Ce manuscrit porte sur la synthèse d'observateurs grand gain pour des systèmes non linéaires à sorties échantillonnées et retardées. Trois contributions sont proposées à la lecture de ce manuscrit. La première contribution, pour une classe de systèmes Multi-entrées / Multi-sorties uniformément observables et dont les sorties sont regroupées en un seul bloc, met en jeu le problème du processus d'acquisition des mesures de sorties (continues, échantillonnées, retardées ou non) et propose un cadre commun pour l'ensemble des cas possibles. La deuxième contribution propose un observateur grand gain filtré sur cette même classe de systèmes dans l'optique de réduire la sensibilité au bruit de mesure, dans le cas où la sortie est continue puis dans le cas où cette dernière est échantillonnée. La dernière contribution vise à étendre la synthèse grand gain standard pour une large classe de systèmes Multi-entrées / Multi-sorties uniformément observables dont les mesures des sorties sont continues. Pour chacune de ces contributions, il a été montré que l'erreur d'observation de chacun des observateurs proposés converge exponentiellement vers zéro en l'absence d'incertitudes sur le système. Toutes ces contributions ont été illustrées par différents exemples issus de plusieurs domaines d'étude. / This manuscript deals with the synthesis of high gain observers for nonlinear systems with sampled and delayed outputs. Three contributions are proposed for consideration in this manuscript. The first contribution, for a class of Multi-input / Multi-output systems whose outputs are grouped into a single block, involves the problem of the acquisition process of output measurements (continuous, sampled, delayed or not) and proposes a common framework for all possible cases. The second contribution proposes a filtered high gain observer on this same class of systems in order to reduce the sensitivity to measurement noise, in the case where the output is continuous and then in the case where the latter is sampled. The last contribution aims to extend the standard high gain synthesis for a large class of uniformly observable Multi-input / Multi-output systems with continuous output measurements. For each of these contributions, it has been shown that the observation error of each of the proposed observers converges exponentially towards zero in the absence of uncertainties in the system. All these contributions have been illustrated through several examples from different fields of study.

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Instrumented umanned aerial vehicles represent a new way of measuring turbulence in the atmospheric boundary layer. However, autonomous measurements require control methods with disturbance-rejection and altitude command-following capabilities. Filtered dynamic inversion is a control method with desirable disturbance-rejection and command-following properties, and this controller requires limited model information. We implement filtered dynamic inversion as the pitch controller in an altitude-hold autopilot. We design and numerically simulate the continuous-time and discrete-time filtered-dynamic-inversion controllers with anti-windup on a nonlinear aircraft model. Finally, we present results from a flight experiment comparing the filtered-dynamic-inversion controller to a classical proportional-integral controller. The experimental results show that the filtered-dynamic-inversion controller performs better than a proportional-integral controller at certain values of the parameter.

Universidade Federal do Triângulo Mineiro / This work is dedicated to the study of the problem of active noise control, evaluating some numerical and experimental methodologies. The analysis is restricted to the case of noises in ducts, in which the acoustic propagation phenomena are modeled. Four approaches for this type of models are presented. The first one is formulated by using the basic equations of the acoustics. This procedure generates an infinite dimension model of the duct. In the second approach, the infinite model is truncated by using Taylor s series. The third approach performs a modal expansion using the poles of the infinite dimension model, and, in the fourth, it is also considered a modal expansion, but in this case, by taking into account zeros and poles of the infinite dimension model. The four models studied are discussed and compared in the present contribution. A second part of this work is concerned with active noise control techniques. Monochannel (which uses only a loud speaker and a microphone) and multi-channel (which uses several loud speakers and microphones) controllers are studied. The studied active noise controllers use LMS adaptive algorithms. The noise signals are filtered using X-LMS techniques. These types of controller are usually simple and robust. The coefficients of the controller (modeled as a digital filter) are determined by using an online adaptive procedure looking for minimizing the noise levels. The control methodologies are tested numerically by using the mathematical model of the acoustic duct proposed. With the aim of validating experimentally these controllers a test rig instrumented with loud speakers and microphones was built, and the algorithms were implemented using a personal computer. At the remaining, the numerical and experimental results are discussed and some suggestions are presented in order to continue future works. / Este trabalho formula e discorre sobre o problema de controle ativo de ruído e avalia algumas metodologias de controle tanto numérica como experimentalmente. A análise é restrita ao caso de ruídos em dutos, onde o fenômeno da propagação acústico é analiticamente modelado. Apresentam-se quatro abordagens para tal modelagem. A primeira, formulada a partir das equações fundamentais da acústica, gera um modelo de dimensão infinita para o duto. A segunda aproxima o modelo infinito por uma série truncada de Taylor. A terceira formulação realiza uma expansão modal, a partir dos pólos do modelo de dimensão infinita e a quarta, também realiza uma expansão modal, mas considera tanto os pólos como os zeros do modelo infinito dimensional. No trabalho são discutidos e comparados os quatro modelos numéricos propostos. Numa segunda parte este trabalho discorre-se sobre diversas técnicas de controle ativo de ruído em dutos. São estudados controladores do tipo mono canal, que utilizam um sensor e um atuador apenas e controladores do tipo multicanal, com vários sensores e atuadores. Todos os controladores ativos de ruído (CAR) estudados utilizam algoritmos adaptativos do tipo LMS (Least Mean Square) e técnicas de filtragem-X LMS. Este tipo de controlador tem como características marcantes à simplicidade e a robustez. Os coeficientes do controlador, modelado como um filtro digital, são adaptados on-line segundo uma estratégia que busca minimizar os ruídos não desejados. Estas metodologias de controle são testadas numericamente a partir do modelo matemático proposto para o duto acústico. Para avaliar também experimentalmente tais controladores, montou-se uma bancada de testes constituída por um duto de PVC instrumentada com alto falantes e microfones sendo os algoritmos de controle implementados em um microcomputador pessoal devidamente configurado. O trabalho encerra discutindo os resultados numéricos e experimentais obtidos e sugerindo desdobramentos a serem investigados no futuro. / Doutor em Engenharia Mecânica

In this thesis an algorithm is demonstrated for the reconstruction of hard-field Tomography images through localized block areas, obtained in parallel and from a multiresolution framework. Block areas are subsequently tiled to put together the full size image. Given its properties to preserve its compact support after being ramp filtered, the wavelet transform has received to date much attention as a promising solution in radiation dose reduction in medical imaging, through the reconstruction of essentially localised regions. In this work, this characteristic is exploited with the aim of reducing the time and complexity of the standard reconstruction algorithm. Independently reconstructing block images with geometry allowing to cover completely the reconstructed frame as a single output image, allows the individual blocks to be reconstructed in parallel, and to experience its performance in a multiprocessor hardware reconfigurable system (i.e. FPGA). Projection data from simulated Radon Transform (RT) was obtained at 180 evenly spaced angles. In order to define every relevant block area within the sinogram, forward RT was performed over template phantoms representing block frames. Reconstruction was then performed in a domain beyond the block frame limits, to allow calibration overlaps when fitting of adjacent block images. The 256 by 256 Shepp-Logan phantom was used to test the methodology of both parallel multiresolution and parallel block reconstruction generalisations. It is shown that the reconstruction time of a single block image in a 3-scale multiresolution framework, compared to the standard methodology, performs around 48 times faster. By assuming a parallel implementation, it can implied that the reconstruction time of a single tile, should be very close related to the reconstruction time of the full size and resolution image.

The work described herein deals with the synthesis and characterization of lead chalcogenide thin films and nanocrystals. The first part of thesis describes the properties of semiconductors followed by an analysis on the chemical vapour deposition and nanoparticulate formation. In the next part of thesis, single-source precursors of type thioselenophosphinato, selenoureato, dithiocarbamato and dithiocarbanato complexes of lead have been synthesised and characterised. As-synthesised compounds have been utilised for the fabrication of lead sulfide and lead selenide thin films by aerosol-assisted chemical vapour deposition as well as nanocrystals by colloidal injection method. Lead sulfide thin films were also deposited by liquid-liquid interface from lead dithiocarbanato at room temperature. The as grown thin films of lead sulfide and lead selenide have been characterised by XRD, SEM and energy dispersive x-ray (EDX) analysis. In the second part of the thesis, preparation of lead sulfide and lead selenide nanocrystals in olive oil at low growth temperatures (50-60°C) is described and have shown that by controlling experimental conditions, well-defined particles with tunable emission in mid and far-infrared region can be synthesised. Furthermore, compositionally-tuned PbSxSe1-x nanocrystals has also been prepared by adding controlled amount of sulur and selenium ingredients into lead oxide. Homogenous distribution of sulfur and selenium within alloyed nanocrystals is confirmed by transmission electron microscope studies. Moreover, attempts have been made to prepare quaternary (PbTe/Se/S) nanocrystals of lead chalcogenides and depth (1.9-5.8 nm) profile analysis by x-ray photoelectron spectroscopy confirmed the formation of core/shell/shell type structure i.e. PbTe/S/Se.

Cette thèse est composée de deux parties: une première partie traite le problème de changement de régime et une deuxième partie concerne le processusautorégressif à seuil dont les innovations ne sont pas indépendantes. Toutefois, ces deux domaines de la statistique et des probabilités se rejoignent dans la littérature et donc dans mon projet de recherche. Dans la première partie, nous étudions le problème de changements derégime. Il existe plusieurs méthodes pour la détection de ruptures mais les principales méthodes sont : la méthode de moindres carrés pénalisés (PLS)et la méthode de derivée filtrée (FD) introduit par Basseville et Nikirov. D’autres méthodes existent telles que la méthode Bayésienne de changementde points. Nous avons validé la nouvelle méthode de dérivée filtrée et taux de fausses découvertes (FDqV) sur des données réelles (des données du vent sur des éoliennes et des données du battement du coeur). Bien naturellement, nous avons donné une extension de la méthode FDqV sur le cas des variables aléatoires faiblement dépendantes.Dans la deuxième partie, nous étudions le modèle autorégressif à seuil (en anglais Threshold Autoregessive Model (TAR)). Le TAR est étudié dans la littérature par plusieurs auteurs tels que Tong(1983), Petrucelli(1984, 1986), Chan(1993). Les applications du modèle TAR sont nombreuses par exemple en économie, en biologie, l'environnement, etc. Jusqu'à présent, le modèle TAR étudié concerne le cas où les innovations sont indépendantes. Dans ce projet, nous avons étudié le cas où les innovations sont non corrélées. Nous avons établi les comportements asymptotiques des estimateurs du modèle. Ces résultats concernent la convergence presque sûre, la convergence en loi et la convergence uniforme des paramètres. / This thesis has two parts: the first part deals the change points problem and the second concerns the weak threshold autoregressive model (TAR); the errors are not correlated.In the first part, we treat the change point analysis. In the litterature, it exists two popular methods: The Penalized Least Square (PLS) and the Filtered Derivative introduced by Basseville end Nikirov.We give a new method of filtered derivative and false discovery rate (FDqV) on real data (the wind turbines and heartbeats series). Also, we studied an extension of FDqV method on weakly dependent random variables.In the second part, we spotlight the weak threshold autoregressive (TAR) model. The TAR model is studied by many authors such that Tong(1983), Petrucelli(1984, 1986). there exist many applications, for example in economics, biological and many others. The weak TAR model treated is the case where the innovations are not correlated.

[EN] Common audio systems are designed with the intent of creating real and immersive scenarios that allow the user to experience a particular acoustic sensation that does not depend on the room he is perceiving the sound. However, acoustic devices and multichannel rendering systems working inside a room, can impair the global audio effect and thus the 3D spatial sound. In order to preserve the spatial sound characteristics of multichannel rendering techniques, adaptive filtering schemes are presented in this dissertation to compensate these electroacoustic effects and to achieve the immersive sensation of the desired acoustic system. Adaptive filtering offers a solution to the room equalization problem that is doubly interesting. First of all, it iteratively solves the room inversion problem, which can become computationally complex to obtain when direct methods are used. Secondly, the use of adaptive filters allows to follow the time-varying room conditions. In this regard, adaptive equalization (AE) filters try to cancel the echoes due to the room effects. In this work, we consider this problem and propose effective and robust linear schemes to solve this equalization problem by using adaptive filters. To do this, different adaptive filtering schemes are introduced in the AE context. These filtering schemes are based on three strategies previously introduced in the literature: the convex combination of filters, the biasing of the filter weights and the block-based filtering. More specifically, and motivated by the sparse nature of the acoustic impulse response and its corresponding optimal inverse filter, we introduce different adaptive equalization algorithms. In addition, since audio immersive systems usually require the use of multiple transducers, the multichannel adaptive equalization problem should be also taken into account when new single-channel approaches are presented, in the sense that they can be straightforwardly extended to the multichannel case. On the other hand, when dealing with audio devices, consideration must be given to the nonlinearities of the system in order to properly equalize the electroacoustic system. For that purpose, we propose a novel nonlinear filtered-x approach to compensate both room reverberation and nonlinear distortion with memory caused by the amplifier and loudspeaker devices. Finally, it is important to validate the algorithms proposed in a real-time implementation. Thus, some initial research results demonstrate that an adaptive equalizer can be used to compensate room distortions. / [ES] Los sistemas de audio actuales están diseñados con la idea de crear escenarios reales e inmersivos que permitan al usuario experimentar determinadas sensaciones acústicas que no dependan de la sala o situación donde se esté percibiendo el sonido. Sin embargo, los dispositivos acústicos y los sistemas multicanal funcionando dentro de salas, pueden perjudicar el efecto global sonoro y de esta forma, el sonido espacial 3D. Para poder preservar las características espaciales sonoras de los sistemas de reproducción multicanal, en esta tesis se presentan los esquemas de filtrado adaptativo para compensar dichos efectos electroacústicos y conseguir la sensación inmersiva del sistema sonoro deseado. El filtrado adaptativo ofrece una solución al problema de salas que es interesante por dos motivos. Por un lado, resuelve de forma iterativa el problema de inversión de salas, que puede llegar a ser computacionalmente costoso para los métodos de inversión directos existentes. Por otro lado, el uso de filtros adaptativos permite seguir las variaciones cambiantes de los efectos de la sala de escucha. A este respecto, los filtros de ecualización adaptativa (AE) intentan cancelar los ecos introducidos por la sala de escucha. En esta tesis se considera este problema y se proponen esquemas lineales efectivos y robustos para resolver el problema de ecualización mediante filtros adaptativos. Para conseguirlo, se introducen diferentes esquemas de filtrado adaptativo para AE. Estos esquemas de filtrado se basan en tres estrategias ya usadas en la literatura: la combinación convexa de filtros, el sesgado de los coeficientes del filtro y el filtrado basado en bloques. Más especificamente y motivado por la naturaleza dispersiva de las respuestas al impulso acústicas y de sus correspondientes filtros inversos óptimos, se presentan diversos algoritmos adaptativos de ecualización específicos. Además, ya que los sistemas de audio inmersivos requieren usar normalmente múltiples trasductores, se debe considerar también el problema de ecualización multicanal adaptativa cuando se diseñan nuevas estrategias de filtrado adaptativo para sistemas monocanal, ya que éstas deben ser fácilmente extrapolables al caso multicanal. Por otro lado, cuando se utilizan dispositivos acústicos, se debe considerar la existencia de no linearidades en el sistema elactroacústico, para poder ecualizarlo correctamente. Por este motivo, se propone un nuevo modelo no lineal de filtrado-x que compense a la vez la reverberación introducida por la sala y la distorsión no lineal con memoria provocada por el amplificador y el altavoz. Por último, es importante validar los algoritmos propuestos mediante implementaciones en tiempo real, para asegurarnos que pueden realizarse. Para ello, se presentan algunos resultados experimentales iniciales que muestran la idoneidad de la ecualización adaptativa en problemas de compensación de salas. / [CAT] Els sistemes d'àudio actuals es dissenyen amb l'objectiu de crear ambients reals i immersius que permeten a l'usuari experimentar una sensació acústica particular que no depèn de la sala on està percebent el so. No obstant això, els dispositius acústics i els sistemes de renderització multicanal treballant dins d'una sala poden arribar a modificar l'efecte global de l'àudio i per tant, l'efecte 3D del so a l'espai. Amb l'objectiu de conservar les característiques espacials del so obtingut amb tècniques de renderització multicanal, aquesta tesi doctoral presenta esquemes de filtrat adaptatiu per a compensar aquests efectes electroacústics i aconseguir una sensació immersiva del sistema acústic desitjat. El filtrat adaptatiu presenta una solució al problema d'equalització de sales que es interessant baix dos punts de vista. Per una banda, el filtrat adaptatiu resol de forma iterativa el problema inversió de sales, que pot arribar a ser molt complexe computacionalment quan s'utilitzen mètodes directes. Per altra banda, l'ús de filtres adaptatius permet fer un seguiment de les condicions canviants de la sala amb el temps. Més concretament, els filtres d'equalització adaptatius (EA) intenten cancel·lar els ecos produïts per la sala. A aquesta tesi, considerem aquest problema i proposem esquemes lineals efectius i robustos per a resoldre aquest problema d'equalització mitjançant filtres adaptatius. Per aconseguir-ho, diferent esquemes de filtrat adaptatiu es presenten dins del context del problema d'EA. Aquests esquemes de filtrat es basen en tres estratègies ja presentades a l'estat de l'art: la combinació convexa de filtres, el sesgat dels pesos del filtre i el filtrat basat en blocs. Més concretament, i motivat per la naturalesa dispersa de la resposta a l'impuls acústica i el corresponent filtre òptim invers, presentem diferents algorismes d'equalització adaptativa. A més a més, com que els sistemes d'àudio immersiu normalment requereixen l'ús de múltiples transductors, cal considerar també el problema d'equalització adaptativa multicanal quan es presenten noves solucions de canal simple, ja que aquestes s'han de poder estendre fàcilment al cas multicanal. Un altre aspecte a considerar quan es treballa amb dispositius d'àudio és el de les no linealitats del sistema a l'hora d'equalitzar correctament el sistema electroacústic. Amb aquest objectiu, a aquesta tesi es proposa una nova tècnica basada en filtrat-x no lineal, per a compensar tant la reverberació de la sala com la distorsió no lineal amb memòria introduïda per l'amplificador i els altaveus. Per últim, és important validar la implementació en temps real dels algorismes proposats. Amb aquest objectiu, alguns resultats inicials demostren la idoneïtat de l'equalització adaptativa en problemes de compensació de sales. / Fuster Criado, L. (2015). Linear and nonlinear room compensation of audio rendering systems [Tesis doctoral no publicada]. Universitat Politècnica de València. https://doi.org/10.4995/Thesis/10251/59459 / TESIS

The thesis deals with the biological data averaging applied to a periodical and repetitive signal, specifically to an ECG signals. There were used signals from MIT-BIH Arrhythmia database and ÚBMI database. Averaging was realized with constant, floating and exponential Windows, where was used the method of addition of the filtered residue. This method is intended to capture the slow variations from the input to the output signal. The outcomes of these methods can be used as a basis for further work, or function as an example of principled methods. Methods and its outcomes were created in Matlab.

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