Power quality has become one of major concerns in the power industry. It can be described as the reliability of the electric power to maintain continuity operation of end-use equipment. Power quality problems are defined as deviation of voltage or current waveforms from the ideal value. The expansion plan of wind power generation has raised concern regarding how it influences the voltage and current signals. The variability nature of wind energy and the requirements of wind power generation increase the potential problems such as frequency and harmonic distortions. In order to analyze and mitigate problems in wind power generation, it is important to monitor power quality in wind farm. Therefore, the more accurate and reliable parameter estimation methods suitable for wind power generation are needed. Three parameter estimation methods are proposed in this thesis to estimate the unknown parameters, i.e. amplitude and phase angle of fundamental and harmonic components, DC component and system frequency, during the dynamic change in wind farm. In the first method, a self-tuning procedure is introduced to least square method to increase the immunity of the algorithm to noise. In the second method, nonrecursive Newton Type Algorithm is utilised to estimate the unknown parameters by obtaining the left pseudoinverse of Jacobian matrix. In the last technique, unscented transformation is used to replace the linearization procedure to obtain mean and covariance which will be used in Kalman filter method. All of the proposed methods have been tested rigorously using computer simulated data and have shown their capability to track the unknown parameters under extreme distortions. The performances of proposed methods have also been compared using real recorded data from several wind farms in Europe and have demonstrated high correlation. This comparison has verified that UKF requires the shortest processing time and STLS requires the longest.

Este trabalho apresenta um método para a estimação da freqüência em sistemas elétricos de potência utilizando filtros adaptativos baseados no algoritmo dos mínimos quadrados (LMS - least mean square). A análise do sistema de potência é realizada através da conversão das tensões trifásicas em um sinal complexo pela aplicação da transformada \'alfa\'\'beta\', cuja forma complexa foi direcionada ao algoritmo de filtragem adaptativa. O método é baseado na aplicação da filtragem adaptativa para a realização de rastreio do sinal de entrada, o que permite verificar o seu comportamento variante no tempo. O algoritmo proposto foi testado através de formas de ondas geradas com o software Matlab e de simulações realizadas através do software Alternative Transients Program (ATP). É importante salientar que nas simulações do ATP foram modelados diversos equipamentos que constituem o sistema elétrico de potência, incluindo um gerador síncrono com regulação de velocidade, linhas de transmissão com variação em freqüência e transformadores de potência com suas respectivas curvas de saturação. Estas modelagens tiveram por objetivo gerar dados das mais diversas e distintas situações para a verificação e análise da metodologia proposta. Os resultados da pesquisa mostram a excelência na aplicabilidade do algoritmo proposto na estimação da freqüência de um sistema elétrico, mesmo com sinais ruidosos, além do rastreio fiel da freqüência em situações de manobra e operação. Alguns dos resultados apresentados comparam as estimações obtidas pela técnica proposta em relação às estimações de um determinado relé comercial, habilitado à supervisão da freqüência. / This work presents a method for frequency estimation in power systems using adaptive filters based in the algorithm of least mean square (LMS). The analysis of the power system is made through the conversion of the three-phase voltages in a complex signal with the application of \'alfa\'\'beta\' transform, whose complex form was directed to the algorithm of adaptive filtering. The method is based on the application of the adaptive filtering for tracking the input signal, and it allows verifying its variant behavior in time. The algorithm was tested through waveforms generated by Matlab software and simulations carried out through Alternative Transients Program (ATP) software. It is important to point out that in the simulations using ATP many diferent power system equipments had been modeled, including a synchronous generator with speed regulation, transmission lines with variation in frequency and power transformers with their saturation curves. The objective of these tests was to generate data for diverse and distinct situations for the verification and the analysis of the proposed methodology. The results of the research show the excellence in the applicability of the algorithm considered in frequency estimation of an electrical system, even with noisy signals, as well as the tracking of the frequency during operation. Some of the results are compared to the ones presented by a commercial relay set to track frequency.

Este trabalho apresenta um método para a estimação da freqüência em sistemas elétricos de potência utilizando filtros adaptativos baseados no algoritmo dos mínimos quadrados (LMS - least mean square). A análise do sistema de potência é realizada através da conversão das tensões trifásicas em um sinal complexo pela aplicação da transformada \'alfa\'\'beta\', cuja forma complexa foi direcionada ao algoritmo de filtragem adaptativa. O método é baseado na aplicação da filtragem adaptativa para a realização de rastreio do sinal de entrada, o que permite verificar o seu comportamento variante no tempo. O algoritmo proposto foi testado através de formas de ondas geradas com o software Matlab e de simulações realizadas através do software Alternative Transients Program (ATP). É importante salientar que nas simulações do ATP foram modelados diversos equipamentos que constituem o sistema elétrico de potência, incluindo um gerador síncrono com regulação de velocidade, linhas de transmissão com variação em freqüência e transformadores de potência com suas respectivas curvas de saturação. Estas modelagens tiveram por objetivo gerar dados das mais diversas e distintas situações para a verificação e análise da metodologia proposta. Os resultados da pesquisa mostram a excelência na aplicabilidade do algoritmo proposto na estimação da freqüência de um sistema elétrico, mesmo com sinais ruidosos, além do rastreio fiel da freqüência em situações de manobra e operação. Alguns dos resultados apresentados comparam as estimações obtidas pela técnica proposta em relação às estimações de um determinado relé comercial, habilitado à supervisão da freqüência. / This work presents a method for frequency estimation in power systems using adaptive filters based in the algorithm of least mean square (LMS). The analysis of the power system is made through the conversion of the three-phase voltages in a complex signal with the application of \'alfa\'\'beta\' transform, whose complex form was directed to the algorithm of adaptive filtering. The method is based on the application of the adaptive filtering for tracking the input signal, and it allows verifying its variant behavior in time. The algorithm was tested through waveforms generated by Matlab software and simulations carried out through Alternative Transients Program (ATP) software. It is important to point out that in the simulations using ATP many diferent power system equipments had been modeled, including a synchronous generator with speed regulation, transmission lines with variation in frequency and power transformers with their saturation curves. The objective of these tests was to generate data for diverse and distinct situations for the verification and the analysis of the proposed methodology. The results of the research show the excellence in the applicability of the algorithm considered in frequency estimation of an electrical system, even with noisy signals, as well as the tracking of the frequency during operation. Some of the results are compared to the ones presented by a commercial relay set to track frequency.

In this thesis, noncoherent differential demodulation of CPM signals with joint carrier frequency offset and symbol timing estimation is investigated. CPM is very attractive for wireless communications owing to major properties: good spectral efficiency and a constant envelope property. In order to demodulate the received CPM signal differentially, the symbol timing and the carrier frequency offset have to be estimated accurately. There are numerous methods developed for the purpose. However, we have not encountered studies (which are based on autocorrelation estimation and hence suitable for blind synchronization) that give expectable performance for both M-ary and partial response signaling. Thus, in this thesis we analyze a feedforward blind estimation scheme, which recovers the symbol timing and the frequency offset of M-ary CPM signals and partial response CPM signals. In addition, we surveyed low complexity symbol detection methods for CPM signals. Reduced state Viterbi differential detector incorporated to the joint frequency offset and symbol timing estimator is also examined. The performance of the examined demodulator scheme is assessed for the AWGN channel by computer simulations.

Le terme élastographie désigne les techniques d'imagerie dédiées à l'étude des propriétés mécaniques des tissus biologiques in vivo. Au cours de cette thèse, nous nous sommes intéressés à deux de ces techniques. La première est l'élastographie quasi-statique par ultrasons permettant de mesurer les déformations locales induites dans un tissu sous l'action d'une contrainte globale. La seconde est l'élastographie par résonance magnétique (ERM) permettant d'accéder localement à une estimation du module de cisaillement. Pour cette dernière technique, une onde de cisaillement est générée au sein du milieu puis imagée a l'aide d'une séquence IRM spécifique. Les images d'ondes acquises permettent la reconstruction du module de cisaillement local. Dans le cadre de ces travaux, une nouvelle technique d'acquisition de l'image d'onde de cisaillement a été proposée, ainsi qu'une méthode de reconstruction du module de cisaillement basée sur l'estimation locale de fréquence par rapport de filtres. Un autre axe de recherche a consisté en l'étude de l'effet d'une précontrainte appliquée à un milieu sur son module de cisaillement mesuré par ERM. Cet effet a tout d'abord été étudié sur des milieux homogènes puis avec des milieux test hétérogènes. Dans ce dernier cas, l'utilisation de l'élastographie quasi-statique par ultrasons s'avère nécessaire pour accéder à la déformation locale du milieu. Cette dernière information a été combinée avec les informations obtenues en ERM pour extraire pour chaque région d'intérêt une courbe déformation/module de cisaillement / The term elastography refers to imaging techniques dedicated to the in vivo investigation of the mechanical properties of biological tissues. During this thesis, we focused on two elastography techniques. The first one is quasi-static ultrasound elastography, able to locally estimate tissue strain induced by a global deformation of a medium. The second one is Magnetic Resonance Elastography (MRE), able to measure the local shear modulus. In MRE, a shear wave is generated within the medium and imaged using a specific MRI sequence. The resulting wave images are then processed to estimate the local shear modulus. A new acquisition scheme of the shear wave images was proposed during this thesis. A method, based on local frequency estimation, was also developed for the estimation of the local shear modulus using the properties of a ratio of filters. Another research axis was the study of the effect of a prestrain application on the measured shear modulus. This effect was first studied with homogeneous media and then with heterogeneous test objects. In this last case, the use of quasi-static ultrasound elastography was necessary to locally access to the medium strain. This information was then combined with the information obtained using MRE to extract, for each region of interest, a strain/shear modulus curve

This Master's thesis is focused on creating of an algorithm which calculates changes of negative-sequence voltages and currents from their instantaneous values. That allows to conduct localization of asymmetrical faults in MV network in line with the Vdip method, which is based on monitoring the changes of negative-sequence components at distribution substations and at a sub-transmission station. The algorithm is being developed in Matlab environment with continuous implementation of partial procedures which are being assessed and compared with each other. A study of phasor estimating methods is carried out with pointing out related problems which are mainly caused by Ripple control and deviation of system frequency from its nominal value. Optimization precautions are designed to mitigate these problems. For elimination of the Ripple control effects a method based on averaging is presented. The deviation of system frequency is dealt with by resampling the original data recordings. The analysis processes are tested by both simulation signals and real measured data. The optimized algorithm enables precise calculation of negative-sequence components changes which is the main contribution of this thesis. The constructed algorithm is used in verification of the Vdip method on physical model of MV network. For these purposes a simple distribution network is created within which ground faults on different places and with different resistances are realised. The results of localization are not convincing which is mainly caused by specific features of laboratory power line models which are constructed with heterogenous parameters.

Bei der Überführung eines wissenschaftlichen Laborsystems zur automatischen Spracherkennung in eine reale Anwendung ergeben sich verschiedene praktische Problemstellungen, von denen eine der Verlust an Erkennungsleistung durch umgebende akustische Störungen ist. Im Gegensatz zu additiven Störungen wie Lüfterrauschen o. ä. hat die Wissenschaft bislang die Störung des Raumhalls bei der Spracherkennung nahezu ignoriert. Dabei besitzen, wie in der vorliegenden Dissertation deutlich gezeigt wird, bereits geringfügig hallende Räume einen stark störenden Einfluss auf die Leistungsfähigkeit von Spracherkennern. Mit dem Ziel, die Erkennungsleistung wieder in einen praktisch benutzbaren Bereich zu bringen, nimmt sich die Arbeit dieser Problemstellung an und schlägt Lösungen vor. Der Hintergrund der wissenschaftlichen Aktivitäten ist die Erstellung von funktionsfähigen Sprachbenutzerinterfaces für Gerätesteuerungen im Wohn- und Büroumfeld, wie z.~B. bei der Hausautomation. Aus diesem Grund werden praktische Randbedingungen wie die Restriktionen von embedded Computerplattformen in die Lösungsfindung einbezogen. Die Argumentation beginnt bei der Beschreibung der raumakustischen Umgebung und der Ausbreitung von Schallfeldern in Räumen. Es wird theoretisch gezeigt, dass die Störung eines Sprachsignals durch Hall von zwei Parametern abhängig ist: der Sprecher-Mikrofon-Distanz (SMD) und der Nachhallzeit T60. Um die Abhängigkeit der Erkennungsleistung vom Grad der Hallstörung zu ermitteln, wird eine Anzahl von Erkennungsexperimenten durchgeführt, die den Einfluss von T60 und SMD nachweisen. Weitere Experimente zeigen, dass die Spracherkennung kaum durch hochfrequente Hallanteile beeinträchtigt wird, wohl aber durch tieffrequente. In einer Literaturrecherche wird ein Überblick über den Stand der Technik zu Maßnahmen gegeben, die den störenden Einfluss des Halls unterdrücken bzw. kompensieren können. Jedoch wird auch gezeigt, dass, obwohl bei einigen Maßnahmen von Verbesserungen berichtet wird, keiner der gefundenen Ansätze den o. a. praktischen Einsatzbedingungen genügt. In dieser Arbeit wird die Methode Harmonicity-based Feature Analysis (HFA) vorgeschlagen. Sie basiert auf drei Ideen, die aus den Betrachtungen der vorangehenden Kapitel abgeleitet werden. Experimentelle Ergebnisse weisen die Verbesserung der Erkennungsleistung in halligen Umgebungen nach. Es werden sogar praktisch relevante Erkennungsraten erzielt, wenn die Methode mit verhalltem Training kombiniert wird. Die HFA wird gegen Ansätze aus der Literatur evaluiert, die ebenfalls praktischen Implementierungskriterien genügen. Auch Kombinationen der HFA und einigen dieser Ansätze werden getestet. Im letzten Kapitel werden die beiden Basistechnologien Stimm\-haft-Stimmlos-Entscheidung und Grundfrequenzdetektion umfangreich unter Hallbedingungen getestet, da sie Voraussetzung für die Funktionsfähigkeit der HFA sind. Als Ergebnis wird dargestellt, dass derzeit für beide Technologien kein Verfahren existiert, das unter Hallbedingungen robust arbeitet. Es kann allerdings gezeigt werden, dass die HFA trotz der Unsicherheiten der Verfahren arbeitet und signifikante Steigerungen der Erkennungsleistung erreicht. / Automatic speech recognition (ASR) systems used in real-world indoor scenarios suffer from performance degradation if noise and reverberation conditions differ from the training conditions of the recognizer. This thesis deals with the problem of room reverberation as a cause of distortion in ASR systems. The background of this research is the design of practical command and control applications, such as a voice controlled light switch in rooms or similar applications. Therefore, the design aims to incorporate several restricting working conditions for the recognizer and still achieve a high level of robustness. One of those design restrictions is the minimisation of computational complexity to allow the practical implementation on an embedded processor. One chapter comprehensively describes the room acoustic environment, including the behavior of the sound field in rooms. It addresses the speaker room microphone (SRM) system which is expressed in the time domain as the room impulse response (RIR). The convolution of the RIR with the clean speech signal yields the reverberant signal at the microphone. A thorough analysis proposes that the degree of the distortion caused by reverberation is dependent on two parameters, the reverberation time T60 and the speaker-to-microphone distance (SMD). To evaluate the dependency of the recognition rate on the degree of distortion, a number of experiments has been successfully conducted, confirming the above mentioned dependency of the two parameters, T60 and SMD. Further experiments have shown that ASR is barely affected by high-frequency reverberation, whereas low frequency reverberation has a detrimental effect on the recognition rate. A literature survey concludes that, although several approaches exist which claim significant improvements, none of them fulfils the above mentioned practical implementation criteria. Within this thesis, a new approach entitled 'harmonicity-based feature analysis' (HFA) is proposed. It is based on three ideas that are derived in former chapters. Experimental results prove that HFA is able to enhance the recognition rate in reverberant environments. Even practical applicable results are achieved when HFA is combined with reverberant training. The method is further evaluated against three other approaches from the literature. Also combinations of methods are tested. In a last chapter the two base technologies fundamental frequency (F0) estimation and voiced unvoiced decision (VUD) are evaluated in reverberant environments, since they are necessary to run HFA. This evaluation aims to find one optimal method for each of these technologies. The results show that all F0 estimation methods and also the VUD methods have a strong decreasing performance in reverberant environments. Nevertheless it is shown that HFA is able to deal with uncertainties of these base technologies as such that the recognition performance still improves.

Le problème de séparation aveugle de sources a but de retrouver un ensemble des sources signaux statistiquement indépendants à partir seulement d’un ensemble des observations du capteur. Ces observations peuvent être modélisées comme un mélanges linéaires instantané ou convolutifs de sources. Dans cette thèse, les sources signaux sont supposées être cyclostationnaire où leurs fréquences cycles peuvent être connues ou inconnu par avance. Premièrement, nous avons établi des relations entre le spectre, spectre de puissance d’un signal source et leurs composants, puis nous avons proposé deux nouveaux algorithmes pour estimer sa fréquences cycliques. Ensuite, pour la séparation aveugle de sources en mélanges instantanés, nous présentons quatre algorithmes basés sur diagonalisation conjoint approchées orthogonale (ou non-orthogonales) d’une famille des matrices cycliques multiples moment temporel, or l’approche matricielle crayon pour extraire les sources signaux. Nous introduisons aussi et prouver une nouvelle condition identifiabilité pour montrer quel type de sources cyclostationnaires d’entrée peuvent être séparées basées sur des statistiques cyclostationnarité à l’ordre deux. Pour la séparation aveugle de sources en mélanges convolutifs, nous présentons un algorithme en deux étapes basées sur une approche dans le domaine temporel pour récupérer les signaux source. Les simulations numériques sont utilisés dans cette thèse pour démontrer l’efficacité de nos approches proposées, et de comparer les performances avec leurs méthodes précédentes / Blind source separation problem aims to recover a set of statistically independent source signals from a set of sensor observations. These observations can be modeled as an instantaneous or convolutive mixture of the same sources. In this dissertation, the source signals are assumed to be cyclostationary where their cycle frequencies may be known or unknown a priori. First, we establish relations between the spectrum, power spectrum of a source signal and its component, then we propose two novel algorithms to estimate its cycle frequencies. Next, for blind separation of instantaneous mixtures of sources, we present four algorithms based on orthogonal (or non-orthogonal) approximate diagonalization of the multiple cyclic temporal moment matrices, and the matrix pencil approach to extract the source signal. We also introduce and prove a new identifiability condition to show which kind of input cyclostationary sources can be separated based on second-order cyclostationarity statistics. For blind separation of convolutive mixtures of sources signal or blind deconvolution of FIR MIMO systems, we present a two-steps algorithm based on time domain approach for recovering the source signals. Numerical simulations are used throughout this thesis to demonstrate the effectiveness of our proposed approaches, and compare theirs performances with previous methods

Submitted by Renata Lopes (renatasil82@gmail.com) on 2016-02-16T09:52:46Z No. of bitstreams: 1 alamsilvamenezes.pdf: 8301590 bytes, checksum: aed618e30f38206da4bf4f329924f87e (MD5) / Approved for entry into archive by Adriana Oliveira (adriana.oliveira@ufjf.edu.br) on 2016-02-26T12:30:53Z (GMT) No. of bitstreams: 1 alamsilvamenezes.pdf: 8301590 bytes, checksum: aed618e30f38206da4bf4f329924f87e (MD5) / Made available in DSpace on 2016-02-26T12:30:53Z (GMT). No. of bitstreams: 1 alamsilvamenezes.pdf: 8301590 bytes, checksum: aed618e30f38206da4bf4f329924f87e (MD5) Previous issue date: 2014-09-01 / A presente tese propõe soluções ao problema da explicitação do conteúdo espectral de processos estacionários e não estacionários, com aplicações na estimação de frequência, estimação da densidade espectral de potência e no monitoramento do espectro. A técnica de estimação de frequência proposta nesta tese, baseada na warped discrete Fourier transform, apresenta, de acordo com as simulações computacionais, o melhor desempenho frente às demais técnicas comparadas, atingindo o Cramer-Rao bound para uma ampla faixa de relação sinal ruído. Em relação a estimação da densidade espectral de potência, a Hartley Multitaper method, proposta nesta tese, apresenta desempenho similar à multitaper method, em termos da variância de estimação e da polarização do espectro, mas simpli cação de implementação. Uma técnica para monitoramento do espectro para sistemas power line communication é proposta, levando em consideração o conceito de quanta e a diversidade observada quando os sinais são aquisitados a partir da rede de energia elétrica e do ar. Baseando-se em sinais sintéticos, gerados em computador, assim como dados de medição do espectro, obtidos utilizando uma antena e o cabo de energia elétrica como elementos sensores, veri fica-se que o desempenho da técnica proposta supera a monitoração padrão, sobretudo quando a diversidade gerada pelo cabo e pela antena sobre o sinal monitorado é explorada na detecção. / This dissertation aims at discussing solutions to deal with spectral analysis of stationary and non-stationary processes for frequency estimation, power spectral density estimation and spectral monitoring applications. The frequency estimation techniques are assessed through computer simulations. The proposed technique for frequency estimation is based on warped discrete Fourier transform outperforms other techniques, achieving the Cramer-Rao Bound for a wide range of signal to noise ratio. Regarding the power spectral density estimation, the proposed Hartley Multitaper Method shows similar performance, in terms of variance of estimates and polarization spectrum; however, it can simplify the implementation complexity. The introduced spectrum sensing technique is based on quanta de nition and the diversity o ered by the signals acquired from the electric power grids and the air. Based on computer-generation data and those one obtained during a measurement campaign, which one in this thesis is evaluated using synthetic signals, generated by computer, as well as measurement data of the spectrum. The numerical results show that the proposed technique outperforms a previous technique and can attain the very detection ratio and the very low false alarm when the diversity yielded by electric power grid and air is exploited.

Submitted by Renata Lopes (renatasil82@gmail.com) on 2017-04-25T17:30:25Z No. of bitstreams: 1 carloshenriquenascimentomartins.pdf: 10221613 bytes, checksum: 0d6eef0f715fc0f9f68bf12a390dcd55 (MD5) / Approved for entry into archive by Adriana Oliveira (adriana.oliveira@ufjf.edu.br) on 2017-04-26T12:22:24Z (GMT) No. of bitstreams: 1 carloshenriquenascimentomartins.pdf: 10221613 bytes, checksum: 0d6eef0f715fc0f9f68bf12a390dcd55 (MD5) / Made available in DSpace on 2017-04-26T12:22:24Z (GMT). No. of bitstreams: 1 carloshenriquenascimentomartins.pdf: 10221613 bytes, checksum: 0d6eef0f715fc0f9f68bf12a390dcd55 (MD5) Previous issue date: 2015-03-26 / CAPES - Coordenação de Aperfeiçoamento de Pessoal de Nível Superior / Esta tese apresenta as etapas de desenvolvimento de um sistema de monitoramento de parâmentos de qualidade de energia dedicado a análise de harmônicos variantes no tempo através do equipamento denominado AHVT (Analisador de Harmônicos Variantes no Tempo). O desenvolvimento do trabalho é composto por: (i) estudo e implementação MATLAB de algoritmos para processamento em tempo real, com capacidade de sintonização dos componentes harmônicos; (ii) análise e desenvolvimento de estratégias para detecção e validação da presença de interharmônicos próximos à frequência fundamental e suas consequência na estimação de parâmetros como fase, amplitude e frequência para o componente fundamental, (iii) proposta de implementação do dispositivo, sistema de aquisição/ condicionamento de sinais/ filtragem, sistema de conversão analógico digital e plataforma de processamentoDSP/FPGA, sistema de transmissão de dados e plataformas de análise online/offline dos eventos de harmônicos variantes no tempo; (iv) plataforma de simulação do Analisador de Harmônicos Variantes no Tempo (AHVT) para estudo dos métodos de trigger para detecção e captura dos eventos. / In this work is presented the steps of development and implementation of a Power Quality paramaters monitoring system with main goal events denomined ”time arying harmonics”named of Time Varying Harmonic Analyzer. The development is comprises:(i) research and implementation of real time algorithms with capable to tuning harmonic waves,(ii) Analyze and research/development of strategies for detect and validation of interharmonics with frequencies near of fundamental, and conseguencies and challenges to phase, magnitude and frequency estimation with presence interharmonic waveform (iii) The proposal of a hardware design including analog to digital conversion and digital signal processing plataform, broadcast data link and IHM(Interface Human Machine) for online and offline analyzes to time varying harmonic analyzer;(iiii)off-line simulation plataform of Analisador de Harmônicos Variantes no Tempo Time Varying Harmonic Analyzer (TVHA) to trigger detect methods to detection and capture of waveforms.