, 10 and 30 seconds.
Hindi digits recognition system on speech data collected in different natural...csandit
This paper presents a baseline digits speech recognizer for Hindi language. The recording environment is different for all speakers, since the data is collected in their respective homes. The different environment refers to vehicle horn noises in some road facing rooms, internal background noises in some rooms like opening doors, silence in some rooms etc. All these recordings are used for training acoustic model. The Acoustic Model is trained on 8 speakers’ audio data. The vocabulary size of the recognizer is 10 words. HTK toolkit is used for building
acoustic model and evaluating the recognition rate of the recognizer. The efficiency of the recognizer developed on recorded data, is shown at the end of the paper and possible directions for future research work are suggested.
This presentation was provided by Steph Pollock of The American Psychological Association’s Journals Program, and Damita Snow, of The American Society of Civil Engineers (ASCE), for the initial session of NISO's 2024 Training Series "DEIA in the Scholarly Landscape." Session One: 'Setting Expectations: a DEIA Primer,' was held June 6, 2024.
A workshop hosted by the South African Journal of Science aimed at postgraduate students and early career researchers with little or no experience in writing and publishing journal articles.
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This document discusses a proposed Recurrent-Convolutional Encoder-Decoder (R-CED) network for speech enhancement. The R-CED network aims to overcome challenges with existing methods by estimating the a priori and posteriori signal-to-noise ratios to separate noise from speech. The R-CED consists of convolutional layers with increasing and decreasing numbers of filters to encode and decode features. Performance will be evaluated using metrics like PESQ, STOI, CER, MSE, SNR, and SDR. The proposed method aims to improve speech enhancement accuracy and recover enhanced speech quality compared to other techniques.
The document outlines a methodology for developing an automatic speech recognition system, including extracting features from speech signals, building acoustic models using tools like hidden Markov models and Gaussian mixture models, and recognizing speech patterns to convert spoken words to text. It discusses challenges in applying speech recognition to under-resourced tribal languages in India and the need to preserve indigenous languages and cultures. The proposed research aims to implement and evaluate speech recognition techniques for tribal languages to help document and promote endangered languages.
Emotional telugu speech signals classification based on k nn classifiereSAT Publishing House
IJRET : International Journal of Research in Engineering and Technology is an international peer reviewed, online journal published by eSAT Publishing House for the enhancement of research in various disciplines of Engineering and Technology. The aim and scope of the journal is to provide an academic medium and an important reference for the advancement and dissemination of research results that support high-level learning, teaching and research in the fields of Engineering and Technology. We bring together Scientists, Academician, Field Engineers, Scholars and Students of related fields of Engineering and Technology.
Emotional telugu speech signals classification based on k nn classifiereSAT Journals
Abstract Speech processing is the study of speech signals, and the methods used to process them. In application such as speech coding, speech synthesis, speech recognition and speaker recognition technology, speech processing is employed. In speech classification, the computation of prosody effects from speech signals plays a major role. In emotional speech signals pitch and frequency is a most important parameters. Normally, the pitch value of sad and happy speech signals has a great difference and the frequency value of happy is higher than sad speech. But, in some cases the frequency of happy speech is nearly similar to sad speech or frequency of sad speech is similar to happy speech. In such situation, it is difficult to recognize the exact speech signal. To reduce such drawbacks, in this paper we propose a Telugu speech emotion classification system with three features like Energy Entropy, Short Time Energy, Zero Crossing Rate and K-NN classifier for the classification. Features are extracted from the speech signals and given to the K-NN. The implementation result shows the effectiveness of proposed speech emotion classification system in classifying the Telugu speech signals based on their prosody effects. The performance of the proposed speech emotion classification system is evaluated by conducting cross validation on the Telugu speech database. Keywords: Emotion Classification, K-NN classifier, Energy Entropy, Short Time Energy, Zero Crossing Rate.
Gender voice recognition stands for an imperative research field in acoustics and speech processing as human voice shows very remarkable aspects. This study investigates speech signals to devise a gender classifier by speech analysis to forecast the gender of the speaker by investigating diverse parameters of the voice sample. A database has 2270 voice samples of celebrities, both male and female. Through Mel frequency cepstrum coefficient (MFCC), vector quantization (VQ), and machine learning algorithm (J 48), an accuracy of about 100% is achieved by the proposed classification technique based on data mining and Java script.
This document provides an overview of speech recognition systems and recent progress in the field. It discusses different types of speech recognition including isolated word, connected word, continuous speech, and spontaneous speech. Various techniques used in speech recognition are also summarized, such as simulated evolutionary computation, artificial neural networks, fuzzy logic, Kalman filters, and Hidden Markov Models. The document reviews several papers published between 2004-2012 that studied speech recognition methods including using dynamic spectral subband centroids, Kalman filters, biomimetic computing techniques, noise estimation, and modulation filtering. It concludes that Hidden Markov Models combined with MFCC features provide good recognition results and are suitable for large vocabulary, speaker-independent, continuous speech recognition.
This document provides an overview of speech recognition systems and recent progress in the field. It discusses different types of speech recognition including isolated word, connected word, continuous speech, and spontaneous speech. Various techniques used in speech recognition are also summarized, such as simulated evolutionary computation, artificial neural networks, fuzzy logic, Kalman filters, and Hidden Markov Models. The document reviews several papers published between 2004-2012 that studied speech recognition methods including using dynamic spectral subband centroids, Kalman filters, biomimetic computing techniques, noise estimation, and modulation filtering. It concludes that Hidden Markov Models combined with MFCC features provide good recognition results for large vocabulary, speaker-independent, continuous speech recognition.
Speech emotion recognition with light gradient boosting decision trees machineIJECEIAES
Speech emotion recognition aims to identify the emotion expressed in the speech by analyzing the audio signals. In this work, data augmentation is first performed on the audio samples to increase the number of samples for better model learning. The audio samples are comprehensively encoded as the frequency and temporal domain features. In the classification, a light gradient boosting machine is leveraged. The hyperparameter tuning of the light gradient boosting machine is performed to determine the optimal hyperparameter settings. As the speech emotion recognition datasets are imbalanced, the class weights are regulated to be inversely proportional to the sample distribution where minority classes are assigned higher class weights. The experimental results demonstrate that the proposed method outshines the state-of-the-art methods with 84.91% accuracy on the Berlin database of emotional speech (emo-DB) dataset, 67.72% on the Ryerson audio-visual database of emotional speech and song (RAVDESS) dataset, and 62.94% on the interactive emotional dyadic motion capture (IEMOCAP) dataset.
IMPROVING MYANMAR AUTOMATIC SPEECH RECOGNITION WITH OPTIMIZATION OF CONVOLUTI...ijnlc
Researchers of many nations have developed automatic speech recognition (ASR) to show their national improvement in information and communication technology for their languages. This work intends to improve the ASR performance for Myanmar language by changing different Convolutional Neural Network (CNN) hyperparameters such as number of feature maps and pooling size. CNN has the abilities of reducing in spectral variations and modeling spectral correlations that exist in the signal due to the locality and pooling operation. Therefore, the impact of the hyperparameters on CNN accuracy in ASR tasks is investigated. A 42-hr-data set is used as training data and the ASR performance was evaluated on two open
test sets: web news and recorded data. As Myanmar language is a syllable-timed language, ASR based on syllable was built and compared with ASR based on word. As the result, it gained 16.7% word error rate (WER) and 11.5% syllable error rate (SER) on TestSet1. And it also achieved 21.83% WER and 15.76% SER on TestSet2.
IMPROVING MYANMAR AUTOMATIC SPEECH RECOGNITION WITH OPTIMIZATION OF CONVOLUTI...kevig
Researchers of many nations have developed automatic speech recognition (ASR) to show their national
improvement in information and communication technology for their languages. This work intends to
improve the ASR performance for Myanmar language by changing different Convolutional Neural Network
(CNN) hyperparameters such as number of feature maps and pooling size. CNN has the abilities of
reducing in spectral variations and modeling spectral correlations that exist in the signal due to the locality
and pooling operation. Therefore, the impact of the hyperparameters on CNN accuracy in ASR tasks is
investigated. A 42-hr-data set is used as training data and the ASR performance was evaluated on two open
test sets: web news and recorded data. As Myanmar language is a syllable-timed language, ASR based on
syllable was built and compared with ASR based on word. As the result, it gained 16.7% word error rate
(WER) and 11.5% syllable error rate (SER) on TestSet1. And it also achieved 21.83% WER and 15.76%
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Utterance Based Speaker Identification Using ANNIJCSEA Journal
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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.
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.
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 recognition techniques are one of the most important modern technologies. Many different systems have been developed in terms of methods used in the extraction of features and methods of classification. Voice recognition includes two areas: speech recognition and speaker recognition, where the research is confined to the field of speech recognition. The research presents a proposal to improve the performance of single word recognition systems by an algorithm that combines more than one of the techniques used in character extraction and modulation of the neural network to study the effects of recognition science and study the effect of noise on the proposed system. In this research four systems of speech recognition were studied, the first system adopted the MFCC algorithm to extract the features. The second system adopted the PLP algorithm, while the third system was based on combining the two previous algorithms in addition to the zero-passing rate. In the fourth system, the neural network used in the differentiation process was modified and the error ratio was determined. The impact of noise on these previous systems. The outcomes were looked at regarding the rate of recognizable proof and the season of preparing the neural network for every system independently, to get a rate of distinguishing proof and quiet up to 98% utilizing the proposed framework.
SPEECH RECOGNITION BY IMPROVING THE PERFORMANCE OF ALGORITHMS USED IN DISCRIM...ijcsit
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Accents of English have been investigated for many years both from the perspective of native and non-native speakers of the language. Various research results imply that non-native speakers of English language produce certain speech characteristics which are uncommon in native speakers’ speech. This is because non-native speakers do not produce the same tongue movement as native speakers. This paper presents an isolated English word recognition system devised with the speech of local Bangladeshi people, who are also non-native speakers of English language. Here, we have also noticed a different speech characteristic which is not available within the speech of native English speakers. Two acoustic features, ‘pitch’ and ‘formants’ have been utilized to develop the system. The system is speaker-independent and stands on Template based approach. The recognition method applied here is very simple and the recognition accuracy is also very satisfactory.
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.
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Hindi digits recognition system on speech data collected in different natural...csandit
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Chapter wise All Notes of First year Basic Civil Engineering.pptxDenish Jangid
Chapter wise All Notes of First year Basic Civil Engineering
Syllabus
Chapter-1
Introduction to objective, scope and outcome the subject
Chapter 2
Introduction: Scope and Specialization of Civil Engineering, Role of civil Engineer in Society, Impact of infrastructural development on economy of country.
Chapter 3
Surveying: Object Principles & Types of Surveying; Site Plans, Plans & Maps; Scales & Unit of different Measurements.
Linear Measurements: Instruments used. Linear Measurement by Tape, Ranging out Survey Lines and overcoming Obstructions; Measurements on sloping ground; Tape corrections, conventional symbols. Angular Measurements: Instruments used; Introduction to Compass Surveying, Bearings and Longitude & Latitude of a Line, Introduction to total station.
Levelling: Instrument used Object of levelling, Methods of levelling in brief, and Contour maps.
Chapter 4
Buildings: Selection of site for Buildings, Layout of Building Plan, Types of buildings, Plinth area, carpet area, floor space index, Introduction to building byelaws, concept of sun light & ventilation. Components of Buildings & their functions, Basic concept of R.C.C., Introduction to types of foundation
Chapter 5
Transportation: Introduction to Transportation Engineering; Traffic and Road Safety: Types and Characteristics of Various Modes of Transportation; Various Road Traffic Signs, Causes of Accidents and Road Safety Measures.
Chapter 6
Environmental Engineering: Environmental Pollution, Environmental Acts and Regulations, Functional Concepts of Ecology, Basics of Species, Biodiversity, Ecosystem, Hydrological Cycle; Chemical Cycles: Carbon, Nitrogen & Phosphorus; Energy Flow in Ecosystems.
Water Pollution: Water Quality standards, Introduction to Treatment & Disposal of Waste Water. Reuse and Saving of Water, Rain Water Harvesting. Solid Waste Management: Classification of Solid Waste, Collection, Transportation and Disposal of Solid. Recycling of Solid Waste: Energy Recovery, Sanitary Landfill, On-Site Sanitation. Air & Noise Pollution: Primary and Secondary air pollutants, Harmful effects of Air Pollution, Control of Air Pollution. . Noise Pollution Harmful Effects of noise pollution, control of noise pollution, Global warming & Climate Change, Ozone depletion, Greenhouse effect
Text Books:
1. Palancharmy, Basic Civil Engineering, McGraw Hill publishers.
2. Satheesh Gopi, Basic Civil Engineering, Pearson Publishers.
3. Ketki Rangwala Dalal, Essentials of Civil Engineering, Charotar Publishing House.
4. BCP, Surveying volume 1
Chapter wise All Notes of First year Basic Civil Engineering.pptx
Powerpoint on Linear Predictive coding.pptx
1. “LPC Analysis of Kannada Syllables”
Under the Guidance of
Prof. K. Indira
Dept. of E&C, RIT, Bangalore
Presented by
Vinodkumar A G - 1MS20EC127
Sridhar B - 1MS20EC111
Vinayak M B - 1MS20EC125
Vinay Swastik H - 1MS20EC124
MAJOR PROJECT
2. Table of Contents
1. Introduction and overview of the project
2. Problem statement
3. Project objectives and scope
4. Literature survey
5. Methodology, Proposed Work and Preliminary Results
6. References
3. INTRODUCTION
• The project will involve implementing the LPC algorithm to model the vocal tract, extracting the
formant frequencies from the LPC model, and comparing the results with known formant frequencies.
• The estimation of formant frequencies is an important task in speech signal processing, as it provides
information about the spectral characteristics of the vocal tract. Linear Predictive Coding (LPC) is a
widely used technique for speech analysis and has been shown to be effective for estimating formant
frequencies.
• Overall, the goal of this project is to develop a practical understanding of LPC and its applications in
speech analysis and to gain insights into the spectral characteristics of the vocal tract.
4. PROBLEM STATEMENT
To estimate Formant frequencies of the Kannada
Vowels and Consonants using Linear Predictive
Coefficients and verify using Praat tool.
5. Project Objectives
The main objective of this report is to estimate the formant frequencies of Kannada
vowels and consonants using Linear Predictive Coding (LPC) analysis.
The formant frequencies of speech sounds provide information about the resonant
properties of the vocal tract, which are essential for understanding the acoustic properties
of speech sounds.
The estimated formant frequencies can be used for further analysis and modeling of
Kannada speech signals.
An additional objective may involve applying the estimated formant and pitch
frequencies in practical speech processing tasks. This could include applications such as
speech recognition, speech synthesis, or speaker identification specific to Kannada
language.
6. Literature Survey
1.Performance Analysis of Kannada Phonetics: Vowels, Fricatives and Stop
Consonants Using LP Spectrum.
-By Shivakumar M and Latha Mariswamy
• A dataset of Kannada speech samples is collected, including a variety of phonetic units such as vowels,
fricatives, and stop consonants, recorded from native Kannada speakers.
• LP spectrum analysis is performed on the collected speech samples to estimate the spectral envelope and
formant frequencies.
• The study presents the findings of the spectral analysis, highlighting the spectral characteristics, patterns,
and variations observed in Kannada vowels, fricatives, and stop consonants.
• The accuracy and effectiveness of the LP spectrum analysis in capturing the phonetic properties of Kannada
are evaluated by comparing the estimated spectral features with known phonetic characteristics.
7. 2.Formants and LPC Analysis of Kannada Vowels Speech Signal
-By K. Indira, Sadashiva Chakrasali and Umesh Bilembagi
• The speech signal is down-sampled by a factor of 6 after passing through a low-pass filter, resulting
in a sampling frequency of 7350Hz.
• Pre-emphasis is applied to enhance the power of high frequency signals before LPC coefficients are
extracted using an autoregressive filter of varying order.
• The LPC filter is then used to obtain the LP residual, and frequency responses of LPC filters for
different orders are compared with the formants of corresponding vowels noted from a tool.
8. 3.Extraction of Speech Pitch and Formant Frequencies using
Discrete Wavelet Transform.
-By Sajad Hamzenejadi, Seyed Amir Yousef Hosseini Goki and Mahdieh Ghazvini
• The paper proposes a method for estimating speech pitch and formant frequencies using Discrete
Wavelet
Transform.
• DWT is used to decompose the speech signal into sub-bands, and the pitch and formant frequencies are
estimated from each sub-band.
• The method is advantageous because it captures both time and frequency information, and is efficiently
implemented using filter banks.
• The proposed method is shown to outperform existing methods for pitch and formant frequency
estimation in terms of accuracy and robustness.
9. 4.Formant Text to Speech Synthesis Using Artificial Neural Networks.
-By Gurinder Kaur and Parminder Singh
• The paper proposes a method for formant-based Text-to-Speech (TTS)
synthesis using Artificial Neural Networks (ANN).
• The method involves training an ANN on a set of formant frequency
parameters and their corresponding phonetic labels to generate synthetic
speech.
• The paper discusses the advantages of using formant-based synthesis over
concatenative TTS, including improved naturalness and flexibility.
• The proposed method is shown to achieve high-quality speech synthesis with
low computational complexity and outperform existing methods in terms of
naturalness and intelligibility.
18. Conclusion:
• In this work Kannada vowels and consonants were recorded from different age groups.
• Formants frequencies of corresponding Vowels and Consonants were computed. The variation of
formant frequencies across different gender and different age groups are shown in tables.
• The analysis is carried out separately for male and female speakers. The preliminary analysis of
frequency domain characteristics of vowels shows significant variations across different gender
and age groups.
• The importance of F1, F2, F3, F4 (formants) and their impact on order of the LPC filter have been
studied thoroughly in great details. The results have indicated the significant dependency of
speech signal characteristics on gender and different age groups.
19. Future work:
1.LPC-based formant frequency estimation can be used for speech enhancement,
speech recognition, voice conversion, and speech pathology diagnosis.
2.In speech enhancement, LPC can help to remove noise and other unwanted
distortions from speech signals.
3.In speech recognition, LPC-based formant frequency estimation can improve
accuracy and mitigate the effects of noise and other distortions.
4.In voice conversion, LPC can be used to convert the formant frequencies of one
speaker's voice to those of another speaker.
5.In speech pathology diagnosis, LPC-based formant frequency estimation can be
used to identify deviations from normal speech patterns and assist in diagnosis.
20. REFERENCES
1. Latha, M., M. Shivakumar, and R. Manjula. "Performance Analysis of Kannada Phonetics: Vowels,
Fricatives and Stop Consonants Using LP Spectrum." SN Computer Science 1, no. 2 (2020): 84.
2. Chakrasali Sadashiva, Umesh Bilembagi, and K. Indira. "Formants and LPC analysis of
Kannada vowel speech signals." In 2018 3rd IEEE International Conference on Recent
Trends in Electronics,Information & Communication Technology (RTEICT), pp. 945-948.
IEEE, 2018.
3. Dhiman Chowdhury , Md. Raju Ahmed Ripan,Md. Mehedihasan “Speech Features:
Pitch and Formant Extraction of Vowel Sounds Using Autocorrelation and Frequency
Domain Spectral Analysis” International conference on Innovation in Engineering
and Technology (ICIET),27-29 Dec 2018.
4. Sajad Hamzenejad , Seyed Amir Yousef Hosseini Goki, Mahdieh Ghazvini “Extraction
of Speech Pitch and Formant Frequencies using Discrete Wavelet Transform”, 2019 7th
Iranian Joint Congress on Fuzzy and Intelligent Systems (CFIS).
21. 5. U. Shrawankar and V. Thakare, “Feature Extraction for a Speech Recognition System in Noisy
Environment: A Study”, in Proc. Second Int. Conf. on Computer Engineering and Applications, 19-21
Mar. 2010.
6. EV Raghavendra, P. Vijay Aditya and K. Prahalad, "Speech synthesis using artificial neural networks
2010 National Conference On Communications (NCC) Chennai, India 2010, pp: 1-5, dor 10.1109/NCC
2010 5430190
7. Reddy MV, Hanumanthappa M. Kannada phonemes to speech dictionary: statistical approach. Int J Eng
Res Appl. 2017;7(1):77–80.
8. Sarika Hegde KK, Achary KK, Shetty S. Statistical analysis of features and classification of alpha
syllabary, sounds in Kannada language. New York: Springer; 2014.
9. Formant Text To Speech Synthesis Using Artificial Neural Networks, 2019 Second International
Conference on Advanced Computational and Communication Paradigms (ICACCP).
22. This Photo by Unknown author is licensed under CC BY-SA.