<p>Recent advances in technology have allowed researchers to collect high-dimensional biological data simultaneously. In genomic studies, for instance, measurements from tens of thousands of genes are taken from individuals across several experimental groups. In time course microarray experiments, gene expression is measured at several time points for each individual across the whole genome resulting in massive amount of data. In such experiments, researchers are faced with two types of high-dimensionality. The first is global high-dimensionality, which is common to all genomic experiments. The global high-dimensionality arises because inference is being done on tens of thousands of genes resulting in multiplicity. This challenge is often dealt with statistical methods for multiple comparison, such as the Bonferroni correction or false discovery rate (FDR). We refer to the second type of high-dimensionality as gene specific high-dimensionality, which arises in time course microarry experiments due to the fact that, in such experiments, sample size is often smaller than the number of time points ($n</p> <p>In this thesis, we use the growth curve model (GCM), which is a generalized multivariate analysis of variance (GMANOVA) model, and propose a moderated test statistic for testing a special case of the general linear hypothesis, which is specially useful for identifying genes that are expressed. We use the trace test for the GCM and modify it so that it can be used in high-dimensional situations. We consider two types of moderation: the Moore-Penrose generalized inverse and Stein's shrinkage estimator of $ S $. We performed extensive simulations to show performance of the moderated test, and compared the results with original trace test. We calculated empirical level and power of the test under many scenarios. Although the focus is on hypothesis testing, we also provided moderated maximum likelihood estimator for the parameter matrix and assessed its performance by investigating bias and mean squared error of the estimator and compared the results with those of the maximum likelihood estimators. Since the parameters are matrices, we consider distance measures in both power and level comparisons as well as when investigating bias and mean squared error. We also illustrated our approach using time course microarray data taken from a study on Lung Cancer. We were able to filter out 1053 genes as non-noise genes from a pool of 22,277 genes which is approximately 5\% of the total number of genes. This is in sync with results from most biological experiments where around 5\% genes are found to be differentially expressed.</p> / Master of Science (MSc)

[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. / [CA] 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]. Universitat Politècnica de València. https://doi.org/10.4995/Thesis/10251/59459

Stochastic differential equations(SDEs) play an important role in many branches of engineering and science including economics, finance, chemistry, biology, mechanics etc. SDEs (with m-dimensional Wiener process) arising in many applications do not have explicit solutions, which implies the development of effective numerical methods for such systems. For SDEs, one can classify the numerical methods into three classes: fully implicit methods, semi-implicit methods and explicit methods. In order to solve SDEs, the computation of Newton iteration is necessary for the implicit and semi-implicit methods whereas for the explicit methods we do not need such computation. In this thesis the common theme is to construct explicit numerical methods with strong order 1.0 and 1.5 for solving Itˆo SDEs. The five-stage Milstein(FSM)methods, split-step forward Milstein(SSFM)methods and M-stage split-step strong Taylor(M-SSST) methods are constructed for solving SDEs. The FSM, SSFM and M-SSST methods are fully explicit methods. It is proved that the FSM and SSFM methods are convergent with strong order 1.0, and M-SSST methods are convergent with strong order 1.5.Stiffness is a very important issue for the numerical treatment of SDEs, similar to the case of deterministic ordinary differential equations. Stochastic stiffness can lead someone to use smaller step-size for the numerical simulation of the SDEs. However, such issues can be handled using numerical methods with better stability properties. The analysis of stability (with multidimensional Wiener process) shows that the mean-square stable regions of the FSM methods are unbounded. The analysis of stability shows that the mean-square stable regions of the FSM and SSFM methods are larger than the Milstein and three-stage Milstein methods. The M-SSST methods possess large mean square stability region as compared to the order 1.5 strong Itˆo-Taylor method. SDE systems simulated with the FSM, SSFM and M-SSST methods show the computational efficiency of the methods. In this work, we also consider the problem of computing numerical solutions for stochastic delay differential equations(SDDEs) of Itˆo form with a constant lag in the argument. The fully explicit methods, the predictor-corrector Euler(PCE)methods, are constructed for solving SDDEs. It is proved that the PCE methods are convergent with strong order γ = ½ in the mean-square sense. The conditions under which the PCE methods are MS-stable and GMS-stable are less restrictive as compared to the conditions for the Euler method.

Submitted by Renata Lopes (renatasil82@gmail.com) on 2016-02-11T10:56:22Z No. of bitstreams: 1 juandavidvalenciapayan.pdf: 2769602 bytes, checksum: f450fbb83bbe8b4e5a8cdb7c9b4fa338 (MD5) / Approved for entry into archive by Adriana Oliveira (adriana.oliveira@ufjf.edu.br) on 2016-02-26T11:57:12Z (GMT) No. of bitstreams: 1 juandavidvalenciapayan.pdf: 2769602 bytes, checksum: f450fbb83bbe8b4e5a8cdb7c9b4fa338 (MD5) / Made available in DSpace on 2016-02-26T11:57:12Z (GMT). No. of bitstreams: 1 juandavidvalenciapayan.pdf: 2769602 bytes, checksum: f450fbb83bbe8b4e5a8cdb7c9b4fa338 (MD5) Previous issue date: 2014-03-27 / CAPES - Coordenação de Aperfeiçoamento de Pessoal de Nível Superior / Esta contribuição descreve uma caracterização estatística da rede de baixa tensão Brasileira residencial como meio de comunicação. As discussões são baseadas em canais estimados obtidos em uma campanha de medição realizada em quatro apartamentos diferentes, com tamanhos que variam de 50 até 90 metros quadrados. Os parâmetros considerados para esta análise são o Root Mean Square Delay Spread, o ganho médio do canal e a capacidade do canal. Para efeitos de comparação com a rede de potencia dos Estados Unidos, a banda de frequência utilizada foi de 1:705 até 30 MHz. A análise relatada mostra que o Root Mean Square Delay Spread e o ganho médio do canal não podem ser modelados como variáveis log-normal. Os resultados obtidos geram dúvidas em relação aos atuais encontrados na literatura, em que se afirma que tanto o Root Mean Square Delay Spread quanto o ganho médio do canal seguem uma distribuição log-normal. Foram medidos também a impedância de acesso e o ruído do canal de comunicação via rede elétrica. Além disso, os conceitos de cooperação para melhorar o desempenho dos sistemas de comunicação via rede elétrica foram analisados, mais especificamente na rede de baixa tensão Brasileira residencial. Para isso, foram analisados a performance dos protocolos de Amplify-and-Forward e Decode-and-Forward, em conjunto com as técnicas de combinação Equal-Gain Combining, Selection-Combining e Maximal-Ratio Combining. A análise sobre os dados medidos cobriram uma faixa de frequência de 1:705 a 100 MHz. Os dados medidos abordam quatro cenários para possíveis localizações do nó Relay. Os resultados obtidos mostram que o Amplify-and- Forward é de aplicabilidade limitada no contexto de comunicação via rede elétrica e o oposto é válido para o protocolo de Decode-and-Forward, principalmente se a probabilidade de erro de detecção de símbolos no nó Relay tende a zero. / This thesis outlines a statistical characterization of the Brazilian In-Home Low-Voltage Electric Power Grid as a communication medium. The discussions are based on estimated channels obtained in a measurement campaign carried out in four different apartments with sizes ranging from 50 up to 90 square meters. The parameters considered for this analysis are the Root Mean Square Delay Spread, the average channel gain, and the channel capacity. For the sake of comparison with the Electric Power Grid in United States, the frequency band ranging from 1:705 up to 30 MHz was set. The reported analysis shows that the Root Mean Square Delay Spread and the average channel gain cannot be modeled as log-normal variables, this cast doubt the current results found in the literature, in which is stated that both the Root Mean Square Delay Spread and the average channel gain follow a log-normal distribution. This was followed by the Power Line Communication access impedance and noise measurements in the In-Home Low-Voltage Electric Power Grid. Additionally, the suitability of cooperation concepts for improving the performance of Power Line Communication systems was analyzed, more specifically in the Brazilian In-Home Low-Voltage Electric Power Grid. For this purpose, the performance of the Amplify-and-Forward and Decode-and- Forward protocols, together with the Equal-Gain Combining, Selection Combining, and Maximal-Ratio Combining techniques were analyzed. The analysis was carried out on the measured data covering a frequency band from 1:705 up to 100 MHz. The measured data addresses four scenarios for possible relay node locations. The attained results show that the Amplify-and-Forward is of limited applicability in the Power Line Communication context and the opposite is valid to the Decode-and-Forward protocol, mainly if the error probability of detecting symbols at the relay node is zero.

In developing an understanding of real-world problems, researchers develop mathematical and statistical models. Various model selection methods exist which can be used to obtain a mathematical model that best describes the real-world situation in some or other sense. These methods aim to assess the merits of competing models by concentrating on a particular criterion. Each selection method is associated with its own criterion and is named accordingly. The better known ones include Akaike's Information Criterion, Mallows' Cp and cross-validation, to name a few. The value of the criterion is calculated for each model and the model corresponding to the minimum value of the criterion is then selected as the "best" model. / Mathematical Sciences / M. Sc. (Statistics)

Η παρούσα μεταπτυχιακή διατριβή εντάσσεται ερευνητικά στην περιοχή της Στατιστικής Θεωρίας Αποφάσεων και ειδικότερα στην εκτίμηση των παραμέτρων στο μοντέλο της διπαραμετρικής εκθετικής κατανομής με παράμετρο θέσης μ και παράμετρο κλίμακος σ. Θεωρούμε το πρόβλημα εκτίμησης των παραμέτρων κλίμακας μ και θέσης σ, όταν μ≤c, όπου c είναι μία γνωστή σταθερά. Αποδεικνύουμε ότι σε σχέση με το κριτήριο του Μέσου Τετραγωνικού Σφάλματος (ΜΤΣ), οι βέλτιστοι αναλλοίωτοι εκτιμητές των μ και σ, είναι μη αποδεκτοί όταν μ≤c, και προτείνουμε βελτιωμένους. Επίσης συγκρίνουμε του εκτιμητές αυτούς σε σχέση με το κριτήριο του Pitman. Επιπλέον, προτείνουμε εκτιμητές που είναι καλύτεροι από τους βέλτιστους αναλλοίωτους εκτιμητές, όταν μ≤c, ως προς την συνάρτηση ζημίας LINEX. Τέλος, η θεωρία που αναπτύσσεται εφαρμόζεται σε δύο ανεξάρτητα δείγματα προερχόμενα από εκθετική κατανομή. / The present master thesis deals with the estimation of the location parameter μ and the scale parameter σ of the two-parameter exponential distribution. We consider the problem of estimation of locasion parameter μ and the scale parameter σ, when it is known apriori that μ≤c, where c is a known constant. We establish that with respect to the mean square error (mse) criterion the best affine estimators of μ and σ in the absence of information μ≤c are inadmissible and we propose estimators which are better than these estimators. Also, we compare these estimators with respect to the Pitman Nearness criterion. We propose estimators which are better than the standard estimators in the unrestricted case with respect to the suitable choise of LINEX loss. Finally, the theory developed is applied to the problem of estimating the location and scale parameters of two exponential distributions when the location parameters are ordered.

The proliferation of modern wireless networks increases demand for high capacity and throughput in order to provide faster, more robust, efficient and broadband services to end users. Mobile WiMAX and LTE are examples of such networks in which for some cases they have exposed limited connectivity due to harsh environment. Relay stations are preferred to overcome problems of weak or no access for such network devices, that are placed in specific positions to maintain high quality of data transfer at low cost and provide the required connectivity anywhere anytime. These stations should be equipped with an antenna system capable of establishing communication between base station (backhaul link) and end users (access link). This thesis focuses on the design and development of a new antenna system that is suitable for a relay-based wireless network. Planar geometries of microstrip patch antennas are utilized. The antenna system comprises two antenna modules: a new design of a single antenna for access link and a new design of an antenna array for backhaul link realization. Both antenna specifications are compatible with the IEEE802.16j protocol standard. Hence, relay station should be capable of pointing its radiation pattern to the base station antenna, thus to achieve the desired radiation pattern of the relay station, a new beam-forming module is proposed, designed and developed to generate the proper radiation pattern. The beam-forming module incorporating digital phase shifters and attenuator chips is fabricated and tested. The optimization process using the Least Mean Square (LMS) algorithm is considered in this study to assign the proper phase and amplitude that is necessary to each radiation element excitation current, to produce the desired steered radiation pattern. A comprehensive study on the coupling effects for several relative positions between two new backhaul and access link antenna elements is performed. Two new antenna configurations for coupling reduction are tested and the simulated and measured results in terms of antenna radiation performances were compared and commented.

Cette thèse porte sur l’étude du positionnement optimale des réseaux de capteurs pour la localisation de sources. Nous avons étudié deux approches: l’approche basée sur les performances de l’estimation en termes d’erreur quadratique moyenne et l’approche basée sur le seuil statistique de résolution (SSR).Pour le première approche, nous avons considéré les bornes inférieures de l’erreur quadratique moyenne qui sont utilisés généralement pour évaluer la performance d’estimation indépendamment du type d’estimateur considéré. Nous avons étudié deux types de bornes: la borne Cramér-Rao (BCR) pour le modèle où les paramètres sont supposés déterministes et la borne Weiss-Weinstein (BWW) pour le modèle où les paramètres sont supposés aléatoires. Nous avons dérivé les expressions analytiques de ces bornes pour développer des outils statistiques afin d’optimiser la géométrie des réseaux de capteurs. Par rapport à la BCR, la borne BWW peut capturer le décrochement de l’EQM des estimateurs dans la zone non-asymptotique. De plus, les expressions analytiques de la BWW pour un modèle Gaussien général à moyenne paramétré ou à covariance matrice paramétré sont donnés explicitement. Basé sur ces expressions analytiques, nous avons étudié l’impact de la géométrie des réseaux de capteurs sur les performances d’estimation en utilisant les réseaux de capteurs 3D et 2D pour deux modèles des observations concernant les signaux sources: (i) le modèle déterministe et (ii) le modèle stochastique. Nous en avons ensuite déduit des conditions concernant les propriétés d’isotropie et de découplage.Pour la deuxième approche, nous avons considéré le seuil statistique de résolution qui caractérise la séparation minimale entre les deux sources. Dans cette thèse, nous avons étudié le SSR pour le contexte Bayésien moins étudié dans la littérature. Nous avons introduit un modèle des observations linéarisé basé sur le critère de probabilité d’erreur minimale. Ensuite, nous avons présenté deux approches Bayésiennes pour le SSR, l’une basée sur la théorie de l’information et l’autre basée sur la théorie de la détection. Ces approches pourront être utilisée pour améliorer la capacité de résolution des systèmes. / This thesis deals with the array geometry optimization problem in the context of sources localization. We have considered two approaches for the array geometry optimization: the performance estimation in terms of mean square error approach and the statistical resolution limit (SRL) approach. In the first approach, the lower bounds on the mean square error which are usually used in array processing to evaluate the estimation performance independently of the considered estimator have been considered. We have investigated two kinds of lower bounds: the well-known Cramér-Rao bound (CRB) for the deterministic model in which the parameters are assumed to be deterministic, and the Weiss-Weinstein bound (WWB) which is less studied, for the Bayesian model, in which, the parameters are assumed to be random with some prior distributions. We have proposed closed-form expressions of these bounds, which can be used as a statistical tool for array geometry design. Compared to the CRB, the WWB can predict the threshold effect of the MSE in the non-asymptotic area. Moreover, the closed-form expressions of the WWB proposed for a general Gaussian model with parameterized mean or parameterized covariance matrix can also be useful for other problems. Based on these closed-form expressions, the 3D array geometry and the classical planar array geometry have been investigated under (i) the conditional observation model in which the source signal is modeled as a deterministic sequence and under (ii) the unconditional observation model in which the source signal is modeled as a Gaussian random process. Conditions concerning the isotropic and uncoupling properties were then derived.In the second approach, we have considered the statistical resolution limit which characterizes the minimal separation between the two closed spaced sources which still allows to determine correctly the number of sources. In this thesis, we are interested in the SRL in the Bayesian context which is less studied in the literature. Based on the linearized observation model with the minimum probability of error, we have introduced the two Bayesian approaches of the SRL based on the detection and information theories which could lead to some interesting tools for the system design.

Cette thèse porte sur l'analyse des performances en traitement du signal et se compose de deux parties: Premièrement, nous étudions les bornes inférieures dans la caractérisation et la prédiction des performances en termes d'erreur quadratique moyenne (EQM). Les bornes inférieures de l'EQM donne la variance minimale qu'un estimateur peut atteindre et peuvent être divisées en deux catégories: les bornes déterministes pour le modèle où les paramètres sont supposés déterministes (mais inconnus), et les bornes Bayésiennes pour le modèle où les paramètres sont supposés aléatoires. En particulier, nous dérivons les expressions analytiques de ces bornes pour deux applications différentes: (i) La première est la localisation des sources en utilisant un radar multiple-input multiple-output (MIMO). Nous considérons les bornes inférieures dans deux contextes c'est-à-dire avec ou sans erreurs de modèle. (ii) La deuxième est l'estimation de phase d'impulsion de pulsars à rayon X qui est une solution potentielle pour la navigation autonome dans l'espace. Pour cette application, nous avons calculé plusieurs bornes inférieures de l'EQM dans le contexte de données modélisées par une loi de Poisson (complétant ainsi les travaux disponibles dans la littérature où les données sont modélisées par une loi gaussienne). Deuxièmement, nous étudions le seuil statistique de résolution limite (SRL), qui est la distance minimale en termes des paramètres d'intérêts entre les deux signaux permettant de séparer / estimer correctement les paramètres d'intérêt. Plus précisément, nous dérivons le SRL dans deux contextes: le traitement d'antenne et le radar MIMO en utilisant deux approches basées sur la théorie de l'estimation et sur la théorie de l'information. Finalement, nous proposons des expressions compactes du SRL dans le cas d'erreurs de modèle. / This manuscript concerns the performance analysis in signal processing and consists into two parts : First, we study the lower bounds in characterizing and predicting the estimation performance in terms of mean square error (MSE). The lower bounds on the MSE give the minimum variance that an estimator can expect to achieve and it can be divided into two categories depending on the parameter assumption: the so-called deterministic bounds dealing with the deterministic unknown parameters, and the so-called Bayesian bounds dealing with the random unknown parameter. Particularly, we derive the closed-form expressions of the lower bounds for two applications in two different fields: (i) The first one is the target localization using the multiple-input multiple-output (MIMO) radar in which we derive the lower bounds in the contexts with and without modeling errors, respectively. (ii) The other one is the pulse phase estimation of X-ray pulsars which is a potential solution for autonomous deep space navigation. In this application, we show the potential universality of lower bounds to tackle problems with parameterized probability density function (pdf) different from classical Gaussian pdf since in X-ray pulse phase estimation, observations are modeled with a Poisson distribution. Second, we study the statistical resolution limit (SRL) which is the minimal distance in terms of the parameter of interest between two signals allowing to correctly separate/estimate the parameters of interest. More precisely, we derive the SRL in two contexts: array processing and MIMO radar by using two approaches based on the estimation theory and information theory. We also present in this thesis the usefulness of SRL in optimizing the array system.

Submitted by Rosivalda Pereira (mrs.pereira@ufma.br) on 2017-06-23T20:42:44Z No. of bitstreams: 1 CesarCasteloBranco.pdf: 11257769 bytes, checksum: 911c33f2f0ba5c1c0948888e713724f6 (MD5) / Made available in DSpace on 2017-06-23T20:42:44Z (GMT). No. of bitstreams: 1 CesarCasteloBranco.pdf: 11257769 bytes, checksum: 911c33f2f0ba5c1c0948888e713724f6 (MD5) Previous issue date: 2016-12-12 / Coordenação de Aperfeiçoamento de Pessoal de Nível Superior (CAPES) / Conselho Nacional de Desenvolvimento Científico e Tecnológico (CNPQ) / This work proposes new methodologies to optimize the choice of the parameters of the proportionate normalized least-mean-square (PNLMS) adaptive algorithms. The proposed approaches use procedures based on two optimization methods, namely, the golden section and tabu search methods. Such procedures are applied to determine the optimal parameters in each iteration of the adaptation process of the PNLMS and improved PNLMS (IPNLMS) algorithms. The objective function for the proposed procedures is based on the a posteriori estimation error. Performance studies carried out to evaluate the impact of the PNLMS and IPNLMS parameters in the behavior of these algorithms shows that, with the aid of optimization techniques to choose properly such parameters, the performance of these algorithms may be improved in terms of convergence speed for the identification of plants with high sparseness degree. The main goal of the proposed methodologies is to improve the distribution of the adaptation energy between the coefficients of the PNLMS and IPNLMS algorithms, using parameter values that lead to the minimal estimation error of each iteration of the adaptation process. Numerical tests performed (considering various scenarios in which the plant impulse response is sparse) show that the proposed methodologies achieve convergence speeds faster than the PNLMS and IPNLMS algorithms, and other algorithms of the PNLMS class, such as the sparseness controlled IPNLMS (SC-IPNLMS) algorithm. / Neste trabalho, novas metodologias para otimizar a escolha dos parâmetros dos algoritmos adaptativos LMS normalizados proporcionais (PNLMS) são propostas. As abordagens propostas usam procedimentos baseados em dois métodos de otimização, a saber, os métodos da razão áurea e da busca tabu. Tais procedimentos são empregados para determinar os parâmetros ótimos em cada iteração do processo de adaptação dos algoritmos PNLMS e PNLMS melhorado (IPNLMS). A função objetivo adotada pelos procedimentos propostos é baseada no erro de estimação a posteriori. O estudo de desempenho realizado para avaliar o impacto dos parâmetros dos algoritmos PNLMS e IPNLMS no comportamento dos mesmos mostram que, com o auxílio de técnicas de otimização para escolher adequadamente tais parâmetros, o desempenho destes algoritmos pode ser melhorado, em termos de velocidade de convergência, para a identificação de plantas com elevado grau de esparsidade. O principal objetivo das metodologias propostas é melhorar a distribuição da energia de ativação entre os coeficientes dos algoritmos PNLMS e IPNLMS, usando valores de parâmetros que levam ao erro de estimação mínimo em cada iteração do processo de adaptação. Testes numéricos realizados (considerando diversos cenários nos quais a resposta impulsiva da planta é esparsa) mostram que as metodologias propostas alcançam velocidades de convergência superiores às dos algoritmos PNLMS e IPNLMS, além de outros algoritmos da classe PNLMS, tais como o algoritmo IPNLMS com controle de esparsidade (SCIPNLMS).