1) The document proposes using linear prediction (LP) residual and neural networks to classify audio clips. LP residual captures audio-specific information not present in spectral features alone.
2) Autoassociative neural networks (AANN) are used to capture information from the LP residual, which is difficult to extract using signal processing. Multilayer perceptrons (MLP) then classify the audio using AANN-extracted features.
3) The approach is tested on classifying clips into 5 categories (speech, music, noise, cartoon, advertisement) using residual features captured by AANN in addition to spectral features, achieving better performance than spectral features alone.
IJERA (International journal of Engineering Research and Applications) is International online, ... peer reviewed journal. For more detail or submit your article, please visit www.ijera.com
This document proposes an automatic emotion recognition system that analyzes audio information to classify human emotions. It uses spectral features and MFCC coefficients for feature extraction from voice signals. Then, a deep learning-based LSTM algorithm is used for classification. The system is evaluated on three audio datasets. Recurrent convolutional neural networks are proposed to capture temporal and frequency dependencies in speech spectrograms. The system aims to improve on existing methods which have lower accuracy and require more computational resources for implementation.
IJERD (www.ijerd.com) International Journal of Engineering Research and Devel...IJERD Editor
This document summarizes and compares several techniques for enhancing the intelligibility of speech signals corrupted by noise. It describes single channel techniques like spectral subtraction, spectral subtraction with oversubtraction, and nonlinear spectral subtraction. It also covers multi-channel techniques such as adaptive noise cancellation and multisensory beamforming. Additionally, it discusses spectral subtraction using adaptive averaging, noise reduction using enhanced Wiener filtering, and other adaptive neuro-fuzzy techniques for speech enhancement. The goal of these techniques is to improve the quality and intelligibility of noisy speech signals.
T Silva, D D Karunaratna, G N Wikramanayake, K P Hewagamage, G K A Dias (2004) "Speaker Search and Indexing for Multimedia Databases" In:6th International Information Technology Conference, Edited by:V.K. Samaranayake et al. pp. 157-162. Infotel Lanka Society, Colombo, Sri Lanka: IITC Nov 29-Dec 1, ISBN: 955-8974-01-3
Knn a machine learning approach to recognize a musical instrumentIJARIIT
An outline is provided of a proposed system to recognize musical instruments using machine learning techniques. The system first extracts features from audio files using the MIR toolbox in Matlab. It then uses a hybrid feature selection method and vector quantization to identify instruments. Specifically, the key audio descriptors are selected and feature vectors are generated and matched to standard vectors to classify the instrument. The k-nearest neighbors algorithm is used for classification. Preliminary results show the system can accurately recognize instruments based on extracted acoustic features.
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.
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.
IJERA (International journal of Engineering Research and Applications) is International online, ... peer reviewed journal. For more detail or submit your article, please visit www.ijera.com
This document proposes an automatic emotion recognition system that analyzes audio information to classify human emotions. It uses spectral features and MFCC coefficients for feature extraction from voice signals. Then, a deep learning-based LSTM algorithm is used for classification. The system is evaluated on three audio datasets. Recurrent convolutional neural networks are proposed to capture temporal and frequency dependencies in speech spectrograms. The system aims to improve on existing methods which have lower accuracy and require more computational resources for implementation.
IJERD (www.ijerd.com) International Journal of Engineering Research and Devel...IJERD Editor
This document summarizes and compares several techniques for enhancing the intelligibility of speech signals corrupted by noise. It describes single channel techniques like spectral subtraction, spectral subtraction with oversubtraction, and nonlinear spectral subtraction. It also covers multi-channel techniques such as adaptive noise cancellation and multisensory beamforming. Additionally, it discusses spectral subtraction using adaptive averaging, noise reduction using enhanced Wiener filtering, and other adaptive neuro-fuzzy techniques for speech enhancement. The goal of these techniques is to improve the quality and intelligibility of noisy speech signals.
T Silva, D D Karunaratna, G N Wikramanayake, K P Hewagamage, G K A Dias (2004) "Speaker Search and Indexing for Multimedia Databases" In:6th International Information Technology Conference, Edited by:V.K. Samaranayake et al. pp. 157-162. Infotel Lanka Society, Colombo, Sri Lanka: IITC Nov 29-Dec 1, ISBN: 955-8974-01-3
Knn a machine learning approach to recognize a musical instrumentIJARIIT
An outline is provided of a proposed system to recognize musical instruments using machine learning techniques. The system first extracts features from audio files using the MIR toolbox in Matlab. It then uses a hybrid feature selection method and vector quantization to identify instruments. Specifically, the key audio descriptors are selected and feature vectors are generated and matched to standard vectors to classify the instrument. The k-nearest neighbors algorithm is used for classification. Preliminary results show the system can accurately recognize instruments based on extracted acoustic features.
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.
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.
This document proposes a video genre classification method using only audio features extracted from video clips. It uses Multivariate Adaptive Regression Splines (MARS) to build classification models for different genres based on low-level audio features like MFCCs, zero crossing rate, short-time energy, etc. extracted from a dataset of news, cartoons, sports, music and dahmas video clips. The models are able to accurately classify video genres with an overall classification rate of 91.83% based on the important audio features identified for each genre by the MARS models.
This document discusses using deep neural networks for speech enhancement by finding a mapping between noisy and clean speech signals. It aims to handle a wide range of noises by using a large training dataset with many noise/speech combinations. Techniques like global variance equalization and dropout are used to improve generalization. Experimental results show improvements over MMSE techniques, with the ability to suppress nonstationary noise and avoid musical artifacts. The introduction provides background on speech enhancement, recognition using HMMs and other models, and the role of deep learning advances.
Designing an Efficient Multimodal Biometric System using Palmprint and Speech...IDES Editor
This document summarizes a research paper that proposes a multimodal biometric system using palmprint and speech signals. It extracts features from each modality using different methods. For speech, it uses Subband Cepstral Coefficients extracted via a wavelet packet transform. For palmprint, it uses a Modified Canonical Form method. The features are fused at the score level using a weighted sum rule. The system is tested on a database of over 300 subjects, and results show improved recognition rates compared to single modalities.
Novel Approach of Implementing Psychoacoustic model for MPEG-1 Audioinventy
Research Inventy : International Journal of Engineering and Science is published by the group of young academic and industrial researchers with 12 Issues per year. It is an online as well as print version open access journal that provides rapid publication (monthly) of articles in all areas of the subject such as: civil, mechanical, chemical, electronic and computer engineering as well as production and information technology. The Journal welcomes the submission of manuscripts that meet the general criteria of significance and scientific excellence. Papers will be published by rapid process within 20 days after acceptance and peer review process takes only 7 days. All articles published in Research Inventy will be peer-reviewed.
This document discusses optimization techniques for designing ultra-wideband planar monopole antennas. It presents two powerful design methodologies: size optimization using design of experiments, and topology optimization using binary particle swarm optimization. Size optimization is a systematic approach that varies geometric parameters to achieve design goals with a small number of simulations. Topology optimization determines the optimal metal distribution within the design area without a predefined shape using an automatic approach. These techniques are demonstrated by designing UWB antennas and band-notched UWB antennas, improving efficiency over trial-and-error approaches.
DATA HIDING IN AUDIO SIGNALS USING WAVELET TRANSFORM WITH ENHANCED SECURITYcsandit
Rapid increase in data transmission over internet results in emphasis on information security.
Audio steganography is used for secure transmission of secret data with audio signal as the
carrier. In the proposed method, cover audio file is transformed from space domain to wavelet
domain using lifting scheme, leading to secure data hiding. Text message is encrypted using
dynamic encryption algorithm. Cipher text is then hidden in wavelet coefficients of cover audio
signal. Signal to Noise Ratio (SNR) and Squared Pearson Correlation Coefficient (SPCC)
values are computed to judge the quality of the stego audio signal. Results show that stego
audio signal is perceptually indistinguishable from the cover audio signal. Stego audio signal is
robust even in presence of external noise. Proposed method provides secure and least error
data extraction.
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.
1. The document is a scheme of work for Form 4 students at SM Sains Seremban covering 10 units from 2010.
2. Each unit covers obtaining information, processing information, presenting information, grammar, vocabulary and educational emphasis over 3 levels of difficulty.
3. The units cover topics in science like the Earth, heat, natural resources, energy, forces, human body systems, genetics, nutrition and more.
The document is a research paper that studies using a neural network model for fingerprint recognition. It discusses how fingerprint recognition is an important technique for security and restricting intruders. The paper proposes using an artificial neural network with backpropagation training to recognize fingerprints. It describes collecting fingerprint images, classifying them, enhancing the images, and training the neural network to match images and recognize fingerprints with high accuracy. The methodology, implementation, and results of using a backpropagation neural network for fingerprint recognition are analyzed.
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.
This document presents a text-dependent speaker recognition system using neural networks that aims to improve recognition accuracy. It proposes changing the number of Mel Frequency Cepstral Coefficients (MFCCs) used in training. Voice Activity Detection is also used as a preprocessing step. Experimental results show recognition accuracy increases from 70.41% to a maximum of 92.91% as the number of MFCCs increases from 14 to 20, but then decreases with more MFCCs. The system is implemented on a Raspberry Pi for hardware acceleration.
QUALITATIVE ANALYSIS OF PLP IN LSTM FOR BANGLA SPEECH RECOGNITIONijma
This document summarizes a study that compares different acoustic feature extraction methods (LPC, MFCC, PLP) for a Bangla speech recognition system using LSTM neural networks. It finds that PLP outperforms MFCC and LPC based on statistical distance measurements of phoneme coefficients. PLP shows better distinction between phonemes compared to MFCC and LPC. While RNN/LSTM are inherently slow, combining PLP with faster networks like Transformers may improve performance for large datasets.
International Journal of Engineering and Science Invention (IJESI)inventionjournals
International Journal of Engineering and Science Invention (IJESI) is an international journal intended for professionals and researchers in all fields of computer science and electronics. IJESI publishes research articles and reviews within the whole field Engineering Science and Technology, new teaching methods, assessment, validation and the impact of new technologies and it will continue to provide information on the latest trends and developments in this ever-expanding subject. The publications of papers are selected through double peer reviewed to ensure originality, relevance, and readability. The articles published in our journal can be accessed online.
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 proposes a new localization method called A2L (Angle to Landmark) for wireless sensor networks. A2L uses angle of arrival measurements between sensor nodes and a subset of nodes equipped with GPS (landmarks) to determine the positions of non-landmark nodes. Compared to previous methods like APS and AHLoS that also use angle and distance measurements, simulations show that A2L can locate a greater number of nodes with higher accuracy while requiring fewer connections between nodes. The method is also low-cost since it does not require each node to have GPS or other expensive equipment.
1. The document is a scheme of work for Form 4 students in Sains Seremban, Seremban for the year 2011. It outlines 11 units to be covered from weeks 1-34 with objectives, activities, and emphasis for each unit.
2. The units cover topics in science, technology, environment and other subjects. For each unit, students will obtain information through listening, reading, and instructions. They will then process the information by identifying definitions, classifying data, and making inferences.
3. Students will present information using methods like notes, reports, diagrams and charts. Grammar, vocabulary, and 21st century skills are integrated into the lessons with an emphasis on thinking skills,
Bayesian distance metric learning and its application in automatic speaker re...IJECEIAES
This document proposes a state-of-the-art automatic speaker recognition system based on Bayesian distance metric learning as a feature extractor. It explores constraints on the distance between modified and simplified i-vector pairs from the same speaker and different speakers. An approximation of the distance metric is used as a weighted covariance matrix from the higher eigenvectors of the covariance matrix, which is used to estimate the posterior distribution of the metric distance. This Bayesian distance learning approach achieves better performance than advanced methods and is insensitive to normalization compared to cosine scores. It is also effective with limited training data.
IRJET- Music Genre Recognition using Convolution Neural NetworkIRJET Journal
1. The document describes a study that uses a Convolutional Neural Network (CNN) model to classify music genres based on labeled Mel spectrograms of audio clips.
2. A CNN model is trained on a dataset of 1000 audio clips across 10 genres. The trained model is then used to classify new, unlabeled audio clips by genre based on their Mel spectrogram representation.
3. CNNs are well-suited for this task as their convolutional layers can extract hierarchical features from the Mel spectrogram images that are indicative of different genres. The study aims to develop an automated music genre classification system using deep learning techniques.
This document discusses a proposed system for classifying audio scenes in action movies. It aims to provide scene recognition and detection by separating audio classes and obtaining better sound classification accuracy. The system extracts audio features like zero-crossing rate, short-time energy, volume root mean square, and volume dynamic range. It then uses hidden Markov models and support vector machines to classify audio scenes, labeling them as happy, miserable, or action scenes. Sound event types classified include gunshots, screams, car crashes, talking, laughter, fighting, shouting, and background crowd noise. The goal is to index and retrieve interesting events from action movies to engage viewers.
A computationally efficient learning model to classify audio signal attributesIJECEIAES
The era of machine learning has opened up groundbreaking realities and opportunities in the field of medical diagnosis. However, it is also observed that faster and proper diagnosis of any diseases/medical conditions require proper analysis and classification of digital signal data. It indicates the proper identification of tumors in the brain. Brain magnetic resonance imaging (MRI) data has to be appropriately classified, and similarly, pulse signal analysis is required to evaluate the human heart operating condition. Several studies have used machine learning (ML) modeling to classify speech signals, but very few studies have explored the classification of audio signal attributes in the context of intelligent healthcare monitoring. The study thereby aims to introduce novel mathematical modeling to analyze and classify synthetic pulse audio signal attributes with cost-effective computation. The numerical modeling is composed of several functional blocks where deep neural network-based learning (DNNL) plays a crucial role during the training phase, and also it is further combined with a recurrent structure of long-short term memory (R-LSTM) feedback connections (FCs). The design approaches further experiment in a numerical computing environment in terms of accuracy and computational aspects. The classification outcome of the proposed approach shows that it attains approximately 85% accuracy, which is comparable to the baseline approaches and execution time.
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
IRJET- Machine Learning and Noise Reduction Techniques for Music Genre Classi...IRJET Journal
This document discusses using machine learning and deep learning techniques to classify music genres automatically. It proposes applying noise reduction techniques to audio files using Fourier analysis before feeding them into models. A convolutional neural network is trained on mel-spectrograms of audio to classify genres. Supervised machine learning models like random forest and XGBoost are also explored using extracted audio features. The proposed system applies noise reduction to preprocessed audio then uses a CNN or supervised learning models to classify music genres.
This document proposes a video genre classification method using only audio features extracted from video clips. It uses Multivariate Adaptive Regression Splines (MARS) to build classification models for different genres based on low-level audio features like MFCCs, zero crossing rate, short-time energy, etc. extracted from a dataset of news, cartoons, sports, music and dahmas video clips. The models are able to accurately classify video genres with an overall classification rate of 91.83% based on the important audio features identified for each genre by the MARS models.
This document discusses using deep neural networks for speech enhancement by finding a mapping between noisy and clean speech signals. It aims to handle a wide range of noises by using a large training dataset with many noise/speech combinations. Techniques like global variance equalization and dropout are used to improve generalization. Experimental results show improvements over MMSE techniques, with the ability to suppress nonstationary noise and avoid musical artifacts. The introduction provides background on speech enhancement, recognition using HMMs and other models, and the role of deep learning advances.
Designing an Efficient Multimodal Biometric System using Palmprint and Speech...IDES Editor
This document summarizes a research paper that proposes a multimodal biometric system using palmprint and speech signals. It extracts features from each modality using different methods. For speech, it uses Subband Cepstral Coefficients extracted via a wavelet packet transform. For palmprint, it uses a Modified Canonical Form method. The features are fused at the score level using a weighted sum rule. The system is tested on a database of over 300 subjects, and results show improved recognition rates compared to single modalities.
Novel Approach of Implementing Psychoacoustic model for MPEG-1 Audioinventy
Research Inventy : International Journal of Engineering and Science is published by the group of young academic and industrial researchers with 12 Issues per year. It is an online as well as print version open access journal that provides rapid publication (monthly) of articles in all areas of the subject such as: civil, mechanical, chemical, electronic and computer engineering as well as production and information technology. The Journal welcomes the submission of manuscripts that meet the general criteria of significance and scientific excellence. Papers will be published by rapid process within 20 days after acceptance and peer review process takes only 7 days. All articles published in Research Inventy will be peer-reviewed.
This document discusses optimization techniques for designing ultra-wideband planar monopole antennas. It presents two powerful design methodologies: size optimization using design of experiments, and topology optimization using binary particle swarm optimization. Size optimization is a systematic approach that varies geometric parameters to achieve design goals with a small number of simulations. Topology optimization determines the optimal metal distribution within the design area without a predefined shape using an automatic approach. These techniques are demonstrated by designing UWB antennas and band-notched UWB antennas, improving efficiency over trial-and-error approaches.
DATA HIDING IN AUDIO SIGNALS USING WAVELET TRANSFORM WITH ENHANCED SECURITYcsandit
Rapid increase in data transmission over internet results in emphasis on information security.
Audio steganography is used for secure transmission of secret data with audio signal as the
carrier. In the proposed method, cover audio file is transformed from space domain to wavelet
domain using lifting scheme, leading to secure data hiding. Text message is encrypted using
dynamic encryption algorithm. Cipher text is then hidden in wavelet coefficients of cover audio
signal. Signal to Noise Ratio (SNR) and Squared Pearson Correlation Coefficient (SPCC)
values are computed to judge the quality of the stego audio signal. Results show that stego
audio signal is perceptually indistinguishable from the cover audio signal. Stego audio signal is
robust even in presence of external noise. Proposed method provides secure and least error
data extraction.
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.
1. The document is a scheme of work for Form 4 students at SM Sains Seremban covering 10 units from 2010.
2. Each unit covers obtaining information, processing information, presenting information, grammar, vocabulary and educational emphasis over 3 levels of difficulty.
3. The units cover topics in science like the Earth, heat, natural resources, energy, forces, human body systems, genetics, nutrition and more.
The document is a research paper that studies using a neural network model for fingerprint recognition. It discusses how fingerprint recognition is an important technique for security and restricting intruders. The paper proposes using an artificial neural network with backpropagation training to recognize fingerprints. It describes collecting fingerprint images, classifying them, enhancing the images, and training the neural network to match images and recognize fingerprints with high accuracy. The methodology, implementation, and results of using a backpropagation neural network for fingerprint recognition are analyzed.
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.
This document presents a text-dependent speaker recognition system using neural networks that aims to improve recognition accuracy. It proposes changing the number of Mel Frequency Cepstral Coefficients (MFCCs) used in training. Voice Activity Detection is also used as a preprocessing step. Experimental results show recognition accuracy increases from 70.41% to a maximum of 92.91% as the number of MFCCs increases from 14 to 20, but then decreases with more MFCCs. The system is implemented on a Raspberry Pi for hardware acceleration.
QUALITATIVE ANALYSIS OF PLP IN LSTM FOR BANGLA SPEECH RECOGNITIONijma
This document summarizes a study that compares different acoustic feature extraction methods (LPC, MFCC, PLP) for a Bangla speech recognition system using LSTM neural networks. It finds that PLP outperforms MFCC and LPC based on statistical distance measurements of phoneme coefficients. PLP shows better distinction between phonemes compared to MFCC and LPC. While RNN/LSTM are inherently slow, combining PLP with faster networks like Transformers may improve performance for large datasets.
International Journal of Engineering and Science Invention (IJESI)inventionjournals
International Journal of Engineering and Science Invention (IJESI) is an international journal intended for professionals and researchers in all fields of computer science and electronics. IJESI publishes research articles and reviews within the whole field Engineering Science and Technology, new teaching methods, assessment, validation and the impact of new technologies and it will continue to provide information on the latest trends and developments in this ever-expanding subject. The publications of papers are selected through double peer reviewed to ensure originality, relevance, and readability. The articles published in our journal can be accessed online.
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 proposes a new localization method called A2L (Angle to Landmark) for wireless sensor networks. A2L uses angle of arrival measurements between sensor nodes and a subset of nodes equipped with GPS (landmarks) to determine the positions of non-landmark nodes. Compared to previous methods like APS and AHLoS that also use angle and distance measurements, simulations show that A2L can locate a greater number of nodes with higher accuracy while requiring fewer connections between nodes. The method is also low-cost since it does not require each node to have GPS or other expensive equipment.
1. The document is a scheme of work for Form 4 students in Sains Seremban, Seremban for the year 2011. It outlines 11 units to be covered from weeks 1-34 with objectives, activities, and emphasis for each unit.
2. The units cover topics in science, technology, environment and other subjects. For each unit, students will obtain information through listening, reading, and instructions. They will then process the information by identifying definitions, classifying data, and making inferences.
3. Students will present information using methods like notes, reports, diagrams and charts. Grammar, vocabulary, and 21st century skills are integrated into the lessons with an emphasis on thinking skills,
Bayesian distance metric learning and its application in automatic speaker re...IJECEIAES
This document proposes a state-of-the-art automatic speaker recognition system based on Bayesian distance metric learning as a feature extractor. It explores constraints on the distance between modified and simplified i-vector pairs from the same speaker and different speakers. An approximation of the distance metric is used as a weighted covariance matrix from the higher eigenvectors of the covariance matrix, which is used to estimate the posterior distribution of the metric distance. This Bayesian distance learning approach achieves better performance than advanced methods and is insensitive to normalization compared to cosine scores. It is also effective with limited training data.
IRJET- Music Genre Recognition using Convolution Neural NetworkIRJET Journal
1. The document describes a study that uses a Convolutional Neural Network (CNN) model to classify music genres based on labeled Mel spectrograms of audio clips.
2. A CNN model is trained on a dataset of 1000 audio clips across 10 genres. The trained model is then used to classify new, unlabeled audio clips by genre based on their Mel spectrogram representation.
3. CNNs are well-suited for this task as their convolutional layers can extract hierarchical features from the Mel spectrogram images that are indicative of different genres. The study aims to develop an automated music genre classification system using deep learning techniques.
This document discusses a proposed system for classifying audio scenes in action movies. It aims to provide scene recognition and detection by separating audio classes and obtaining better sound classification accuracy. The system extracts audio features like zero-crossing rate, short-time energy, volume root mean square, and volume dynamic range. It then uses hidden Markov models and support vector machines to classify audio scenes, labeling them as happy, miserable, or action scenes. Sound event types classified include gunshots, screams, car crashes, talking, laughter, fighting, shouting, and background crowd noise. The goal is to index and retrieve interesting events from action movies to engage viewers.
A computationally efficient learning model to classify audio signal attributesIJECEIAES
The era of machine learning has opened up groundbreaking realities and opportunities in the field of medical diagnosis. However, it is also observed that faster and proper diagnosis of any diseases/medical conditions require proper analysis and classification of digital signal data. It indicates the proper identification of tumors in the brain. Brain magnetic resonance imaging (MRI) data has to be appropriately classified, and similarly, pulse signal analysis is required to evaluate the human heart operating condition. Several studies have used machine learning (ML) modeling to classify speech signals, but very few studies have explored the classification of audio signal attributes in the context of intelligent healthcare monitoring. The study thereby aims to introduce novel mathematical modeling to analyze and classify synthetic pulse audio signal attributes with cost-effective computation. The numerical modeling is composed of several functional blocks where deep neural network-based learning (DNNL) plays a crucial role during the training phase, and also it is further combined with a recurrent structure of long-short term memory (R-LSTM) feedback connections (FCs). The design approaches further experiment in a numerical computing environment in terms of accuracy and computational aspects. The classification outcome of the proposed approach shows that it attains approximately 85% accuracy, which is comparable to the baseline approaches and execution time.
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
IRJET- Machine Learning and Noise Reduction Techniques for Music Genre Classi...IRJET Journal
This document discusses using machine learning and deep learning techniques to classify music genres automatically. It proposes applying noise reduction techniques to audio files using Fourier analysis before feeding them into models. A convolutional neural network is trained on mel-spectrograms of audio to classify genres. Supervised machine learning models like random forest and XGBoost are also explored using extracted audio features. The proposed system applies noise reduction to preprocessed audio then uses a CNN or supervised learning models to classify music genres.
This document summarizes the key components of a voice recognition system, including signal modeling and pattern matching. Signal modeling represents converting speech signals into parameters through operations like spectral shaping and feature extraction. Feature extraction analyzes speech signals through temporal and spectral analysis techniques to obtain parameters like power, pitch, and vocal tract configuration. Pattern matching finds the parameter set from memory that most closely matches the input speech parameters. The document then discusses specific temporal analysis techniques like power and energy analysis, and spectral analysis techniques like filter banks, cepstral analysis, and linear predictive coding analysis used for feature extraction in voice recognition systems.
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.
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.
The document discusses using suprasegmental features present in linear prediction (LP) residual for audio clip classification. It explains that existing audio classification approaches miss important suprasegmental information and that statistics of the autocorrelation sequence of the Hilbert envelope of the LP residual contain audio-specific suprasegmental information that can enhance classification. An experiment is described that demonstrates classifying audio clips into 5 categories using support vector machines based on the variance of the autocorrelation sequence, achieving over 50% accuracy on average. Future work to improve classification performance by combining suprasegmental and other features is discussed.
IRJET- Implementing Musical Instrument Recognition using CNN and SVMIRJET Journal
This document summarizes research on implementing musical instrument recognition using convolutional neural networks (CNNs) and support vector machines (SVMs). The researchers aim to preprocess audio excerpts into images and use CNNs to achieve high accuracy in instrument classification. They will then combine CNN and SVM classifications and take a weighted average to achieve even higher accuracy. The document reviews several related works that used features like MFCCs and classifiers like SVMs, GMMs, and neural networks for instrument recognition. The researchers intend to use mel spectrograms and MFCCs to represent audio as images for CNN classification and improve music information retrieval and organization.
This document provides an overview of recent developments in sound recognition techniques. It discusses several methods for sound recognition, including matching pursuit algorithms with MFCC features, probabilistic distance support vector machines using generalized gamma modeling of STE features, and frequency vector principal component analysis. The document also reviews related literature on environmental sound recognition using time-frequency audio features and sound event recognition. It aims to present an updated survey on sound recognition methods and discuss future research trends in the field.
Literature Survey for Music Genre Classification Using Neural NetworkIRJET Journal
The document discusses literature on classifying music genres using neural networks. It summarizes several past studies that used techniques like convolutional neural networks (CNNs) and mel-frequency cepstral coefficients (MFCCs) on datasets like GTZAN to classify music into genres like blues, classical, country, etc. The document also outlines the system design for a proposed music genre classification system, including collecting the GTZAN dataset, preprocessing the audio files into mel-spectrograms, extracting features using MFCCs, and training a CNN model to classify segments of songs into genres. Classification accuracy of different models from prior studies ranged from 40-80%.
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.
Optimized audio classification and segmentation algorithm by using ensemble m...Venkat Projects
The document proposes an optimized audio classification and segmentation algorithm that segments audio streams into four types - pure speech, music, environment sound, and silence - using ensemble methods. It uses a hybrid classification approach of bagged support vector machines and artificial neural networks. The algorithm aims to accurately segment audio with minimum misclassification and requires less training data, making it suitable for real-time applications. It segments non-speech portions into music or environment sound and further divides speech into silence and pure speech. The algorithm achieves approximately 98% accurate segmentation.
IRJET- Musical Instrument Recognition using CNN and SVMIRJET Journal
This document discusses a study that uses convolutional neural networks (CNNs) and support vector machines (SVMs) to recognize musical instruments in audio recordings. The researchers aim to convert audio excerpts to images and use CNNs to classify instruments, then combine the CNN classifications with SVM classifications to improve accuracy. They discuss related work on instrument recognition using other methods. The proposed model uses MFCC features with SVM and passes audio converted to images through four convolutional layers and fully connected layers in the CNN. Combining the CNN and SVM results through weighted averaging is expected to provide higher accuracy than either method alone for classifying instruments in the IRMAS dataset.
Recognition of music genres using deep learning.IRJET Journal
This document discusses using deep learning techniques to recognize music genres from audio files. It evaluates three approaches: extracting Mel-spectrograms, MFCC plots, and chroma STFT features from audio and using those as input to CNN models. A CNN architecture with 5 conv layers performed best on Mel-spectrograms, achieving over 90% accuracy. MFCC plots achieved over 70% accuracy. Chroma STFT features performed worst at around 57% accuracy. In conclusion, Mel-spectrograms were found to be the most effective audio feature for music genre classification using deep learning.
Automatic Music Generation Using Deep LearningIRJET Journal
This document discusses automatic music generation using deep learning. It begins with an abstract describing how music is generated in the form of a sequence of ABC notes using deep learning concepts. LSTM or GRUs are commonly used for music generation as recurrent neural networks that can efficiently model sequences. The main purpose of the project described is to generate melodious and rhythmic music automatically using a recurrent neural network. It reviews approaches like WaveNet and LSTM for music generation and tools like Magenta and DeepJazz. The design uses a character RNN and LSTM network to classify and predict the next character in an ABC notation sequence to generate music.
CONTENT BASED AUDIO CLASSIFIER & FEATURE EXTRACTION USING ANN TECNIQUESAM Publications
Audio signals which include speech, music and environmental sounds are important types of media. The problem of distinguishing audio signals into these different audio types is thus becoming increasingly significant. A human listener can easily distinguish between different audio types by just listening to a short segment of an audio signal. However, solving this problem using computers has proven to be very difficult. Nevertheless, many systems with modest accuracy could still be implemented. The experimental results demonstrate the effectiveness of our classification system. The complete system is developed in ANN Techniques with Autonomic Computing system
This document discusses audio indexing and classification. It notes that (1) most stored data is multimedia like audio which is difficult to handle manually due to its large volume, and (2) an automatic method is needed to organize and use multimedia data appropriately. It then explores using linear prediction residual and suprasegmental features of audio signals to classify audio clips, as these carry additional perceptual information not captured by existing spectral analysis methods. The residual and suprasegmental features are shown to provide discriminative information between different audio classes.
Building Production Ready Search Pipelines with Spark and MilvusZilliz
Spark is the widely used ETL tool for processing, indexing and ingesting data to serving stack for search. Milvus is the production-ready open-source vector database. In this talk we will show how to use Spark to process unstructured data to extract vector representations, and push the vectors to Milvus vector database for search serving.
UiPath Test Automation using UiPath Test Suite series, part 6DianaGray10
Welcome to UiPath Test Automation using UiPath Test Suite series part 6. In this session, we will cover Test Automation with generative AI and Open AI.
UiPath Test Automation with generative AI and Open AI webinar offers an in-depth exploration of leveraging cutting-edge technologies for test automation within the UiPath platform. Attendees will delve into the integration of generative AI, a test automation solution, with Open AI advanced natural language processing capabilities.
Throughout the session, participants will discover how this synergy empowers testers to automate repetitive tasks, enhance testing accuracy, and expedite the software testing life cycle. Topics covered include the seamless integration process, practical use cases, and the benefits of harnessing AI-driven automation for UiPath testing initiatives. By attending this webinar, testers, and automation professionals can gain valuable insights into harnessing the power of AI to optimize their test automation workflows within the UiPath ecosystem, ultimately driving efficiency and quality in software development processes.
What will you get from this session?
1. Insights into integrating generative AI.
2. Understanding how this integration enhances test automation within the UiPath platform
3. Practical demonstrations
4. Exploration of real-world use cases illustrating the benefits of AI-driven test automation for UiPath
Topics covered:
What is generative AI
Test Automation with generative AI and Open AI.
UiPath integration with generative AI
Speaker:
Deepak Rai, Automation Practice Lead, Boundaryless Group and UiPath MVP
Threats to mobile devices are more prevalent and increasing in scope and complexity. Users of mobile devices desire to take full advantage of the features
available on those devices, but many of the features provide convenience and capability but sacrifice security. This best practices guide outlines steps the users can take to better protect personal devices and information.
HCL Notes and Domino License Cost Reduction in the World of DLAUpanagenda
Webinar Recording: https://www.panagenda.com/webinars/hcl-notes-and-domino-license-cost-reduction-in-the-world-of-dlau/
The introduction of DLAU and the CCB & CCX licensing model caused quite a stir in the HCL community. As a Notes and Domino customer, you may have faced challenges with unexpected user counts and license costs. You probably have questions on how this new licensing approach works and how to benefit from it. Most importantly, you likely have budget constraints and want to save money where possible. Don’t worry, we can help with all of this!
We’ll show you how to fix common misconfigurations that cause higher-than-expected user counts, and how to identify accounts which you can deactivate to save money. There are also frequent patterns that can cause unnecessary cost, like using a person document instead of a mail-in for shared mailboxes. We’ll provide examples and solutions for those as well. And naturally we’ll explain the new licensing model.
Join HCL Ambassador Marc Thomas in this webinar with a special guest appearance from Franz Walder. It will give you the tools and know-how to stay on top of what is going on with Domino licensing. You will be able lower your cost through an optimized configuration and keep it low going forward.
These topics will be covered
- Reducing license cost by finding and fixing misconfigurations and superfluous accounts
- How do CCB and CCX licenses really work?
- Understanding the DLAU tool and how to best utilize it
- Tips for common problem areas, like team mailboxes, functional/test users, etc
- Practical examples and best practices to implement right away
Communications Mining Series - Zero to Hero - Session 1DianaGray10
This session provides introduction to UiPath Communication Mining, importance and platform overview. You will acquire a good understand of the phases in Communication Mining as we go over the platform with you. Topics covered:
• Communication Mining Overview
• Why is it important?
• How can it help today’s business and the benefits
• Phases in Communication Mining
• Demo on Platform overview
• Q/A
Full-RAG: A modern architecture for hyper-personalizationZilliz
Mike Del Balso, CEO & Co-Founder at Tecton, presents "Full RAG," a novel approach to AI recommendation systems, aiming to push beyond the limitations of traditional models through a deep integration of contextual insights and real-time data, leveraging the Retrieval-Augmented Generation architecture. This talk will outline Full RAG's potential to significantly enhance personalization, address engineering challenges such as data management and model training, and introduce data enrichment with reranking as a key solution. Attendees will gain crucial insights into the importance of hyperpersonalization in AI, the capabilities of Full RAG for advanced personalization, and strategies for managing complex data integrations for deploying cutting-edge AI solutions.
AI 101: An Introduction to the Basics and Impact of Artificial IntelligenceIndexBug
Imagine a world where machines not only perform tasks but also learn, adapt, and make decisions. This is the promise of Artificial Intelligence (AI), a technology that's not just enhancing our lives but revolutionizing entire industries.
Best 20 SEO Techniques To Improve Website Visibility In SERPPixlogix Infotech
Boost your website's visibility with proven SEO techniques! Our latest blog dives into essential strategies to enhance your online presence, increase traffic, and rank higher on search engines. From keyword optimization to quality content creation, learn how to make your site stand out in the crowded digital landscape. Discover actionable tips and expert insights to elevate your SEO game.
“An Outlook of the Ongoing and Future Relationship between Blockchain Technologies and Process-aware Information Systems.” Invited talk at the joint workshop on Blockchain for Information Systems (BC4IS) and Blockchain for Trusted Data Sharing (B4TDS), co-located with with the 36th International Conference on Advanced Information Systems Engineering (CAiSE), 3 June 2024, Limassol, Cyprus.
HCL Notes und Domino Lizenzkostenreduzierung in der Welt von DLAUpanagenda
Webinar Recording: https://www.panagenda.com/webinars/hcl-notes-und-domino-lizenzkostenreduzierung-in-der-welt-von-dlau/
DLAU und die Lizenzen nach dem CCB- und CCX-Modell sind für viele in der HCL-Community seit letztem Jahr ein heißes Thema. Als Notes- oder Domino-Kunde haben Sie vielleicht mit unerwartet hohen Benutzerzahlen und Lizenzgebühren zu kämpfen. Sie fragen sich vielleicht, wie diese neue Art der Lizenzierung funktioniert und welchen Nutzen sie Ihnen bringt. Vor allem wollen Sie sicherlich Ihr Budget einhalten und Kosten sparen, wo immer möglich. Das verstehen wir und wir möchten Ihnen dabei helfen!
Wir erklären Ihnen, wie Sie häufige Konfigurationsprobleme lösen können, die dazu führen können, dass mehr Benutzer gezählt werden als nötig, und wie Sie überflüssige oder ungenutzte Konten identifizieren und entfernen können, um Geld zu sparen. Es gibt auch einige Ansätze, die zu unnötigen Ausgaben führen können, z. B. wenn ein Personendokument anstelle eines Mail-Ins für geteilte Mailboxen verwendet wird. Wir zeigen Ihnen solche Fälle und deren Lösungen. Und natürlich erklären wir Ihnen das neue Lizenzmodell.
Nehmen Sie an diesem Webinar teil, bei dem HCL-Ambassador Marc Thomas und Gastredner Franz Walder Ihnen diese neue Welt näherbringen. Es vermittelt Ihnen die Tools und das Know-how, um den Überblick zu bewahren. Sie werden in der Lage sein, Ihre Kosten durch eine optimierte Domino-Konfiguration zu reduzieren und auch in Zukunft gering zu halten.
Diese Themen werden behandelt
- Reduzierung der Lizenzkosten durch Auffinden und Beheben von Fehlkonfigurationen und überflüssigen Konten
- Wie funktionieren CCB- und CCX-Lizenzen wirklich?
- Verstehen des DLAU-Tools und wie man es am besten nutzt
- Tipps für häufige Problembereiche, wie z. B. Team-Postfächer, Funktions-/Testbenutzer usw.
- Praxisbeispiele und Best Practices zum sofortigen Umsetzen
How to Get CNIC Information System with Paksim Ga.pptxdanishmna97
Pakdata Cf is a groundbreaking system designed to streamline and facilitate access to CNIC information. This innovative platform leverages advanced technology to provide users with efficient and secure access to their CNIC details.
GraphSummit Singapore | The Art of the Possible with Graph - Q2 2024Neo4j
Neha Bajwa, Vice President of Product Marketing, Neo4j
Join us as we explore breakthrough innovations enabled by interconnected data and AI. Discover firsthand how organizations use relationships in data to uncover contextual insights and solve our most pressing challenges – from optimizing supply chains, detecting fraud, and improving customer experiences to accelerating drug discoveries.
For the full video of this presentation, please visit: https://www.edge-ai-vision.com/2024/06/building-and-scaling-ai-applications-with-the-nx-ai-manager-a-presentation-from-network-optix/
Robin van Emden, Senior Director of Data Science at Network Optix, presents the “Building and Scaling AI Applications with the Nx AI Manager,” tutorial at the May 2024 Embedded Vision Summit.
In this presentation, van Emden covers the basics of scaling edge AI solutions using the Nx tool kit. He emphasizes the process of developing AI models and deploying them globally. He also showcases the conversion of AI models and the creation of effective edge AI pipelines, with a focus on pre-processing, model conversion, selecting the appropriate inference engine for the target hardware and post-processing.
van Emden shows how Nx can simplify the developer’s life and facilitate a rapid transition from concept to production-ready applications.He provides valuable insights into developing scalable and efficient edge AI solutions, with a strong focus on practical implementation.
Maruthi Prithivirajan, Head of ASEAN & IN Solution Architecture, Neo4j
Get an inside look at the latest Neo4j innovations that enable relationship-driven intelligence at scale. Learn more about the newest cloud integrations and product enhancements that make Neo4j an essential choice for developers building apps with interconnected data and generative AI.
1. AUDIO CLIP CLASSIFICATION USING LP RESIDUAL AND NEURAL NETWORKS
MODELS
Anvita Bajpai and B. Yegnanarayana
Department of Computer Science and Engineering
Indian Institute of Technology Madras, Chennai- 600 036, India
anvita, yegna¡ @cs.iitm.ernet.in
ABSTRACT et al. classify audio into five categories of television (TV)
programs using spectral features [4]. Features based on
In this paper, we demonstrate the presence of audio-specific
amplitude, zero-crossing, bandwidth, band energy in the
information in the linear prediction (LP) residual, obtained
subbands, spectrum and periodicity properties, along with
after removing the predictable part of the signal. We empha-
hidden Markov model (HMM) for classification are explored
size the importance of information present in the LP resid-
for audio indexing applications in [5]. But it was shown
ual of audio signals, which if added to the spectral informa-
that perceptually significant information of audio data is
tion, can give a better performing system. Since it is dif-
present in the form of sequence of events, which can be
ficult to extract information from the residual using known
obtained after removing the predictable part in the audio
signal processing algorithms, neural networks (NN) models
data. Perceptually, there are some discriminating features
are proposed. In this paper, autoassociative neural networks
present in the residual which could help in various audio
(AANN) models are used to capture the audio-specific infor-
indexing tasks. The challenge lies in developing algorithms
mation from the LP residual of signals. Multilayer feedfor-
to capture these perceptually significant features from the
ward neural networks (MLFFNN) models or multilayer per-
residual, as it is difficult to extract information using known
ceptron (MLP) are used to classify the audio data using the
signal processing algorithms.
audio-specific information captured by AANN models.
Objective of this study is to explore the features in
addition to the features that are currently used to improve the
performance of an audio indexing system. In particular, fea-
1. INTRODUCTION
tures not used explicitly or implicitly in the current system
In this era of information technology, the data that we use is are being investigated. Many interesting and perceptually
mostly in the form of audio, video and multimedia. The data, important features are present in the residual signal obtained
once recorded and stored digitally, conveys no significant after removing the predictable part. Thus the main objective
information in order to organize and use it. The volume of of this study is to explore the features present in the linear
data is large, and is increasing daily. Therefore it is difficult prediction (LP) residual for audio clip classification task.
to organize the data manually. We need to have an automatic The reason for considering the residual data for study is that
method to index the data, for further search and retrieval. the residual part of the signal is generally subject to less
Audio plays an important role in classifying multimedia degradation as compared to the system part [6]. The residual
data as it contains significant information, and is easier to data contains higher order correlation among samples. As
process when compared to video data. For these reasons, known signal processing and statistical techniques are not
commercial products of audio retrieval are emerging, e.g., suitable to capture this correlation, an autoassociative neural
(http://www.musclefish.com) [1]. Content-based classifica- networks (AANN) model is proposed to capture these higher
tion of data into different categories is one important step for order correlations among samples of the residual of the
building an audio indexing system. audio data. AANN have already been studied to capture
information from the residual data for tasks such as speaker
In the traditional approach of audio indexing, audio recognition [7]. Further, multilayer feedforward neural
is first converted to text, and then it is given to text-based networks (MLFFNN) models or multilayer perceptron
search engines [2]. Drawbacks of this approach are: a£ (MLP) are proposed for decision making task using the
¢
not having accurate speech recognizer, b£ not using speech audio-specific information captured by AANN models.
¢
information present in form of prosody, and c£ not appli-
¢
cable for non-speech data like music. An elaborate audio
content categorization is proposed by Wold et al. [1], which The paper is organized as follows: Section 2 discusses
divides the audio content into sixteen groups. The authors extraction of the LP residual from audio data. Section 3 dis-
have used mean, variance and autocorrelation of loudness, cusses AANN models for capturing features in LP residual
pitch and bandwidth as audio features and a nearest neigh- for the audio clip classification. Section 4 discusses MLP
borhood classifier for the task. The authors quote 81% models for decision making. Section 5 presents the work-
classification accuracy for an audio database with 400 sound flow of the system. The results of the experimental studies
files. Guo et al. [3] have used features consisting of total are presented in Section 6. Various issues addressed in this
power, subband energies, bandwidth, pitch and MFCCs, and paper and possible directions for the future study are summa-
support vector machines (SVMs) for classification. Wang rized in Section 7.
2299