31 research outputs found

    A Fast and Accurate Pitch Estimation Algorithm Based on the Pseudo Wigner-Ville Distribution

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    Estimation of fundamental frequency (F0) in voiced segments of speech signals, also known as pitch tracking, plays a crucial role in pitch synchronous speech analysis, speech synthesis, and speech manipulation. In this paper, we capitalize on the high time and frequency resolution of the pseudo Wigner-Ville distribution (PWVD) and propose a new PWVD-based pitch estimation method. We devise an efficient algorithm to compute PWVD faster and use cepstrum-based pre-filtering to avoid cross-term interference. Evaluating our approach on a database with speech and electroglottograph (EGG) recordings yields a state-of-the-art mean absolute error (MAE) of around 4Hz. Our approach is also effective at voiced/unvoiced classification and handling sudden frequency changes

    機械学習を用いたコグニティブ無線における変調方式識別に関する研究

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    The current spectrum allocation cannot satisfy the demand for future wireless communications, which prompts extensive studies in search of feasible solutions for the spectrum scarcity. The burden in terms of the spectral efficiency on the radio frequency terminal is intended to be small by cognitive radio (CR) systems that prefer low power transmission, changeable carrier frequencies, and diverse modulation schemes. However, the recent surge in the application of the CR has been accompanied by an indispensable component: the spectrum sensing, to avoid interference towards the primary user. This requirement leads to a complex strategy for sensing and transmission and an increased demand for signal processing at the secondary user. However, the performance of the spectrum sensing can be extended by a robust modulation classification (MC) scheme to distinguish between a primary user and a secondary user along with the interference identification. For instance, the underlying paradigm that enables a concurrent transmission of the primary and secondary links may need a precise measure of the interference that the secondary users cause to the primary users. An adjustment to the transmission power should be made, if there is a change in the modulation of the primary users, implying a noise oor excess at the primary user location; else, the primary user will be subject to interference and a collision may occur.Alternatively, the interweave paradigm that progresses the spectrum efficiency by reusing the allocated spectrum over a temporary space, requires a classification of the intercepted signal into primary and secondary systems. Moreover, a distinction between noise and interference can be accomplished by modulation classification, if spectrum sensing is impossible. Therefore, modulation classification has been a fruitful area of study for over three decades.In this thesis, the modulation classification algorithms using machine learning are investigated while new methods are proposed. Firstly, a supervised machine learning based modulation classification algorithm is proposed. The higher-order cumulants are selected as features, due to its robustness to noise. Stacked denoising autoencoders,which is an extended edition of the neural network, is chosen as the classifier. On one hand stacked pre-train overcomes the shortcoming of local optimization, on the other, denoising function further enhances the anti-noise performance. The performance of this method is compared with the conventional methods in terms of the classification accuracy and execution speed. Secondly, an unsupervised machine learning based modulation classification algorithm is proposed.The features from time-frequency distribution are extracted. Density-based spatial clustering of applications with noise (DBSCAN) is used as the classifier because it is impossible to decide the number of clusters in advance. The simulation reveals that this method has higher classification accuracy than the conventional methods. Moreover, the training phase is unnecessary for this method. Therefore, it has higher workability then supervised method. Finally, the advantages and dis-advantages of them are summarized.For the future work, algorithm optimization is still a challenging task, because the computation capability of hardware is limited. On one hand, for the supervised machine learning, GPU computation is a potential solution for supervised machine learning, to reduce the execution cost. Altering the modulation pool, the network structure has to be redesigned as well. On the other hand, for the unsupervised machine learning, that shifting the symbols to carrier frequency consumes extra computing resources.電気通信大学201

    Respiratory rate derived from smartphone-camera-acquired pulse photoplethysmographic signals

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    A method for deriving respiratory rate from smartphone-camera-acquired pulse photoplethysmographic (SCPPG) signal is presented. Our method exploits respiratory information by examining the pulse wave velocity and dispersion from the SCPPG waveform and we term these indices as the pulse width variability (PWV). A method to combine information from several derived respiration signals is also presented and it is used to combine PWV information with other methods such as pulse amplitude variability (PAV), pulse rate variability (PRV), and respiration-induced amplitude and frequency modulations (AM and FM) in SCPPG signals Evaluation is performed on a database containing SCPPG signals recorded from 30 subjects during controlled respiration experiments at rates from 0.2 to 0.6 Hz with an increment of 0.1 Hz, using three different devices: iPhone 4S, iPod 5, and HTC One M8. Results suggest that spontaneous respiratory rates (0.2–0.4 Hz) can be estimated from SCPPG signals by the PWV- and PRVbased methods with low relative error (median of order 0.5% and interquartile range of order 2.5%). The accuracy can be improved by combining PWV and PRV with other methods such as PAV, AM and/or FM methods. Combination of these methods yielded low relative error for normal respiratory rates, and Institute of Physics and Engineering in Medicine maintained good performance at higher rates (0.5–0.6 Hz) when using the iPhone 4S or iPod 5 devices

    Auditory motivated level-crossing approach to instantaneous frequency estimation

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    Three-Parametric Cubic Interpolation for Estimating the Fundamental Frequency of the Speech Signal

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    In this paper, we propose a three-parametric convolution kernel which is based on the one-parameter Keys kernel. The first part of the paper describes the structure of the three-parameter convolution kernel. Then, a certain analytical expression for finding the position of the maximum of the reconstructed function is given. The second part presents an algorithm for estimating the fundamental frequency of the speech signal processing in the frequency domain using Picking Picks methods and parametric cubic convolution. Furthermore, the results of experiments give the estimated fundamental frequency of speech and sinusoidal signals in order to select the optimal values of the parameters of the proposed convolution kernel. The results of the fundamental frequency estimation according to the mean square error are given by tables and graphics. Consequently, it is used as a basis for a comparative analysis. The analysis derived the optimal parameters of the kernel and the window function that generates the least MSE. Results showed a higher efficiency in comparison to two or three-parameter convolution kernel

    An investigation of gear meshing behaviour of planetary gear systems for improved fault diagnosis

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    This research has presented gear dynamic models and associated simulations to improve gear fault detection. These models include the use of finite element and lumped parameter methods for both fixed axis and planetary gear systems. The findings in this research provide an improved understanding of the gear fault mechanism and advance the gear fault detection capability of the whole drive train system. It also suggests further effective ways of monitoring the whole gear train system

    Sparsity and convex programming in time-frequency processing

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    Cataloged from PDF version.Thesis (Ph.D.): Bilkent University, The Department of Electrical and Electronics Engineering and The Graduate School of Engineering and Science of Bilkent Univesity, 2014.Includes bibliographical references (leaves 120-131).In this thesis sparsity and convex programming-based methods for timefrequency (TF) processing are developed. The proposed methods aim to obtain high resolution and cross-term free TF representations using sparsity and lifted projections. A crucial aspect of Time-Frequency (TF) analysis is the identification of separate components in a multi component signal. Wigner-Ville distribution is the classical tool for representing such signals but suffers from cross-terms. Other methods that are members of Cohen’s class distributions also aim to remove the cross terms by masking the Ambiguity Function (AF) but they result in reduced resolution. Most practical signals with time-varying frequency content are in the form of weighted trajectories on the TF plane and many others are sparse in nature. Therefore the problem can be cast as TF distribution reconstruction using a subset of AF domain coefficients and sparsity assumption in TF domain. Sparsity can be achieved by constraining or minimizing the l1 norm. Projections Onto Convex Sets (POCS) based l1 minimization approach is proposed to obtain a high resolution, cross-term free TF distribution. Several AF domain constraint sets are defined for TF reconstruction. Epigraph set of l1 norm, real part of AF and phase of AF are used during the iterative estimation process. A new kernel estimation method based on a single projection onto the epigraph set of l1 ball in TF domain is also proposed. The kernel based method obtains the TF representation in a faster way than the other optimization based methods. Component estimation from a multicomponent time-varying signal is considered using TF distribution and parametric maximum likelihood (ML) estimation. The initial parameters are obtained via time-frequency techniques. A method, which iterates amplitude and phase parameters separately, is proposed. The method significantly reduces the computational complexity and convergence time.by Zeynel Deprem.Ph.D

    The Hilbert-Huang Transform for Damage Detection in Plate Structures

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    This thesis investigates the detection of structural damage in plate structures using the empirical mode decomposition method along with the Hilbert spectral analysis. In recent years there have been an extensive amount of research associated with the development of health monitoring methods for aerospace systems, such as aging aircraft and Health and Usage Monitoring Systems (HUMS) for rotorcraft. The method developed here exploits a new time-frequency signal processing analysis tool, the Hilbert-Huang transform, along with the Lamb wave propagation for thin plates. With the use of the wave reflections from discontinuities, damage identification methods were developed to determine the presence, location and extent of damage in isotropic and composite plate structures. The ability of the empirical mode decomposition to extract embedded oscillations, to reveal hidden reflections in the data and to provide a high-resolution energy-time-frequency spectrum is used to describe the Lamb waves interactions with various damaged regions

    Efficient material characterization by means of the Doppler effect in microwaves

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    Subject of this thesis is the efficient material characterization and defects detection by means of the Doppler effect with microwaves. The first main goal of the work is to develop a prototype of a microwave Doppler system for Non-Destructive Testing (NDT) purposes. Therefore it is necessary that the Doppler system satisfies the following requirements: non-expensive, easily integrated into industrial process, allows fast measurements. The Doppler system needs to include software for hardware control, measurements, and fast signal processing. The second main goal of the thesis is to establish and experimentally confirm possible practical applications of the Doppler system. The Doppler system consists of the following parts. The hardware part is designed in a way to ensure fast measurement and easy adjustment to the different radar types. The software part of the system contains tools for: hardware control, data acquisition, signal processing and representing data to the user. In this work firstly a new type of 2D Doppler amplitude imaging was developed and formalized. Such a technique is used to derive information about the measured object from several angles of view. In the thesis special attention was paid to the frequency analysis of the mea- sured signals as a means to improve spatial resolution of the radar. In the context of frequency analysis we present 2D Doppler frequency imaging and compare it with amplitude imaging. In the thesis the spatial resolution ability of CW radars is examined and im- proved. We show that the joint frequency and the amplitude signal processing allows to significantly increase the spatial resolution of the radar.Das Thema dieser Dissertation ist die effiziente Materialcharakterisierung und Fehlerdetektion durch Nutzung des Dopplereffektes mittels Mikrowellen. Das erste Hauptziel der Arbeit ist die Entwicklung eines Prototyps eines Mikrowellen-Doppler-Systems im Bereich der zerstörungsfreien Prüfung. Das Doppler-System muss folgenden Voraussetzungen erfüllen: es sollte preisgünstig sein, leicht in industrielle Prozesse integrierbar sein und schnelle Messungen erlauben. Das Doppler-System muss die Software für die Hardware-Kontrolle, den Messablauf und die schnelle Signalverarbeitung beinhalten. Das zweite Hauptziel der Dissertation ist es, mögliche praktische Anwendungsfelder des Doppler-Systems zu identifizieren und experimentell zu bearbeiten. Das Doppler-System besteht aus zwei Teilen. Der Hardware-Teil ist so konstruiert, dass er schnelle Messungen und leichte Anpassungen an verschiedene Sensor- und Radartypen zulässt. Der Software-Teil des Systems beinhaltet Werkzeuge für: Hardware-Kontrolle, Datenerfassung, Signalverarbeitung und Programme, um die Daten für den Benutzer zu präsentieren. In dieser Arbeit wurde zuerst ein neuer Typ der 2D-Doppler-Amplitudenbildgebung entwickelt und formalisiert. Dieser Technik wird dafür benutzt, Informationen über die gemessenen Objekte von verschiedenen Blickpunkten aus zu erhalten. In dieser Doktorarbeit wird der Frequenzanalyse der gemessenen Signale besondere Aufmerksamkeit geschenkt, um die Ortsauflösung des Radars zu verbessern. Im Kontext der Frequenzanalyse wird die 2D-Doppler-Frequenzbildgebung präsentiert und mit der Amplitudenbildgebung vergleichen. In dieser Dissertation werden die räumliche Auflösungsmöglichkeiten von CW-Radaren untersucht und verbessert. Es wird gezeigt, dass es die Frequenz- und Amplitudensignalverarbeitung erlaubt, die Ortsauflösung des Radars erheblich zu erhöhen
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