CoordenaÃÃo de AperfeiÃoamento de Pessoal de NÃvel Superior / Este trabalho apresenta estudos sobre sistemas de comunicaÃÃo cujos sinais utilizados para a transmissÃo das informaÃÃes sÃo nÃo-ortogonais, superpostos em frequÃncia, e com espaÃamento entre portadoras menor do que a taxa de sÃmbolo. As pesquisas estÃo direcionadas na obtenÃÃo de estruturas de transmissor e receptor Ãtimos e sub-Ãtimos, na modelagem e anÃlise matemÃtica dos sistemas incluindo o canal, em propostas de estratÃgias para detecÃÃo de sÃmbolo, e na avaliaÃÃo de desempenho. SÃo propostas sete estratÃgias de recepÃÃo de N sinais m-QAM nÃo-ortogonais atravÃs do canal AWGN. Dentre as estratÃgias de detecÃÃo duas sÃo Ãtimas e as outras cinco sÃo subÃtimas. As duas estruturas de receptores Ãtimos apresentados neste trabalho sÃo: o receptor de mÃxima verossimilhanÃa (ML) clÃssico e o receptor de mÃxima verossimilhanÃa com base na decomposiÃÃo de Gram-Schmidt. Os receptores sub-Ãtimos desenvolvidos neste trabalho sÃo de dois tipos: receptores com equalizaÃÃo linear e receptores com equalizaÃÃo nÃo-linear. O primeiro tipo de receptor à desenvolvido com base nos critÃrios de erro quadrÃtico mÃdio mÃnimo (MMSE) e o de forÃagem a zero (ZF). à apresentado o desenvolvimento analÃtico do projeto de cada uma das arquiteturas de receptores lineares, assim como à determinado o erro dos estimadores. Os receptores com equalizaÃÃo nÃo-linear sÃo baseados no cancelamento de interferÃncia sucessiva (SIC). Neste trabalho, à proposta uma modificaÃÃo no algoritmo do SIC original, resultando em uma nova arquitetura de equalizaÃÃo. O desempenho dos receptores propostos à avaliado em diferentes condiÃÃes de nÃmero de portadoras e de grau de superposiÃÃo espectral atravÃs de simulaÃÃo computacional. Por fim, sÃo apresentados as conlusÃes e as perspectivas futuras de pesquisa. / This work presents studies on communication systems, whose signals used for transmission of information are non-orthogonal, overlapping in frequency and carrier spacing less than the rate of symbols. The research is aimed to obtain structures of transmitter, optimal and sub-optimal receivers using modeling and mathematical analysis of systems including the channel. Furthermore, propose strategies for symbol detection and performance evaluation. Seven strategies of reception to N signals m-QAM non-orthogonal through the AWGN channel. Among the strategies of detection two are optimal and the others five are suboptimal. The two optimal receivers structures presented in this paper are: the classical receiver maximum likelihood (ML) receiver and maximum likelihood based on the Gram-Schmidt decomposition. The suboptimal receivers in this work are of two types: receivers with linear and nonlinear equalization. The first type of receiver is developed based on the criteria of minimum mean square error (MMSE) and the zero forcing (ZF). It is presented the development of analytical design of each linear receiver architectures, as well as determined the error of the estimators. The receivers with nonlinear equalization are based on successive interference cancellation (SIC). In this paper, we propose a modification to the original algorithm of SIC, resulting in a new architecture of equalization. The performance of the proposed receivers is evaluated under different number of carriers and the degree of spectral overlap using computer simulation. Finally, we present the conclusions of this work and future prospects of the research.

Introdução: O inquérito por telefone, quando comparado ao inquérito domiciliar possui vários atrativos, em especial baixo custo operacional e rapidez do processo de divulgação de resultados. No entanto, a exclusão de domicílios sem telefone fixo, pode representar série questão de validade nas estimativas obtidas. Objetivo: Avaliar vícios potenciais nos resultados divulgados no Sistema de Vigilância de Fatores de Risco para Doenças Crônicas por Inquérito Telefônico (VIGITEL) em município de baixa cobertura de domicílios com telefone fixo. Métodos: A partir de resultados levantados pelo Inquérito Domiciliar realizado no município de Rio Branco-AC, com cobertura de 41 por cento dos domicílios com telefone fixo, tentou-se localizar vícios introduzidos nos resultados do Vigitel. Foi usado método alternativo de ponderação para diminuir o vício da estimativa do Vigitel. Resultados: O Vigitel subestima a maioria das prevalências estimadas. Os pesos de pós-estratificação eliminam parcialmente o vício, cuja origem é proveniente de baixa taxa de cobertura de domicílios com telefone fixo. Por outro lado, o uso desses pesos, quando não necessário, potencializou o vício das variáveis não associadas à posse de telefone fixo. Conclusões: Em municípios de baixa taxa de cobertura de domicílios com telefone fixo, torna-se necessária a implementação de novo método de ponderação e estratégia de seleção de variáveis externas para construção dos pesos de pós estratificação, que minimizem o vício nas estimativas das variáveis levantadas / Introdução: O inquérito por telefone, quando comparado ao inquérito domiciliar possui vários atrativos, em especial baixo custo operacional e rapidez do processo de divulgação de resultados. No entanto, a exclusão de domicílios sem telefone fixo, pode representar série questão de validade nas estimativas obtidas. Objetivo: Avaliar vícios potenciais nos resultados divulgados no Sistema de Vigilância de Fatores de Risco para Doenças Crônicas por Inquérito Telefônico (VIGITEL) em município de baixa cobertura de domicílios com telefone fixo. Métodos: A partir de resultados levantados pelo Inquérito Domiciliar realizado no município de Rio Branco-AC, com cobertura de 41 por cento dos domicílios com telefone fixo, tentou-se localizar vícios introduzidos nos resultados do Vigitel. Foi usado método alternativo de ponderação para diminuir o vício da estimativa do Vigitel. Resultados: O Vigitel subestima a maioria das prevalências estimadas. Os pesos de pós-estratificação eliminam parcialmente o vício, cuja origem é proveniente de baixa taxa de cobertura de domicílios com telefone fixo. Por outro lado, o uso desses pesos, quando não necessário, potencializou o vício das variáveis não associadas à posse de telefone fixo. Conclusões: Em municípios de baixa taxa de cobertura de domicílios com telefone fixo, torna-se necessária a implementação de novo método de ponderação e estratégia de seleção de variáveis externas para construção dos pesos de pós estratificação, que minimizem o vício nas estimativas das variáveis levantadas

Submitted by Marlene Santos (marlene.bc.ufg@gmail.com) on 2016-08-29T19:18:54Z No. of bitstreams: 2 Dissertação - Diego Fernando Burgos Beltrán - 2015.pdf: 4368650 bytes, checksum: 0eb56ef14323cf94881fc293cfa9be70 (MD5) license_rdf: 0 bytes, checksum: d41d8cd98f00b204e9800998ecf8427e (MD5) / Approved for entry into archive by Luciana Ferreira (lucgeral@gmail.com) on 2016-08-30T11:24:57Z (GMT) No. of bitstreams: 2 Dissertação - Diego Fernando Burgos Beltrán - 2015.pdf: 4368650 bytes, checksum: 0eb56ef14323cf94881fc293cfa9be70 (MD5) license_rdf: 0 bytes, checksum: d41d8cd98f00b204e9800998ecf8427e (MD5) / Made available in DSpace on 2016-08-30T11:24:57Z (GMT). No. of bitstreams: 2 Dissertação - Diego Fernando Burgos Beltrán - 2015.pdf: 4368650 bytes, checksum: 0eb56ef14323cf94881fc293cfa9be70 (MD5) license_rdf: 0 bytes, checksum: d41d8cd98f00b204e9800998ecf8427e (MD5) Previous issue date: 2015-07-06 / Outro / The continuous technological advances in the areas of electronics and programming made the signal processing techniques much easier to implement, allowing them to be incorporated in the communication systems, improving their performance. This work approaches the problem of estimating direction of arrival or angle of incidence (DOA) of electromagnetic wave fronts of a linear antenna array, and of beamforming of the array. Among the various techniques that exist in the literature, the Least Mean Squared algorithm (LMS) is a deterministic method that stands out for its simplicity, ease of implementation and the tendency to find local minima. On the other hand, the Genetic Algorithm (GA) is a heuristic method that ensures more comprehensive exploration possibilities avoiding the tendency of sticking to local minima, but offering greater difficulty of implementation, and higher computational complexity. The recently proposed Compact Genetic Algorithm (cGA) is a tool that shares all the virtues of GA, but without requiring the large computational cost that a GA entails. Since this method has not yet been used for controlling antenna arrays, this paper proposes to use it as the estimation of DOA and beamforming, in addition to enhance it with a number of modifications to make it more robust and more complete, though making it computationally heavier. This work presents simulations where the proposed adaptive receiver is evaluated under different scenarios of signal to noise ratio (SNR), number of interfering sources and convergence velocity. Moreover, moving users tracking situations are simulated, where the receiver's ability to adapt its radiation pattern is tested. All tests were done in the code division multiple access (CDMA) environment, where the only information available to the receiver are the sources spreading codes. To verify the operation of the cGA, its performance was compared with that of the LMS algorithm simulation under the same simulation conditions. The development of this thesis allowed to publish the articles named Adaptive Beamforming for Moving Targets Using Genetic Algorithms and a CDMA Reference Signal in the IEEE Colombian Conference on Communications and Computing COLCOM 2015, and Adaptive Beamforming for Moving Targets Using Genetic Algorithms in the IEEE Workshop on Engineering Applications WEA 2015 – International Congress on Engineering. The last one was accepted as an extended version to be publish in the magazine INGENIERÍA that belongs to the Distrital Francisco José de Caldas University in Bogotá, Colombia. / Os contínuos avanços tecnológicos nas áreas da eletrônica e da programação tornaram as técnicas de processamento de sinais muito mais fáceis de implementar, permitindo a incorporação delas nos sistemas de comunicação, melhorando a performance destes. Neste trabalho desenvolve-se o problema de estimação da direção de chegada ou ângulo de incidência (DOA) de frentes de ondas eletromagnéticas sobre um arranjo linear de antenas, além da conformação de feixe (beamforming) do arranjo. Dentre as diversas técnicas existentes na literatura, o algoritmo de Mínima Média Quadrática (LMS, do inglês Least Mean Squared) é um método determinístico que se destaca por sua simplicidade, facilidade de implementação e a tendência de encontrar mínimos locais como resposta. Por outro lado, o Algoritmo Genético (AG) é um método heurístico que garante uma exploração mais completa de possibilidades evitando a tendência de cair em mínimos locais, mas oferecendo uma maior dificuldade de implementação, além de maior complexidade computacional. Recentemente, foi proposto o Algoritmo Genético Compacto (AGC), que é uma ferramenta que compartilha todas as virtudes dos Algoritmos Genéticos, porém sem exigir o grande custo computacional que um AG implica. Como este método ainda não foi utilizado para o controle de arranjos de antenas, este trabalho propõe utilizá-lo na estimação da DOA e beamforming, além de agregar-lhe uma série de modificações a fim de torná-lo mais robusto e mais completo, apesar de computacionalmente mais pesado. Neste trabalho exibe-se simulações em que o receptor adaptativo proposto é avaliado sob diferentes situações de relação sinal ruído (SNR), quantidade de fontes interferentes e velocidade de convergência. Além disso, simulam-se situações de rastreamento de usuários em movimento, onde é posta à prova a capacidade do receptor adaptar seu diagrama de radiação. Todos os testes foram feitos no ambiente de multiplicidade de acesso via divisão por códigos (CDMA), onde a única informação disponível no receptor são os códigos de espalhamento das fontes. Para conferir o funcionamento do AGC, comparou-se seu desempenho com aquele do algoritmo LMS sob as mesmas condições de simulação. O desenvolvimento desta tese permitiu a publicação dos artigos Adaptive Beamforming for Moving Targets Using Genetic Algorithms and a CDMA Reference Signal no IEEE Colombian Conference on Communications and Computing COLCOM 2015 e Adaptive Beamforming for Moving Targets Using Genetic Algorithms no IEEE Workshop on Engineering Applications WEA 2015 – International Congress on Engineering, este ultimo foi aceito para ser publicado como uma versão estendida na revista INGENIERÍA da universidade Distrital Francisco José de Caldas de Bogotá, Colômbia.

We address the problem of image denoising for additive white Gaussian noise (AWGN), Poisson noise, and Chi-squared noise scenarios. Thermal noise in electronic circuitry in camera hardware can be modeled as AWGN. Poisson noise is used to model the randomness associated with photon counting during image acquisition. Chi-squared noise statistics are appropriate in imaging modalities such as Magnetic Resonance Imaging (MRI). AWGN is additive, while Poisson noise is neither additive nor multiplicative. Although Chi-squared noise is derived from AWGN statistics, it is non-additive. Mean-square error (MSE) is the most widely used metric to quantify denoising performance. In parametric denoising approaches, the optimal parameters of the denoising function are chosen by employing a minimum mean-square-error (MMSE) criterion. However, the dependence of MSE on the noise-free signal makes MSE computation infeasible in practical scenarios. We circumvent the problem by adopting an MSE estimation approach. The ground-truth-independent estimates of MSE are Stein’s unbiased risk estimate (SURE), Poisson unbiased risk estimate (PURE) and Chi-square unbiased risk estimate (CURE) for AWGN, Poison and Chi-square noise models, respectively. The denoising function is optimized to achieve maximum noise suppression by minimizing the MSE estimates. We have chosen the bilateral filter as the denoising function. Bilateral filter is a nonlinear edge-preserving smoother. The performance of the bilateral filter is governed by the choice of its parameters, which can be optimized to minimize the MSE or its estimate. However, in practical scenarios, MSE cannot be computed due to inaccessibility of the noise-free image. We derive SURE, PURE, and CURE in the context of bilateral filtering and compute the parameters of the bilateral filter that yield the minimum cost (SURE/PURE/CURE). On processing the noisy input with bilateral filter whose optimal parameters are chosen by minimizing MSE estimates (SURE/PURE/CURE), we obtain the estimate closest to the ground truth. We denote the bilateral filter with optimal parameters as SURE-optimal bilateral filter (SOBF), PURE-optimal bilateral filter (POBF) and CURE-optimal bilateral filter (COBF) for AWGN, Poisson and Chi-Squared noise scenarios, respectively. In addition to the globally optimal bilateral filters (SOBF and POBF), we propose spatially adaptive bilateral filter variants, namely, SURE-optimal patch-based bilateral filter (SPBF) and PURE-optimal patch-based bilateral filter (PPBF). SPBF and PPBF yield significant improvements in performance and preserve edges better when compared with their globally-optimal counterparts, SOBF and POBF, respectively. We also propose the SURE-optimal multiresolution bilateral filter (SMBF) where we couple SOBF with wavelet thresholding. For Poisson noise suppression, we propose PURE-optimal multiresolution bilateral filter (PMBF), which is the Poisson counterpart of SMBF. We com-pare the performance of SMBF and PMBF with the state-of-the-art denoising algorithms for AWGN and Poisson noise, respectively. The proposed multiresolution-based bilateral filtering techniques yield denoising performance that is competent with that of the state-of-the-art techniques.

Today, Sport Utility Vehicles are marketed as both on-road and off-road vehicles. This results in a compromise when designing the suspension of the vehicle. If the suspension characteristics are fixed, the vehicle cannot have good handling capabilities on highways and good ride comfort over rough terrain. The rollover propensity of this type of vehicle compared to normal cars is high because it has a combination of a high centre of gravity and a softer suspension. The 4 State Semi-active Suspension System (4S4) that can switch between two discrete spring characteristics as well as two discrete damper characteristics, has been proven to overcome this compromise. The soft suspension setting (soft spring and low damping) is used for ride comfort, while the hard suspension setting (stiff spring and high damping) is used for handling. The following question arises: when is which setting most appropriate? The two main contributing factors are the terrain profile and the driver’s actions. Ride comfort is primarily dependant on the terrain that the vehicle is travelling over. If the terrain can be identified, certain driving styles can be expected for that specific environment. The terrains range from rough and uncomfortable to smooth with high speed manoeuvring. Terrain classification methods are proposed and tested with measured data from the test vehicle on known terrain types. Good results were obtained from the terrain classification methods. Five terrain types were accurately identified from over an hour’s worth of vehicle testing. Handling manoeuvres happen unexpectedly, often to avoid an accident. To improve the handling and therefore safety of the vehicle, the 4S4 can be switched to the hard suspension setting, which results in a reduced body roll angle. This decision should be made quickly with the occupants’ safety as the priority. Methods were investigated that will determine when to switch the suspension to the handling mode based on the kinematics of the vehicle. The switching strategies proposed in this study have the potential, with a little refinement, to make the ride versus handling decision correctly. Copyright 2007, University of Pretoria. All rights reserved. The copyright in this work vests in the University of Pretoria. No part of this work may be reproduced or transmitted in any form or by any means, without the prior written permission of the University of Pretoria. Please cite as follows: Bester, R 2007, The ride comfort versus handling decision for off-road vehicles, MEng dissertation, University of Pretoria, Pretoria, viewed yymmdd < http://upetd.up.ac.za/thesis/available/etd-10252007-111611 / > / Dissertation (MEng (Mechanical Engineering))--University of Pretoria, 2007. / Mechanical and Aeronautical Engineering / unrestricted

Dans les systèmes de communications sans fil, le canal de transmission entre les antennes d’émission et de réception est l’une des principales sources de perturbation pour le signal. Les modulations multiporteuses, telles que l’OFDM (pour orthogonal frequency division multiplexing), sont très robustes contre l’effet des multi-trajets, et permet de retrouver le signal émis avec un faible taux d’erreur, quand elles sont combinées avec un codage canal. L’estimation de canal joue alors un rôle clé dans les performances des systèmes de communications. Dans cette thèse, on étudie des techniques fondées sur les estimateurs LS (pour least square, ou moindres carrés) et MMSE (pour minimum mean square error, ou erreur quadratique moyenne minimum). La technique MMSE est optimale, mais est beaucoup plus complexe que LS, et nécessite la connaissance a priori des moments de second ordre du canal et du bruit. Dans cette présentation, deux méthodes permettant d’atteindre des performances proches de LMMSE en évitant ses inconvénients sont étudiées. Une troisième partie étudie quant à elle les erreurs d’estimation dues aux interpolations. / In wireless communications systems, the transmission channel between the transmitter and the receiver antennas is one of the main sources of disruption for the signal. The multicarrier modulations, such as the orthogonal frequency division multiplexing (OFDM), are very robust against the multipath effect, and allow to recover the transmitted signal with a low error rate, when they are combined with a channel encoding. The channel estimation then plays a key role in the performance of the communications systems. In this PhD thesis, we study techniques based on least square (LS) and minimum mean square error (MMSE) estimators. The MMSE is optimal, but is much more complex than LS, and requires the a priori knowledge of the second order moment of the channel and the noise. In this presentation, two methods that allow to reach a performance close to the one of LMMSE while getting around its drawback are investigated. In another way, a third part of the presentation investigates the errors of estimation due to the interpolations.

[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

abstract: Signal compressed using classical compression methods can be acquired using brute force (i.e. searching for non-zero entries in component-wise). However, sparse solutions require combinatorial searches of high computations. In this thesis, instead, two Bayesian approaches are considered to recover a sparse vector from underdetermined noisy measurements. The first is constructed using a Bernoulli-Gaussian (BG) prior distribution and is assumed to be the true generative model. The second is constructed using a Gamma-Normal (GN) prior distribution and is, therefore, a different (i.e. misspecified) model. To estimate the posterior distribution for the correctly specified scenario, an algorithm based on generalized approximated message passing (GAMP) is constructed, while an algorithm based on sparse Bayesian learning (SBL) is used for the misspecified scenario. Recovering sparse signal using Bayesian framework is one class of algorithms to solve the sparse problem. All classes of algorithms aim to get around the high computations associated with the combinatorial searches. Compressive sensing (CS) is a widely-used terminology attributed to optimize the sparse problem and its applications. Applications such as magnetic resonance imaging (MRI), image acquisition in radar imaging, and facial recognition. In CS literature, the target vector can be recovered either by optimizing an objective function using point estimation, or recovering a distribution of the sparse vector using Bayesian estimation. Although Bayesian framework provides an extra degree of freedom to assume a distribution that is directly applicable to the problem of interest, it is hard to find a theoretical guarantee of convergence. This limitation has shifted some of researches to use a non-Bayesian framework. This thesis tries to close this gab by proposing a Bayesian framework with a suggested theoretical bound for the assumed, not necessarily correct, distribution. In the simulation study, a general lower Bayesian Cram\'er-Rao bound (BCRB) bound is extracted along with misspecified Bayesian Cram\'er-Rao bound (MBCRB) for GN model. Both bounds are validated using mean square error (MSE) performances of the aforementioned algorithms. Also, a quantification of the performance in terms of gains versus losses is introduced as one main finding of this report. / Dissertation/Thesis / Masters Thesis Computer Engineering 2019

SAAB has developed an autonomous underwater vehicle that can mimic a conventional submarine for military fleets to exercise anti-submarine warfare. The AUV actively emits amplified versions of received sonar pulses to create the illusion of being a larger object. To prevent acoustic feedback, the AUV must distinguish between the sound to be actively responded to and its emitted signal. This master thesis has examined techniques aimed at preventing the AUV from responding to previously emitted signals to avoid acoustical feedback, without relying on prior knowledge of either the received signal or the signal emitted by the AUV. The two primary types of algorithms explored for this problem include blind source separation and adaptive filtering. The adaptive filters based on Leaky Least Mean Square and Kalman have shown promising results in attenuating the active response from the received signal. The adaptive filters utilize the fact that a certain hydrophone primarily receives the active response. This hydrophone serves as an estimate of the active response since the signal it captures is considered unknown and is to be removed. The techniques based on blind source separation have utilized the recordings of three hydrophones placed at various locations of the AUV to separate and estimate the received signal from the one emitted by the AUV. The results have demonstrated that neither of the reviewed methods is suitable for implementation on the AUV. The hydrophones are situated at a considerable distance from each other, resulting in distinct time delays between the reception of the two signals. This is usually referred to as a convolutive mixture. This is commonly solved using the frequency domain to transform the convolutive mixture to an instantaneous mixture. However, the fact that the signals share the same frequency spectrum and are adjacent in time has proven highly challenging.

Belastningsskador hos arbetare är ett ökande problem hos olika företag och det har visat sig finnas en tydlig koppling mellan dessa skador och handledens vinkelhastigheten. Det är därför av stort intresse att kunna mäta dessa vinkelhastigheter på ett noggrant och smidigt sätt. Syftet med denna rapport är att utvärdera precisionen av IMU-sensorers förmåga att beräkna vinkelhastigheten av handleden. Detta görs genom att jämföra data från IMU-sensorer med data från ett optiskt spårningssystem (OTS), vilket klassas som en gold standard inom detta område. Ett experiment bestående av åtta övningar utfördes: tre standard rörelser (flexion och rotation i takterna 40, 90 och 140 slag per minut) och fyra simulerade arbeten (målning, pappersvikning, datorarbete och hårföning). Grad av överensstämmelse ges av 1,96 standardavvikelser (SD) för standardrörelserna (10 deltagare) vilka var -31,8 grader/s och 34,2 grader/s, medan för de simulerade arbetena var det -35,1 grader/s och 34,2 grader/s. Det lägsta medelvärdet av medelkvadratavvikelse (RMSD) var 15,7 grader/s och erhölls vid 40 BPM medan den högsta medelvärdet var 93,9 grader/s och erhölls vid målningsövningen. Medelvärdet av korrelationskoefficienten mellan IMU-sensorer och OTS varierade mellan 0,97 och 0,42 och korrelationskoefficienterna av deltagarnas 50:e percentiler av vinkelhastigheten var 0,95 för standardrörelserna och 0,96 för de simulerade arbetena. Medelvärdet av absoluta differensen mellan sensorer och OTS var givet i percentiler (10:e, 50:e och 90:e). Det största spannet för 50:e percentilen gavs vid 140 BPM (18,3 ± 24,6) och det minsta spannet vid 40 BPM (3,5 ± 4,7). Trots att det fanns mindre differenser mellan metodernas mätningar av vinkelhastighet, anser vi att IMU-sensorer har potential att användas för att mäta vinkelhastigheter hos handledens och med vidare utveckling kan den nuvarande differensen minimeras. / Musculoskeletal disorders (MSDs) are increasingly frequent amongst workers and there is a clear connection between work injuries and wrist angular velocities. One of the biggest issues therefore is the currently limited availability of means to measure these angular velocities. The aim of this study is to validate the usability of the IMU sensors to measure angular velocities. This is done by comparing the data from the IMU:s with the data obtained with the optical motion tracking system (OTS), which is considered gold standard within this field of studies. An experiment consisting of eight exercises was conducted: three standard movements (flexion and rotation in the pace 40, 90 and 140 repetitions per minute) and four simulated practical work tasks (painting, folding paper, computer exercise and using a hairdryer). The limits of agreement for the standard movements (10 subjects) were -31,8 degrees/s and 34,2 degrees/s, whereas for the simulated practical work tasks they were -35,1 degrees/s and 28,2 degrees/s. The lowest mean value of the root mean square deviation (RMSD) value was 15,7 degrees/s which represents the 40 BPM task whilst the highest mean value was 93,9 degrees/s which correspond to the painting task. The mean value of the correlation coefficients between the IMU:s and the OTS ranged between 0,97 and 0,42 and the correlation coefficient between the subjects 50:th percentiles of the angular velocity, was 0,95 for the standard movements whilst for the practical work tasks it was 0,96. The mean value of the absolute difference between the sensors and the OTS was given in percentiles (10th, 50th and 90th). The largest range within the 50th percentile occurred during the 140 BPM task (18,3 ± 24,6) and the smallest range during the 40 BPM task (3,5 ± 4,7). Although the measured angular velocities vary to a certain extent between the two methods, we conclude that the IMU sensors present the potential to work as measuring units for wrist angular velocities and with further development the current differences can be minimized. / Forte dnr: 2017-01209 "Enkel och tideffektiv metod att mät, analysera och presentera biomekaniskbelastning för hand-handled"