Interior vehicle acoustics are in close connection with our quality opinion. The noise in vehicle interior is complex and can be considered as a sum of various sound emission sources. A nice sounding vehicle is objective of the development departments for car acoustics. In the process of manufacturing the targets for a qualitatively high-valuable sound must be maintained. However, it is possible that production errors lead to a deviation from the wanted vehicle interior sound. This will result in customer complaints where for example a rattling or squeak refers to a worn-out or defective component. Also in this case, of course, the vehicle interior noise does not fulfill the targets of the process of development. For both cases there is currently no possibility for automated analysis of the vehicle interior noise. In this paper an approach for automated analysis of vehicle interior noise by means of neural algorithms is investigated. The presented system analyses microphone signals from car interior measured at real environmental conditions. This is in contrast to well known techniques, as e.g. acoustic engine test bench. Self-Organizing Maps combined with feature selection algorithms are used for acoustic pattern recognition. The presented system can be used in production process as well as a standalone ECU in car.
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.
Improvement in Traditional Set Partitioning in Hierarchical Trees (SPIHT) Alg...AM Publications
In this paper, an improved SPIHT algorithm based on Huffman coding and discrete wavelet transform for
image compression has been developed. The traditional SPIHT algorithm application is limited in terms of PSNR and
Compression Ratio. The improved SPIHT algorithm gives better performance as compared with traditional SPIHT
algorithm; here we used additional Huffman encoder along with SPIHT encoder. Huffman coding is an entropy
encoding algorithm uses a specific method for choosing the representation for each symbol, resulting in a prefix code.
The input gray scale image is decomposed by 'bior4.4' wavelet and wavelet coefficients are indexed and scanned by
DWT, then applied to encode the coefficient by SPIHT encoder followed by Huffman encoder which gives the
compressed image. At receiver side the decoding process is applied by Huffman decoder followed by SPIHT decoder.
The Reconstructed gray scale image looks similar to the applied gray scale image.
Vehicle logo recognition using histograms of oriented gradient descriptor and...TELKOMNIKA JOURNAL
Most of vehicle have the similar structures and designs. It is extremely complicated and difficult to identify and classify vehicle brands based on their structure and shape. As we requirea quick and reliable response, so vehicle logos are an alternative method of determining the type of a vehicle. In this paper, we propose a method for vehicle logo recognition based on feature selection method in a hybrid way. Vehicle logo images are first characterized by histograms of oriented gradient descriptors and the final features vector are then applied feature selection method to reduce the irrelevant information. Moreover, we release a new benchmark dataset for vehicle logo recognition and retrieval task namely, VLR-40. The experimental results are evaluated on this database which show the efficiency of the proposed approach.
This document discusses optical character recognition for vehicle number plates. It begins with an overview of how the process works, taking an image of a car, locating the number plate area, and using OCR to recognize the characters. It then describes the steps involved in more detail, including image division, detecting the number plate area, recognizing and parsing the plate, and applying OCR to the characters. An example of the process is provided with images. It also discusses techniques for recognizing characters, such as template matching and distance measurement, and provides limitations and references.
This document discusses vehicle license plate recognition (LPR) using image processing techniques in MATLAB. It describes how an LPR system works by capturing an image of a vehicle, preprocessing the image, extracting the license plate region, segmenting the characters, performing character recognition using template matching, and comparing the results to a database. The document provides details on preprocessing techniques like resizing, grayscale conversion and filtering. It also describes methods for license plate extraction using edge detection and morphological operations. Finally, it presents the MATLAB code used for this LPR system and discusses potential issues and future work using deep learning.
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.
A unique sorting algorithm with linear time & space complexityeSAT Journals
Abstract Sorting a list means selection of the particular permutation of the members of that list in which the final permutation contains members in increasing or in decreasing order. Sorted list is prerequisite of some optimized operations such as searching an element from a list, locating or removing an element to/ from a list and merging two sorted list in a database etc. As volume of information is growing up day by day in the world around us and these data are unavoidable to manage for real life situations, the efficient and cost effective sorting algorithms are required. There are several numbers of fundamental and problem oriented sorting algorithms but still now sorting a problem has attracted a great deal of research, perhaps due to the complexity of solving it efficiently and effectively despite of its simple and familiar statements. Algorithms having same efficiency to do a same work using different mechanisms must differ their required time and space. For that reason an algorithm is chosen according to one’s need with respect to space complexity and time complexity. Now a day, space (Memory) is available in market comparatively in cheap cost. So, time complexity is a major issue for an algorithm. Here, the presented approach is to sort a list with linear time and space complexity using divide and conquer rule by partitioning a problem into n (input size) number of sub problems then these sub problems are solved recursively. Required time and space for the algorithm is optimized through reducing the height of the recursive tree and reduced height is too small (as compared to the problem size) to evaluate. So, asymptotic efficiency of this algorithm is very high with respect to time and space. Keywords: sorting, searching, permutation, divide and conquer algorithm, asymptotic efficiency, space complexity, time complexity, and recursion.
A design of license plate recognition system using convolutional neural networkIJECEIAES
This document describes a license plate recognition system using convolutional neural networks. The system uses several preprocessing techniques like Sobel edge detection, morphological operations, and connected component analysis to localize, isolate, and segment characters from license plate images. A convolutional neural network with a four-layer architecture is then used for character recognition. Various hyperparameters of the CNN model like the number of feature maps, connection patterns, and learning parameters are optimized using 10-fold cross-validation. The proposed system achieves 74.7% accuracy in the preprocessing stage and 94.6% accuracy in the character recognition stage using CNN.
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.
Improvement in Traditional Set Partitioning in Hierarchical Trees (SPIHT) Alg...AM Publications
In this paper, an improved SPIHT algorithm based on Huffman coding and discrete wavelet transform for
image compression has been developed. The traditional SPIHT algorithm application is limited in terms of PSNR and
Compression Ratio. The improved SPIHT algorithm gives better performance as compared with traditional SPIHT
algorithm; here we used additional Huffman encoder along with SPIHT encoder. Huffman coding is an entropy
encoding algorithm uses a specific method for choosing the representation for each symbol, resulting in a prefix code.
The input gray scale image is decomposed by 'bior4.4' wavelet and wavelet coefficients are indexed and scanned by
DWT, then applied to encode the coefficient by SPIHT encoder followed by Huffman encoder which gives the
compressed image. At receiver side the decoding process is applied by Huffman decoder followed by SPIHT decoder.
The Reconstructed gray scale image looks similar to the applied gray scale image.
Vehicle logo recognition using histograms of oriented gradient descriptor and...TELKOMNIKA JOURNAL
Most of vehicle have the similar structures and designs. It is extremely complicated and difficult to identify and classify vehicle brands based on their structure and shape. As we requirea quick and reliable response, so vehicle logos are an alternative method of determining the type of a vehicle. In this paper, we propose a method for vehicle logo recognition based on feature selection method in a hybrid way. Vehicle logo images are first characterized by histograms of oriented gradient descriptors and the final features vector are then applied feature selection method to reduce the irrelevant information. Moreover, we release a new benchmark dataset for vehicle logo recognition and retrieval task namely, VLR-40. The experimental results are evaluated on this database which show the efficiency of the proposed approach.
This document discusses optical character recognition for vehicle number plates. It begins with an overview of how the process works, taking an image of a car, locating the number plate area, and using OCR to recognize the characters. It then describes the steps involved in more detail, including image division, detecting the number plate area, recognizing and parsing the plate, and applying OCR to the characters. An example of the process is provided with images. It also discusses techniques for recognizing characters, such as template matching and distance measurement, and provides limitations and references.
This document discusses vehicle license plate recognition (LPR) using image processing techniques in MATLAB. It describes how an LPR system works by capturing an image of a vehicle, preprocessing the image, extracting the license plate region, segmenting the characters, performing character recognition using template matching, and comparing the results to a database. The document provides details on preprocessing techniques like resizing, grayscale conversion and filtering. It also describes methods for license plate extraction using edge detection and morphological operations. Finally, it presents the MATLAB code used for this LPR system and discusses potential issues and future work using deep learning.
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.
A unique sorting algorithm with linear time & space complexityeSAT Journals
Abstract Sorting a list means selection of the particular permutation of the members of that list in which the final permutation contains members in increasing or in decreasing order. Sorted list is prerequisite of some optimized operations such as searching an element from a list, locating or removing an element to/ from a list and merging two sorted list in a database etc. As volume of information is growing up day by day in the world around us and these data are unavoidable to manage for real life situations, the efficient and cost effective sorting algorithms are required. There are several numbers of fundamental and problem oriented sorting algorithms but still now sorting a problem has attracted a great deal of research, perhaps due to the complexity of solving it efficiently and effectively despite of its simple and familiar statements. Algorithms having same efficiency to do a same work using different mechanisms must differ their required time and space. For that reason an algorithm is chosen according to one’s need with respect to space complexity and time complexity. Now a day, space (Memory) is available in market comparatively in cheap cost. So, time complexity is a major issue for an algorithm. Here, the presented approach is to sort a list with linear time and space complexity using divide and conquer rule by partitioning a problem into n (input size) number of sub problems then these sub problems are solved recursively. Required time and space for the algorithm is optimized through reducing the height of the recursive tree and reduced height is too small (as compared to the problem size) to evaluate. So, asymptotic efficiency of this algorithm is very high with respect to time and space. Keywords: sorting, searching, permutation, divide and conquer algorithm, asymptotic efficiency, space complexity, time complexity, and recursion.
A design of license plate recognition system using convolutional neural networkIJECEIAES
This document describes a license plate recognition system using convolutional neural networks. The system uses several preprocessing techniques like Sobel edge detection, morphological operations, and connected component analysis to localize, isolate, and segment characters from license plate images. A convolutional neural network with a four-layer architecture is then used for character recognition. Various hyperparameters of the CNN model like the number of feature maps, connection patterns, and learning parameters are optimized using 10-fold cross-validation. The proposed system achieves 74.7% accuracy in the preprocessing stage and 94.6% accuracy in the character recognition stage using CNN.
This document discusses various types of pollution including noise, air, water, soil, and radioactive pollution. For each type of pollution, it provides definitions, causes, effects on human health and environment, and methods for control and prevention.
Control of industrial noise and protectionhiten patel
Industrial noise can be controlled through several methods: 1) Reducing noise at its source by repairing old machines or adding vibration dampers. 2) Enclosing noisy machines in isolated rooms with sound absorbing materials. 3) Absorbing remaining noise using sound absorbing walls, high ceilings, and empty space near machines. 4) As a last resort, workers should wear protective ear equipment like plugs or muffs when near high noise machines, as excessive noise can cause health issues.
Noise pollution is the most targeted disease causing factor in today's busy life. Physiological, psychological, behavioural effects are described in the presentation.
Active Noise Control in an Automobile for Road NoiseJohn Long
This document summarizes an on-road demonstration of active noise control technology in two passenger vehicles. The demonstration achieved substantial attenuation of low-frequency road noise, with up to 10 dB of narrowband attenuation. The active noise control system was able to transfer effectively between two vehicles of the same make but different sound packages. The system used two parallel controllers to target different spectral content. Initial field trials showed the technology was viable despite some pragmatic concerns.
This document discusses sources of traffic noise and methods to reduce it. It explains that tire-pavement noise becomes the dominant source at higher speeds. Various pavement materials and designs are discussed that can lower noise, such as porous asphalt, gap-graded stone matrix, and asphalt rubber friction courses. Noise barriers are also described as an effective way to reduce noise levels reaching nearby areas. The conclusion emphasizes the importance of implementing quieter pavement techniques to achieve safer and quieter highways.
Noise pollution can negatively impact human health and the environment in several ways. It is defined as unwanted and disturbing sounds that disrupt normal activities or cause harm. Common sources include vehicles, construction equipment, and industrial operations. Effects range from temporary hearing loss and sleep disruption to long-term cardiovascular problems. Reducing noise pollution involves identifying sound sources, modifying noisy machinery, adding sound barriers, and providing protective equipment for workers.
The document discusses the exhaust system of vehicles. It describes the main parts of an exhaust system which include the exhaust manifold, catalytic converter, oxygen sensor, exhaust pipe, and muffler. It explains how mufflers work to reduce noise by using baffles, wave cancellation, resonators, and absorbing materials. Mufflers come in different types like baffle, wave cancellation, resonance, absorber, and combined resonance types. The document also discusses how mufflers and exhaust pipes work to reduce noise produced from engine exhaust gases.
This document describes an Aqua Silencer device that is fitted to vehicle exhaust pipes to reduce noise and air pollution. It consists of a perforated tube inside a water container, surrounded by activated charcoal. Exhaust gases enter the tube and are broken into smaller bubbles before passing through the charcoal. Pollutants like CO, NOx, and SO2 can be dissolved in the water or absorbed by the charcoal. The treated exhaust gases then exit through an opening while treated water is drained periodically. The Aqua Silencer provides pollution control through bubbling the exhaust in water and using charcoal absorption, helping reduce harmful emissions in a cheap and effective way.
Design and implementation of different audio restoration techniques for audio...eSAT Journals
This document summarizes research on designing and implementing different audio restoration techniques for removing distortions like clipping, clicks, and broadband noise from audio signals. It presents methods for declipping audio using sparse representations and frame-based reconstruction. Clicks are addressed using an adaptive filtering method, and broadband noise is reduced via spectral subtraction. The performance of these techniques is evaluated using metrics like SNR and algorithms like OMP. Hardware implementation of click removal is done on a TMS320C6713 DSK board using tools like MATLAB and Code Composer Studio.
Rolling Element Bearing Condition Monitoring using Filtered Acoustic Emission IJECEIAES
This document summarizes research on using filtered acoustic emission signals to monitor the condition of rolling element bearings. The researchers collected acoustic emission data from both healthy and defective bearings. They applied three active noise cancellation techniques (LMS, EMD, wavelet) to filter the noisy acoustic signals and compared their performance based on SNR and MSE, finding that EMD provided the best filtering. Time, frequency, and time-frequency analyses were then used to analyze the filtered signals and diagnose bearing faults. The analyses clearly showed differences between healthy and defective bearings and could detect different types of defects. The research demonstrates that acoustic emission monitoring combined with noise filtering is effective for rolling element bearing condition monitoring and fault diagnosis.
Surface Electromyography (SEMG) Based Fuzzy Logic Controller for Footballer b...IRJET Journal
This document describes a study that uses surface electromyography (SEMG) signals from the gastrocnemius muscle and a fuzzy logic controller to classify different football player foot actions (dorsiflexion, plantarflexion, no action). SEMG signals were collected using electrodes and processing hardware, then analyzed in MATLAB. Three statistical parameters (root mean square, median, standard deviation) of the SEMG signals were calculated and used as inputs for the fuzzy logic controller. Results showed that standard deviation was the best parameter for distinguishing between the different foot actions. The system has applications in human-computer interfaces that recognize different foot motions.
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.
Our study is demonstrated a new type of evolutionary sound synthesis method. This work based on the fly algorithm, a cooperative co-evolution algorithm; it is derived from the Parisian evolution approach. The algorithm has relatively amended the position of individuals (the Flies) represented by 3-D points. The fly algorithm has successfully investigated in different applications, starting with a real-time stereo vision for robotics. Also, the algorithm shows promising results in tomography to reconstruct 3-D images. The final application of the fly algorithm was generating artistic images, such as digital mosaics. In all these applications, the flies’representation started for simple, 3-D points, to complex one, the structure of 9-elements. Our method follows evolutionary digital art with the fly algorithm in representing the pattern of the flies. They represented in a way of having their structure. This structure includes position, colour, rotation angle, and size. Our algorithm has the benefit of graphics processing units (GPUs) to generate the sound waveform using the modern OpenGL shading language.
Describe The Main Functions Of Each Layer In The Osi Model...Amanda Brady
Tone injection is a technique used to increase the constellation size of a signal constellation. It works by mapping each point in the original constellation to multiple equivalent points in an expanded constellation. This allows for embedding additional information by substituting points, improving spectral efficiency. However, it also increases implementation complexity and may degrade performance due to increased decision regions. Tone injection is useful for applications requiring high data rates within bandwidth constraints.
DESIGN REALIZATION AND PERFORMANCE EVALUATION OF AN ACOUSTIC ECHO CANCELLATIO...sipij
Nowadays, in the field of communications, AEC (acoustic echo cancellation) is truly essential with respect
to the quality of multimedia transmission. In this paper, we designed and developed an efficient AEC based
on adaptive filters to improve quality of service in telecommunications against the phenomena of acoustic
echo, which is indeed a problem in hands-free communications.The main advantage of the proposed algorithm is its capacity of tracking non-stationary signals such as acoustic echo. In this work the acoustic echo cancellation (AEC) is modeled using a digital signal
processing technique especially Simulink Blocksets. The algorithm’s code is generated in Matlab Simulink
programming environment. At simulation level, results of simulink implementation prove that module
behavior is realistic when it comes to cancellation of echo in hands free communication using adaptive algorithm.Results obtained with our algorithm in terms of ERLE criteria are confronted to IUT-T recommendation
G.168.
IRJET- A Review on Audible Sound Analysis based on State Clustering throu...IRJET Journal
This document reviews progress in acoustic modeling for statistical parametric speech synthesis, from early hidden Markov models to recent neural network approaches. It discusses how hidden Markov models were previously dominant but artificial neural networks are now replacing them due to improvements in naturalness. The document also examines developing accurate audio classifiers using machine learning techniques on public datasets and improving classification accuracy beyond current levels of 50-79% by employing strategies like cross-validation.
Acoustic Scene Classification by using Combination of MODWPT and Spectral Fea...ijtsrd
Acoustic Scene Classification ASC is classified audio signals to imply about the context of the recorded environment. Audio scene includes a mixture of background sound and a variety of sound events. In this paper, we present the combination of maximal overlap wavelet packet transform MODWPT level 5 and six sets of time domain and frequency domain features are energy entropy, short time energy, spectral roll off, spectral centroid, spectral flux and zero crossing rate over statistic values average and standard deviation. We used DCASE Challenge 2016 dataset to show the properties of machine learning classifiers. There are several classifiers to address the ASC task. We compare the properties of different classifiers K nearest neighbors KNN , Support Vector Machine SVM , and Ensembles Bagged Trees by using combining wavelet and spectral features. The best of classification methodology and feature extraction are essential for ASC task. In this system, we extract at level 5, MODWPT energy 32, relative energy 32 and statistic values 6 from the audio signal and then extracted feature is applied in different classifiers. Mie Mie Oo | Lwin Lwin Oo "Acoustic Scene Classification by using Combination of MODWPT and Spectral Features" 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/ijtsrd27992.pdfPaper URL: https://www.ijtsrd.com/computer-science/multimedia/27992/acoustic-scene-classification-by-using-combination-of-modwpt-and-spectral-features/mie-mie-oo
A GAUSSIAN MIXTURE MODEL BASED SPEECH RECOGNITION SYSTEM USING MATLABsipij
1) The document describes the development of a speaker-dependent speech recognition system using MATLAB. It uses Gaussian mixture models for acoustic modeling and mel-frequency cepstral coefficients for feature extraction.
2) The system is designed to recognize isolated digits 0-9. Voice activity detection is performed to detect segments of speech. Various windowing functions are evaluated to reduce spectral leakage during feature extraction.
3) 13 MFCCs plus energy are extracted from each 30ms frame with 10ms shift. The first and second derivatives are also calculated to capture dynamic information, resulting in a 39-dimensional feature vector.
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
This document summarizes a research paper that proposes using genetic algorithms and discrete wavelet transforms to recognize isolated Arabic words from speech. The paper extracts features from speech signals using 3rd and 4th level Haar wavelet transforms and magnitudes. These features are divided into blocks and classified using a genetic algorithm. The genetic algorithm is initialized with reference files as the population. Binary encoding represents the magnitude of each block. Testing achieved 90% recognition at the 3rd level and 87.5% at the 4th level.
A Combined Voice Activity Detector Based On Singular Value Decomposition and ...CSCJournals
voice activity detector (VAD) is used to separate the speech data included parts from silence parts of the signal. In this paper a new VAD algorithm is represented on the basis of singular value decomposition. There are two sections to perform the feature vector extraction. In first section voiced frames are separated from unvoiced and silence frames. In second section unvoiced frames are silence frames. To perform the above sections, first, windowing the noisy signal then Hankel’s matrix is formed for each frame. The basis of statistical feature extraction of purposed system is slope of singular value curve related to each frame by using linear regression. It is shown that the slope of singular values curve per different SNRs in voiced frames is more than the other types and this property can be to achieve the goal the first part can be used. High similarity between feature vector of unvoiced and silence frame caused to approach for separation of the two categories above cannot be used. So in the second part, the frequency characteristics for identification of unvoiced frames from silent frames have been used. Simulation results show that high speed and accuracy are the advantages of the proposed system.
This document discusses acoustic echo cancellation (AEC) systems using artificial neural network algorithms. It begins with background on AEC and issues with nonlinear echo paths. It then presents an AEC system using an artificial neural network combined with an adaptive filter model to address nonlinear environments. Simulation results on Matlab demonstrate that the proposed neural network approach combined with a Laguerre filter achieves lower error and higher echo return loss than adaptive filter-only methods, showing it effectively reduces echo signals in linear and nonlinear systems. The paper concludes the combined neural network-filter algorithm is a promising approach for acoustic echo cancellation.
This document discusses various types of pollution including noise, air, water, soil, and radioactive pollution. For each type of pollution, it provides definitions, causes, effects on human health and environment, and methods for control and prevention.
Control of industrial noise and protectionhiten patel
Industrial noise can be controlled through several methods: 1) Reducing noise at its source by repairing old machines or adding vibration dampers. 2) Enclosing noisy machines in isolated rooms with sound absorbing materials. 3) Absorbing remaining noise using sound absorbing walls, high ceilings, and empty space near machines. 4) As a last resort, workers should wear protective ear equipment like plugs or muffs when near high noise machines, as excessive noise can cause health issues.
Noise pollution is the most targeted disease causing factor in today's busy life. Physiological, psychological, behavioural effects are described in the presentation.
Active Noise Control in an Automobile for Road NoiseJohn Long
This document summarizes an on-road demonstration of active noise control technology in two passenger vehicles. The demonstration achieved substantial attenuation of low-frequency road noise, with up to 10 dB of narrowband attenuation. The active noise control system was able to transfer effectively between two vehicles of the same make but different sound packages. The system used two parallel controllers to target different spectral content. Initial field trials showed the technology was viable despite some pragmatic concerns.
This document discusses sources of traffic noise and methods to reduce it. It explains that tire-pavement noise becomes the dominant source at higher speeds. Various pavement materials and designs are discussed that can lower noise, such as porous asphalt, gap-graded stone matrix, and asphalt rubber friction courses. Noise barriers are also described as an effective way to reduce noise levels reaching nearby areas. The conclusion emphasizes the importance of implementing quieter pavement techniques to achieve safer and quieter highways.
Noise pollution can negatively impact human health and the environment in several ways. It is defined as unwanted and disturbing sounds that disrupt normal activities or cause harm. Common sources include vehicles, construction equipment, and industrial operations. Effects range from temporary hearing loss and sleep disruption to long-term cardiovascular problems. Reducing noise pollution involves identifying sound sources, modifying noisy machinery, adding sound barriers, and providing protective equipment for workers.
The document discusses the exhaust system of vehicles. It describes the main parts of an exhaust system which include the exhaust manifold, catalytic converter, oxygen sensor, exhaust pipe, and muffler. It explains how mufflers work to reduce noise by using baffles, wave cancellation, resonators, and absorbing materials. Mufflers come in different types like baffle, wave cancellation, resonance, absorber, and combined resonance types. The document also discusses how mufflers and exhaust pipes work to reduce noise produced from engine exhaust gases.
This document describes an Aqua Silencer device that is fitted to vehicle exhaust pipes to reduce noise and air pollution. It consists of a perforated tube inside a water container, surrounded by activated charcoal. Exhaust gases enter the tube and are broken into smaller bubbles before passing through the charcoal. Pollutants like CO, NOx, and SO2 can be dissolved in the water or absorbed by the charcoal. The treated exhaust gases then exit through an opening while treated water is drained periodically. The Aqua Silencer provides pollution control through bubbling the exhaust in water and using charcoal absorption, helping reduce harmful emissions in a cheap and effective way.
Design and implementation of different audio restoration techniques for audio...eSAT Journals
This document summarizes research on designing and implementing different audio restoration techniques for removing distortions like clipping, clicks, and broadband noise from audio signals. It presents methods for declipping audio using sparse representations and frame-based reconstruction. Clicks are addressed using an adaptive filtering method, and broadband noise is reduced via spectral subtraction. The performance of these techniques is evaluated using metrics like SNR and algorithms like OMP. Hardware implementation of click removal is done on a TMS320C6713 DSK board using tools like MATLAB and Code Composer Studio.
Rolling Element Bearing Condition Monitoring using Filtered Acoustic Emission IJECEIAES
This document summarizes research on using filtered acoustic emission signals to monitor the condition of rolling element bearings. The researchers collected acoustic emission data from both healthy and defective bearings. They applied three active noise cancellation techniques (LMS, EMD, wavelet) to filter the noisy acoustic signals and compared their performance based on SNR and MSE, finding that EMD provided the best filtering. Time, frequency, and time-frequency analyses were then used to analyze the filtered signals and diagnose bearing faults. The analyses clearly showed differences between healthy and defective bearings and could detect different types of defects. The research demonstrates that acoustic emission monitoring combined with noise filtering is effective for rolling element bearing condition monitoring and fault diagnosis.
Surface Electromyography (SEMG) Based Fuzzy Logic Controller for Footballer b...IRJET Journal
This document describes a study that uses surface electromyography (SEMG) signals from the gastrocnemius muscle and a fuzzy logic controller to classify different football player foot actions (dorsiflexion, plantarflexion, no action). SEMG signals were collected using electrodes and processing hardware, then analyzed in MATLAB. Three statistical parameters (root mean square, median, standard deviation) of the SEMG signals were calculated and used as inputs for the fuzzy logic controller. Results showed that standard deviation was the best parameter for distinguishing between the different foot actions. The system has applications in human-computer interfaces that recognize different foot motions.
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.
Our study is demonstrated a new type of evolutionary sound synthesis method. This work based on the fly algorithm, a cooperative co-evolution algorithm; it is derived from the Parisian evolution approach. The algorithm has relatively amended the position of individuals (the Flies) represented by 3-D points. The fly algorithm has successfully investigated in different applications, starting with a real-time stereo vision for robotics. Also, the algorithm shows promising results in tomography to reconstruct 3-D images. The final application of the fly algorithm was generating artistic images, such as digital mosaics. In all these applications, the flies’representation started for simple, 3-D points, to complex one, the structure of 9-elements. Our method follows evolutionary digital art with the fly algorithm in representing the pattern of the flies. They represented in a way of having their structure. This structure includes position, colour, rotation angle, and size. Our algorithm has the benefit of graphics processing units (GPUs) to generate the sound waveform using the modern OpenGL shading language.
Describe The Main Functions Of Each Layer In The Osi Model...Amanda Brady
Tone injection is a technique used to increase the constellation size of a signal constellation. It works by mapping each point in the original constellation to multiple equivalent points in an expanded constellation. This allows for embedding additional information by substituting points, improving spectral efficiency. However, it also increases implementation complexity and may degrade performance due to increased decision regions. Tone injection is useful for applications requiring high data rates within bandwidth constraints.
DESIGN REALIZATION AND PERFORMANCE EVALUATION OF AN ACOUSTIC ECHO CANCELLATIO...sipij
Nowadays, in the field of communications, AEC (acoustic echo cancellation) is truly essential with respect
to the quality of multimedia transmission. In this paper, we designed and developed an efficient AEC based
on adaptive filters to improve quality of service in telecommunications against the phenomena of acoustic
echo, which is indeed a problem in hands-free communications.The main advantage of the proposed algorithm is its capacity of tracking non-stationary signals such as acoustic echo. In this work the acoustic echo cancellation (AEC) is modeled using a digital signal
processing technique especially Simulink Blocksets. The algorithm’s code is generated in Matlab Simulink
programming environment. At simulation level, results of simulink implementation prove that module
behavior is realistic when it comes to cancellation of echo in hands free communication using adaptive algorithm.Results obtained with our algorithm in terms of ERLE criteria are confronted to IUT-T recommendation
G.168.
IRJET- A Review on Audible Sound Analysis based on State Clustering throu...IRJET Journal
This document reviews progress in acoustic modeling for statistical parametric speech synthesis, from early hidden Markov models to recent neural network approaches. It discusses how hidden Markov models were previously dominant but artificial neural networks are now replacing them due to improvements in naturalness. The document also examines developing accurate audio classifiers using machine learning techniques on public datasets and improving classification accuracy beyond current levels of 50-79% by employing strategies like cross-validation.
Acoustic Scene Classification by using Combination of MODWPT and Spectral Fea...ijtsrd
Acoustic Scene Classification ASC is classified audio signals to imply about the context of the recorded environment. Audio scene includes a mixture of background sound and a variety of sound events. In this paper, we present the combination of maximal overlap wavelet packet transform MODWPT level 5 and six sets of time domain and frequency domain features are energy entropy, short time energy, spectral roll off, spectral centroid, spectral flux and zero crossing rate over statistic values average and standard deviation. We used DCASE Challenge 2016 dataset to show the properties of machine learning classifiers. There are several classifiers to address the ASC task. We compare the properties of different classifiers K nearest neighbors KNN , Support Vector Machine SVM , and Ensembles Bagged Trees by using combining wavelet and spectral features. The best of classification methodology and feature extraction are essential for ASC task. In this system, we extract at level 5, MODWPT energy 32, relative energy 32 and statistic values 6 from the audio signal and then extracted feature is applied in different classifiers. Mie Mie Oo | Lwin Lwin Oo "Acoustic Scene Classification by using Combination of MODWPT and Spectral Features" 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/ijtsrd27992.pdfPaper URL: https://www.ijtsrd.com/computer-science/multimedia/27992/acoustic-scene-classification-by-using-combination-of-modwpt-and-spectral-features/mie-mie-oo
A GAUSSIAN MIXTURE MODEL BASED SPEECH RECOGNITION SYSTEM USING MATLABsipij
1) The document describes the development of a speaker-dependent speech recognition system using MATLAB. It uses Gaussian mixture models for acoustic modeling and mel-frequency cepstral coefficients for feature extraction.
2) The system is designed to recognize isolated digits 0-9. Voice activity detection is performed to detect segments of speech. Various windowing functions are evaluated to reduce spectral leakage during feature extraction.
3) 13 MFCCs plus energy are extracted from each 30ms frame with 10ms shift. The first and second derivatives are also calculated to capture dynamic information, resulting in a 39-dimensional feature vector.
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
This document summarizes a research paper that proposes using genetic algorithms and discrete wavelet transforms to recognize isolated Arabic words from speech. The paper extracts features from speech signals using 3rd and 4th level Haar wavelet transforms and magnitudes. These features are divided into blocks and classified using a genetic algorithm. The genetic algorithm is initialized with reference files as the population. Binary encoding represents the magnitude of each block. Testing achieved 90% recognition at the 3rd level and 87.5% at the 4th level.
A Combined Voice Activity Detector Based On Singular Value Decomposition and ...CSCJournals
voice activity detector (VAD) is used to separate the speech data included parts from silence parts of the signal. In this paper a new VAD algorithm is represented on the basis of singular value decomposition. There are two sections to perform the feature vector extraction. In first section voiced frames are separated from unvoiced and silence frames. In second section unvoiced frames are silence frames. To perform the above sections, first, windowing the noisy signal then Hankel’s matrix is formed for each frame. The basis of statistical feature extraction of purposed system is slope of singular value curve related to each frame by using linear regression. It is shown that the slope of singular values curve per different SNRs in voiced frames is more than the other types and this property can be to achieve the goal the first part can be used. High similarity between feature vector of unvoiced and silence frame caused to approach for separation of the two categories above cannot be used. So in the second part, the frequency characteristics for identification of unvoiced frames from silent frames have been used. Simulation results show that high speed and accuracy are the advantages of the proposed system.
This document discusses acoustic echo cancellation (AEC) systems using artificial neural network algorithms. It begins with background on AEC and issues with nonlinear echo paths. It then presents an AEC system using an artificial neural network combined with an adaptive filter model to address nonlinear environments. Simulation results on Matlab demonstrate that the proposed neural network approach combined with a Laguerre filter achieves lower error and higher echo return loss than adaptive filter-only methods, showing it effectively reduces echo signals in linear and nonlinear systems. The paper concludes the combined neural network-filter algorithm is a promising approach for acoustic echo cancellation.
Enhanced modulation spectral subtraction for IOVT speech recognition applicationIRJET Journal
This document proposes an Enhanced Modulation Spectral Subtraction (EMSS) method for enhancing noisy speech signals to improve speech recognition performance in Internet of Vehicle Things (IOVT) systems. The EMSS method uses a two-stage analysis-modification-synthesis framework, applying short-time Fourier transform first at the acoustic level and then again at the modulation level to extract modulation spectra. At the modulation level, noise is estimated and subtracted from the noisy modulation spectra. Evaluation on speech emotion recognition tasks with different noise types shows the EMSS method improves the signal-to-noise ratio segmentation by 39-60% compared to traditional spectral subtraction and modulation spectral subtraction methods, achieving about a 50% better recognition performance.
Classification of vehicles based on audio signalsijsc
The document describes a method for classifying vehicles based on their audio signals using quadratic discriminant analysis. Feature vectors containing short-time energy, average zero-crossing rate, and pitch frequency are extracted from periodic segments of vehicle audio signals. The method achieves better classification accuracy by only considering feature vectors with high energy, as these correspond better to the vehicle sounds and exclude low energy background noise regions. Simulation results show the proposed method of separating high energy vectors based on thresholds of average energy and zero-crossing rate improves classification performance compared to considering all vectors.
Classification of Vehicles Based on Audio Signals using Quadratic Discriminan...ijsc
The focusof this paper is on classification of different vehicles using sound emanated from the vehicles. In this paper,quadratic discriminant analysis classifies audio signals of passing vehicles to bus, car, motor, and truck categories based on features such as short time energy, average zero cross rate, and pitch frequency of periodic segments of signals. Simulation results show that just by considering high energy feature vectors, better classification accuracy can be achieved due to the correspondence of low energy regions with noises of the background. To separate these elements, short time energy and average zero cross rate are used simultaneously.In our method, we have used a few features which are easy to be calculated in time domain and enable practical implementation of efficient classifier. Although, the computation complexity is low, the classification accuracy is comparable with other classification methods based on long feature vectors reported in literature for this problem.
This document discusses techniques for automatic speech recognition that incorporate modulation domain enhancement. It begins by introducing the topic of speech enhancement to reduce background noise and improve speech recognition. It then describes several existing noise reduction techniques like Wiener filtering and wavelet-based methods. The document focuses on using spectral subtraction to reduce noise by measuring and subtracting the noise spectrum from the noisy speech spectrum. It presents the workflow which involves segmenting speech into frames, applying FFT to extract magnitude and phase, estimating noise spectrum, and subtracting it from noisy speech. Simulation results on emotion recognition accuracy using KNN filtering, FFT, and mel-frequency cepstral coefficients are shown. The proposed method is found to significantly reduce low noise compared to other techniques.
This document discusses computational auditory scene recognition (CASR), which involves processing and classifying audio signals to understand the context or environment. The authors attempt to classify audio clips into 12 environments using frame-level sub-band energy ratios as features and a k-nearest neighbors classifier. They achieve an overall accuracy of 70% on the classification task. The document outlines the theory, methods, implementation details and results of the classification experiment.
IRJET - Essential Features Extraction from Aaroh and Avroh of Indian Clas...IRJET Journal
The document summarizes research on extracting essential features from the aaroh and avroh of the Indian classical raga Yaman. It discusses extracting various time domain, spectral, cepstral and other features using audio signal processing and machine learning techniques. Time domain features like zero crossing rate and autocorrelation are extracted. Spectral features including the spectrogram, mel-spectrogram, spectral centroid, roll-off and others are analyzed. Cepstral features such as MFCC are also discussed. The goal is to extract relevant features that can be used for audio classification, recognition and other tasks related to Indian classical music.
Similar to Vehicle Noise Pattern Recognition by Self-Organizing Maps (20)
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
How to Make a Field Mandatory in Odoo 17Celine George
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more evident. A crucial element of this impact is the alteration of vegetation cover, which plays a
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Accurate understanding of land use and cover is imperative for the development planning
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9
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Vehicle Noise Pattern Recognition by Self-Organizing Maps
1. Stefan Twieg, Holger Opfer & Helmut Beikirch
Signal Processing: An International Journal (SPIJ) Volume (3) : Issue (6) 180
Vehicle noise pattern recognition by Self-Organizing Maps
Stefan Twieg stefan.twieg@volkswagen.de
Volkswagen AG/Group Research
Wolfsburg, Germany
Holger Opfer opfer.holger@vgj.co.jp
Volkswagen Group Japan
VTT, Tokyo, Japan
Helmut Beikirch helmut.beikirch@uni-rostock.de
Faculty of Computer Science and
Electrical Engineering
University Rostock, Germany
Abstract
Interior vehicle acoustics are in close connection with our quality opinion. The
noise in vehicle interior is complex and can be considered as a sum of various
sound emission sources. A nice sounding vehicle is objective of the development
departments for car acoustics. In the process of manufacturing the targets for a
qualitatively high-valuable sound must be maintained. However, it is possible that
production errors lead to a deviation from the wanted vehicle interior sound. This
will result in customer complaints where for example a rattling or squeak refers to
a worn-out or defective component. Also in this case, of course, the vehicle
interior noise does not fulfill the targets of the process of development. For both
cases there is currently no possibility for automated analysis of the vehicle
interior noise. In this paper an approach for automated analysis of vehicle interior
noise by means of neural algorithms is investigated. The presented system
analyses microphone signals from car interior measured at real environmental
conditions. This is in contrast to well known techniques, as e.g. acoustic engine
test bench. Self-Organizing Maps combined with feature selection algorithms are
used for acoustic pattern recognition. The presented system can be used in
production process as well as a standalone ECU in car.
Keywords: signal classification, SOM, car noise, pattern recognition.
1. MOTIVATION
Car interior noise is a sum of different sound emission sources, e.g. engine, wind, chassis or
suspension. Furthermore, it is a challenging task to examine objective assessment criteria for a
sound. In production process acoustic and vibration technology is used e.g. for analysis at engine
test bench. Engines and test benches are tested in [1] to recognize damages of failure parts.
Suppliers are also checking their parts before delivery by similar methods [2], [3] to achieve a
reliable and correct product. Commonly, the acoustic analysis is done in defined environmental
conditions to get no interference with unknown noises. In this case quite simple mathematic is
2. Stefan Twieg, Holger Opfer & Helmut Beikirch
Signal Processing: An International Journal (SPIJ) Volume (3) : Issue (6) 181
relevant parameter
calculation
DATASETS
extraction of sequences
with good audibility
SOM learning
SOM categorization
mathematic parameters:
one- & multidimensional
SOUND CLASS
relevance estimation
FIGURE 1: Model-design for classification of acoustic in car recordings.
used to detect and to interpret changes. This method is in contrast to the method shown in this
paper where an undefined environment influences the acoustic conditions. For assessment of car
interior sound by algorithms the time signal which is recorded by microphone in car interior has to
be fragmented in different mathematic parameters. The challenge is to find parameters which are
corresponding to failures or changes of car part sounds and not to acoustic changes in
environment. Thus the parameters have to be explicit relevant for sounds of interest. To analyze
these relevant parameter knowledge storage is required to build up experience of parameter
behavior. Therefore a neural net by Kohonen [4], the self-organizing map (SOM), models an input
space where similar patterns can be categorized. For the described purpose the Kohonen-map is
combined with parameter relevance detection algorithm. As a result the developed system
assesses continuously the microphone signals, e.g. in two categories like “error-free” and “error”
sound. In future the described method can be used for applications of sound assessment at
production-line end (at test-bench or test-drive) or in car combined with hands-free kit.
2. MODEL-LAYOUT FOR PATTERN RECOGNITION
An adequate amount of data sets is the basis for acoustic pattern recognition. For this purpose
different noticeable sounds are stored in database. They were recorded at different conditions
like:
• Car type,
• Driving situation (longitudinal and transversal dynamic),
• Car parts (new vs. used and damaged),
• Environment (weather, pavement).
In the following analysis one car type was chosen. For this one 1430 measurements with a length
of 15 to 30 seconds are stored in database. In Figure 1, you can see the model for datasets
analysis and pattern recognition. As next step the one (e.g. rms, sound pressure level) or
multidimensional (e.g. spectrum, DCT, MFCC) parameters are calculated. However, psycho-
acoustic (e.g. loudness, roughness) parameters are calculated too. In the next step the focal point
is
relevance detection to identify parameters which are corresponding to noticeable sounds caused
by wear (e.g. steering rod ball joint) or damage. The relevance detection algorithm has to identify
the connection between:
• the sound classes (e.g. “error-free” and “error” sounds),
• sound classes and parameters.
Each parameter is rated and ordered by rank information. The parameters with best relevance
are used as input features for SOM. In the next step the datasets with best audibility of defined
sound classes are used for training process. There are two defined sound classes “error” and
3. Stefan Twieg, Holger Opfer & Helmut Beikirch
Signal Processing: An International Journal (SPIJ) Volume (3) : Issue (6) 182
“error-free” sound. For more detailed information it is also possible to define more sub-sound-
classes, e.g. different noise emitting parts. The sound class representing sequences are
extracted from database, whereas a huge variance of recording situations is very important. After
learning process the weights of the neural net are adopted. Now the SOM represents the input
space. In labeling process the sound class information is added. Thereafter it is possible to
classify unknown test datasets and to get a feedback of associated sound class.
In the following the described method is combined in a so-called SOM-Agent which will monitor
and assess microphone signals. The knowledge – the neural weights – of the input space and the
sound classes – from labeling process – are transferred to the agent. Relevant parameters are
calculated for the sampled audio data and will be used as input features for the SOM-Agent.
Identification of sound class active neurons will follow. In Figure 2 the created SOM-Agent
characteristic curve for one recording of 11 seconds is shown. The 2 defined sound classes and
an “unknown” class – in case of no active neuron – are represented by an output value in range 0
to 2 on y-axis. The x-axis represents time. Statistics are also computed for the three classes.
3. FUNDAMENTALS
As described in previous section the shown method consists of two main components. First one
is identification and calculation of sound class relevant parameters. For the second step these are
the input features for a self-organizing map which will categorize them in defined sound classes.
3.1 Relevance Identification
Several methods exist for analyzing and calculation of feature influence, feature relevance,
feature (subset) selection or feature extraction. Reduction of feature space to the essential is
main target of several classification tasks. Anyway, the connectivity between the parameters and
classes is determined. Therefore the correlation coefficient [5] – as linear method – or the
information gain [6], [7] – as non-linear method – can be used. In Equation 1 the calculation rule
for correlation coefficient, also known as normalized covariance is given. Thus, the linear
connection between the vectorial values x and y is estimated.
time in seconds
SOM-Agentoutput
/error-free
/error
/unknown
statistic:
time in seconds
SOM-Agentoutput
/error-free
/error
/unknown
statistic:
FIGURE 2: Autonomous detection of noticeable acoustic changes in car interior. The SOM-Agent
characteristic curve (blue) toggles between three sound classes “error-free”, “error” and “unknown”
sound, e.g. if output value is “1” the microphone recording was made in a car with damaged or
worn-out parts.
4. Stefan Twieg, Holger Opfer & Helmut Beikirch
Signal Processing: An International Journal (SPIJ) Volume (3) : Issue (6) 183
( )
( )( )
( ) ( ) ( ) ( )2222
,
∑∑∑∑
∑ ∑∑
−−
−
=
yynxxn
yxxyn
yxr (1)
On the other hand the information gain ( IG , Equation 2 at [6]) is calculated by entropy H and
represents a non-linear probabilistic method. The information gain returns the contribution of one
attribute to class decision making.
( ) ( ) ( )
( )
∑∈
−=
AValuesv
V
V
SH
S
S
SHSIG (2)
Whereas S is the set and A represents the class information of S . VS is the subset of the
selected class.
Further known filter methods (parameter selection by pre-processing) are for example: Mutual
Information, Fisher Discriminant, RELIEF, PCA, ICA, FCBF (Fast Correlation Based Filtering, [8],
[9]). As you can see in [8], [9] the FCBF algorithm solves the two main aspects of feature subset
selection. First one is in our case the decision whether mathematic parameter (feature) is relevant
or not for a acoustic condition (sound class). In second step the algorithm determines whether a
relevant parameter is redundant.
The symmetrical uncertainty (SU, Equation 3) is used for determination of nonlinear connection
between the two sources x and y . It is based on information gain but it is normalized on interval
[0,1].
( ) ( )
( ) ( )
+
=
yHxH
yxIG
yxSU
|
2, (3)
In correlation analysis for a dataset the relevance and redundancy are assessed by this formula.
If we are assuming that the values of one source contain same information like second source the
SU output will be 1. If SU output is 0 the sources are independent of each other. The symmetrical
uncertainty is used in FCBF algorithm as measure for correlation between two sources. In this
algorithm two main aspects are shown.
First one is the SU calculation for the feature-class-SU:
• Serves as measure for correlation between each feature and the class information.
• For each feature the SU to all classes is calculated and stored in a list.
• For this list a threshold determines if features SU is too low. A feature is deleted if
threshold is undercut.
• For each class the residual features are ordered in descending order and stored in a final
list.
In second process the redundancy examination occurs by feature-feature-SU.
• Serves as a measure for correlation between each combination of the listed features.
• For each feature the SU is calculated to all other features.
• Comparison of feature-feature-SU and feature-class-SU.
• Features with very high SU are deleted.
• The remaining features are representing the best features in dataset.
In most cases the efficiency of the algorithm is shown by a classifier. For this case the recognition
rate is the measure of assess. Anyway, as well known no algorithm will be the best or worst
solution for all datasets. In most cases algorithms are optimized for all-round use or for dataset.
As you can see in analysis (Figure 3, [9]) of UCI library [10] datasets the FCBF algorithm has best
performance for CoIL2000 (compared to three reduction methods). For Lung-Cancer and Splice
dataset on the contrary all methods have nearly same recognition rates.
5. Stefan Twieg, Holger Opfer & Helmut Beikirch
Signal Processing: An International Journal (SPIJ) Volume (3) : Issue (6) 184
3.2 Self-Organizing-Map
The neurons of a self organizing map [4] are represented by a multidimensional weight vector. If
the neural net is adopted the weight vectors are already adjusted (self organized without
supervise) to the input space (defined by the features/parameters). Field of application is the
recognition of clusters in high dimensional data. Each input node has a connection by weights to
each neuron of the map. One way of visualization is the component plane where the weights are
shown as contour or surface plot. However, more detailed information for SOM-visualization (e.g.
U-Matrix) methods is discussed in [11].
Representation of the n SOM-neurons nww ,,1 K is done by the weight vectors nww
r
K
r
,,1
. The
number of elements in w
r
is equal to the number m of input features mxx ,,1 K . The initialization
of the weights is done randomly or by prior input space knowledge. Adjustment and allocation of
neurons to the input space is made in different ways, e.g. as line, two- or three-dimensional map.
The adaptation is done by a learning rule with defined step size. The interconnected neurons are
adjusted by means of following equation:
( ) ( ) ( ) ( )( )nwnxnwnw kkk
rrrr
−⋅+=+ α1 (4)
In Equation 4 the calculation rule (from [4]) for adaptation of weight vectors kw
r
for selected
neuron k is shown. The step size α is chosen in interval 10 << α . There is no general rule for
determination of α . If batch learning is used the input vector x
r
with m features is randomly
selected out of dataset. If we have a closer look to the learning process the adjustment of the
weights results in a movement of the neuron in the direction of the training vector. The selection
of the trained neuron occurs by a distance function, e.g. Euclid distance. The neuron which has
lowest distance to input vector x
r
is adopted. However, the algorithm has to be updated in the
way that the connected neurons are covering the complete input space to get a map which is
topology preservative. For this purpose a neighborhood function ϕ defines a radius for the
selected winner neuron which has lowest distance to input vector. Thus, neurons inside the
radius were adopted by the input vector, too. Furthermore a distance function ( )ki wwd , allows a
division in different neighborhood-distances R for a weighted neuron adaptation. Symbolically
neurons which are closest to the winner neuron have the distance 1=R and neurons in next
radius have a distance of 2=R and so on. The degree of adaptation which defines the amount
of movement for the winning neuron in the direction of input vector x
r
depends on the
neighborhood function ( ) Rww ki ≤,ϕ with a range of values in [0,1]. The neurons iw with a small
distance will be adapted much more than neurons with a big distance. Also the design of the
neighborhood function can be varied depending on computational resources.
Figure 3: Comparison of recognition rates of FCBF, ReliefF, CFS-SF and FOCUS-SF algorithm. ([9],
“Acc records 10-fold cross-validation accuracy rate (%) and p-Val records the probability associated with
a paired two-tailed t-Test.”).
6. Stefan Twieg, Holger Opfer & Helmut Beikirch
Signal Processing: An International Journal (SPIJ) Volume (3) : Issue (6) 185
• Computationally efficient Stair-Neighborhood function
( ) ( )
( )
≤
=
sonst
Rwwd
wwdww
ki
kiki
;0
,;
,
1
,ϕ (5)
• Computationally intensive Gaussian-Neighborhood function
( )
( )
( )
≤=
−
sonst
Rwwdeww ki
wwd
ki
ki
;0
,;,
2
2
2
,
σ
ϕ (6)
In comparison Equation 6 realizes a more detailed and non-linear neighborhood-function than
Equation 5. Finally, after insertion in Equation 4 the learning rule is shown updated in Equation 7.
( ) ( ) ( ) ( ) ( )( )nwnxwwnwnw ikiii
rrrr
−⋅⋅+=+ ,1 ϕα (7)
Commonly the adaptation process is done by a defined amount of training cycles. The adaptation
process shown in Equation 4 and 7 can be optimized by a fast adjustment at the beginning and a
slow one at the end of training. This behavior can be realized by a time-varying step-size ( )tα .
( )
−=
ET
t
t 1αα (8)
By means of the fixed ratio between actual time t to the number of cycles ET in Equation 8 a
linear function is given. Non-linear step-size functions are also possible alternatives. Also it is
possible to modify the neighborhood functions (Equation 5 and 6) with a time descending factor.
For interpretation of the self-organizing map two main aspects were analyzed:
• Characteristics of SOM-neuron-weights in dependence of input vector.
• Class assignment (sound classes) of neurons with a maximum reaction (smallest
distance) at defined stimulation (a.k.a. labeling process).
4. METHOD DESCRIPTION AND EVALUATION
As a basis for the training of the SOM-Agent (see Figure 2) a noise database with different
measurements is used from which the mathematical properties are calculated. For training, 11
different noise types were defined which come from worn-out coupling rods, wheel bearings,
suspension strut mountings and other running gear components from the front of a defined motor
vehicle type. Noises which originate from the normal sound of a series vehicle without worn-out
components represent a further noise type. In sum there are 129 measurements - with a length
between 10 and 30 seconds – for which finally two training classes are defined:
“C1”: error
“C2”: error-free
The time constant for the window length for which the parameters are calculated and averaged is
ms250 . For each of the 129 datasets several parameter-sets are calculated and marked as class
one or two. For the dimension reduction and relevance determination for the features the FCBF-
algorithm was selected. From the 17 available parameters the parameters shown in Table 1 are
determined as relevant by the FCBF-algorithm described in section C1. The training process runs
T cycles, in which a randomly selected parameter-set is presented to the network. In next step
all measurements are used for evaluation by SOM-Agent. Therefore all 129 acoustic records are
7. Stefan Twieg, Holger Opfer & Helmut Beikirch
Signal Processing: An International Journal (SPIJ) Volume (3) : Issue (6) 186
presented one after the other and the characteristic curve shown in figure 2 is calculated. The
parameters must be estimated for the acoustic signals continuously. Thus, these time continuous
parameters equal time variable data rows where each column set is an input vector of self-
organizing map. Through determination of the active neurons in dependence of the class the
acoustic time signals can be assessed and equipped with class information (“C1” or “C2”). For
reliable condition recognition a buffer (e.g. ring-buffer with 12000 samples) is filled with these
data rows and is analyzed periodically. For this purpose a subset of the stored data in buffer is
used to estimate the class. Depending on distribution of active neurons the following output is
calculated:
• SOM-Agent output Ns ∈ in interval ]2,1,0[=s , where
- 1=s corresponds with error-noise class “C1” and
- 2=s corresponds with error-free-noise class “C2”.
• SOM-output 0=s if input data didn’t activate any neuron. That’s why Zero corresponds
to an unknown state “U”.
For visualization a low pass filtered curve is generated from signal s :
( ) ( ) ( )ns
L
ns
L
ns TPLP ⋅+−
−=
1
1
1
1 (9)
A statistical evaluation is done afterwards either for the values of s or LPs . In the case of strong
fluctuating curves of s the curve of LPs will generate an averaged value over the window length
characterized by L . However diffuse values (e.g. 5.1=s ) are not used for statistic analysis, that’s
why thresholds are implemented (have a look at colored areas in Figure 2). These thresholds are
optimized for a clear assessment between two classes. The probability of occurrence for the
classes (in the case of use of LPs within the threshold regions) is estimated over the length of
time signals. In the legend of Figure 2 the probability of occurrence is shown for the analyzed
record. As a result class “C1” was determined by SOM-Agent for 70% of analyzed time.
For training process and parameterization of self-organizing map the obviously noticeable
sequences are extracted out of 129 measurements. Hence a dataset with 2456 patterns with 17
parameters is generated. The 6 most relevant parameters were presented to 2 different self-
organizing maps with 169 or 1000 neurons and 6 inputs. The Gaussian function described in
Equation 6 was used as a neighborhood function. Neurons within the neighborhood radius R
(defines the width of the bell curve) are adopted by following rule (out of Equation 7, 8, 9):
( ) ( )
( )
( ) ( )( )nwnxe
T
t
nwnw i
R
wwd
E
ii
ki
rrrr
−⋅⋅
−+=+
−
2
,
11 α (10)
As distance function, the Euclidean distance is used:
Roughness (Time)
Speed
Sharpness (Time)
Spectral Centroid
Yaw Rate
Loudness (Time)
TABLE 1: The 6 most relevant parameters are estimated by means of FCBF-algorithm.
8. Stefan Twieg, Holger Opfer & Helmut Beikirch
Signal Processing: An International Journal (SPIJ) Volume (3) : Issue (6) 187
( ) ( )
2
1
, ∑=
−=−=
N
x
kxixkiki wwwwwwd (11)
In figure 4 the evaluation of the recognition rates is shown for the SOM1-agent (169 neurons). On
x-axis the acoustic error conditions (class "C1") in the range 1 to 11 are reproduced. Number 12
is the acoustic original state without any worn-out or damaged part (class "C2"). On y-axis the
averaged probabilities of occurrence are reproduced.
The results of SOM1-Agent (Figure 4) show that primarily the noise classes 1-8 are assessed
much better than 9-11. Just in the case of noise type 9 many measurements are classified as
“unknown”. Also many of the original sound records (class 12) are assessed as unknown
condition (some records are assessed for more than 70% as unknown). This behavior shows that
no neuron for one of the defined classes “error” or “error-free” was activated. This bad
performance could be caused by too little records for this noise type. Anyway, the noise of a
worn-out drive shaft is only noticeable for a few situations when the car is accelerated or high
steering angle at low speed occurs. This is in contrast to steering rods where noticeable noise is
emitted in many driving situations. Definitely all records without any noticeable noise are not
considered for the database but there are still records where the noise occur e.g. for only one
time in 10 seconds. However, for this case the SOM-Agent shall assess the moment where noise
is noticeable as “error” class and all other as “error-free”.
acoustic pattern recognition for microphone records in car
[SOM: n=169 T=20000 M=6]
0
10
20
30
40
50
60
70
80
90
100
0 1 2 3 4 5 6 7 8 9 10 11 12 13
1-11 noise classes / 12 original car sound class
probabilityofoccurrence[%]
C1/error C2/error-free unknown
Figure 4: SOM1-Agent statistic: A calculated probability of occurrence for 129 measurements, SOM with
169 neurons and 6 parameters is used. Legend defines whether the analyzed measurements are
assessed as “error”, “error-free” or “unknown” condition.
9. Stefan Twieg, Holger Opfer & Helmut Beikirch
Signal Processing: An International Journal (SPIJ) Volume (3) : Issue (6) 188
In comparison the SOM2-Agent (Figure 5) assesses much less time domain data (from 129
measurements) as “unknown”. Just the original sound records are classified to more than 50% of
the time for the correct category “C2” (“error-free”). This is a result of the increased amount of
neurons (1000), the corresponding expansion of the input space and the more detailed
information of the trained input space. In contrast to SOM1-Agent, 6 times more neurons are
used.
5. CONSLUSION & FUTURE WORK
The SOM2-Agent produces a reliable statement of the matching category for the analyzed 129
acoustic in-car records. In comparison to SOM1-Agent (Figure 4 and 5) much less time-parts are
declared as “unknown”. All car interior time signals of an error-free car are classified for more
than 50% of time to the right category “C2”. The probability of occurrence of the other two
categories is near less than 10%. Anyway, noises of worn-out components are assessed at least
for more than 40% of analyzed time to the right category/class. So it is important to make an
decision by means of the relations between all statistic outputs “C1”, “C2” and “U”. For example, if
class C1 [%] C2 [%] U [%]
1 95,0 0,3 2,4
2 94,6 1,3 1,6
3 90,9 1,1 2,1
4 92,0 1,3 4,4
5 87,0 2,9 3,6
6 88,7 1,7 3,7
7 93,4 0,6 2,6
8 83,5 3,4 2,4
9 69,5 4,6 4,8
10 67,6 2,8 3,0
11 83,0 3,3 3,8
12 1,8 87,3 3,4
Table 2: Averaged probabilities of occurrence (SOM2-Agent) for 129 measurements and for different
error noise (class 1-11) and error-free (class 12) records.
acoustic pattern recognition for microphone records in car
[SOM-Agent: n=1000 T=20000 M=6]
0
10
20
30
40
50
60
70
80
90
100
0 1 2 3 4 5 6 7 8 9 10 11 12 13
1-11 noise classes / 12 original car sound class
probabilityofoccurrence[%]
C1/error C2/error-free unknown
Figure 5: SOM2-Agent statistic: A calculated probability of occurrence for 129 measurements, SOM with
1000 neurons and 6 parameters is used. Legend defines whether the analyzed measurements are
assessed as “error”, “error-free” or “unknown” condition.
10. Stefan Twieg, Holger Opfer & Helmut Beikirch
Signal Processing: An International Journal (SPIJ) Volume (3) : Issue (6) 189
“C2” is much higher than “C1” it is likely that the analyzed signal contains no information for a
worn-out part.
In Table 2 all averaged probabilities of occurrence of SOM2-Agent are listed. The noise types of
the class 9 and 10 are recognized worst. All other noise types emitted from defect or worn-out
components are recognized as a “error”-sound (“C1”) correctly via at least 83 % of the analyzed
time. The “error-free”-sound (“C2”) - reproduced through class 12 - is evaluated above 87 % of
the analyzed time correctly by the agent. However, the average shows one more time that the
introduced method works well but nevertheless further improvements will be focused on the
outliers. The presented method seems to be good for pattern recognition in acoustic records due
to combination of traditional pattern recognition, signal processing and statistic analyze. However,
it is understood that the results will be proof for the analyzed environment conditions (e.g. car
type, microphone type, measurement procedure). Furthermore, the analysis of the influence of
different car types on the features and pattern recognition is necessary in further studies. It was
shown that the number of the neurons has a decisive influence on the accuracy and reliability of
the SOM-agent. It has to be examined also by means of further measurements how far the
number of neurons, adaptation accuracy and over-fitting influence the training and assessment
process.
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