Les récentes évolutions des technologies de l'information et de la communication font qu'il est aujourd'hui facile de consulter des catalogues de morceaux de musique conséquents. De nouvelles représentations et de nouveaux algorithmes doivent de ce fait être développés afin de disposer d'une vision représentative de ces catalogues et de naviguer avec agilité dans leurs contenus. Ceci nécessite une caractérisation efficace des morceaux de musique par l'intermédiaire de descriptions macroscopiques pertinentes. Dans cette thèse, nous nous focalisons sur l'estimation de la structure des morceaux de musique : il s'agit de produire pour chaque morceau une description de son organisation par une séquence de quelques dizaines de segments structurels, définis par leurs frontières (un instant de début et un instant de fin) et par une étiquette représentant leur contenu sonore.La notion de structure musicale peut correspondre à de multiples acceptions selon les propriétés musicales choisies et l'échelle temporelle considérée. Nous introduisons le concept de structure “sémiotique" qui permet de définir une méthodologie d'annotation couvrant un vaste ensemble de styles musicaux. La détermination des segments structurels est fondée sur l'analyse des similarités entre segments au sein du morceau, sur la cohérence de leur organisation interne (modèle “système-contraste") et sur les relations contextuelles qu'ils entretiennent les uns avec les autres. Un corpus de 383 morceaux a été annoté selon cette méthodologie et mis à disposition de la communauté scientifique.En termes de contributions algorithmiques, cette thèse se concentre en premier lieu sur l'estimation des frontières structurelles, en formulant le processus de segmentation comme l'optimisation d'un coût composé de deux termes~: le premier correspond à la caractérisation des segments structurels par des critères audio et le second reflète la régularité de la structure obtenue en référence à une “pulsation structurelle". Dans le cadre de cette formulation, nous comparons plusieurs contraintes de régularité et nous étudions la combinaison de critères audio par fusion. L'estimation des étiquettes structurelles est pour sa part abordée sous l'angle d'un processus de sélection d'automates à états finis : nous proposons un critère auto-adaptatif de sélection de modèles probabilistes que nous appliquons à une description du contenu tonal. Nous présentons également une méthode d'étiquetage des segments dérivée du modèle système-contraste.Nous évaluons différents systèmes d'estimation automatique de structure musicale basés sur ces approches dans le cadre de campagnes d'évaluation nationales et internationales (Quaero, MIREX), et nous complétons cette étude par quelques éléments de diagnostic additionnels. / Recent progress in information and communication technologies makes it easier to access large collections of digitized music. New representations and algorithms must be developed in order to get a representative overview of these collections, and to browse their content efficiently. It is therefore necessary to characterize music pieces through relevant macroscopic descriptions. In this thesis, we focus on the estimation of the structure of music pieces : the goal is to produce for each piece a description of its organization by means of a sequence of a few dozen structural segments, each of them defined by its boundaries (starting time and ending time) and a label reflecting its audio content.The notion of music structure corresponds to a wide range of meanings depending on the musical properties and the temporal scale under consideration. We introduce an annotation methodology based on the concept of “semiotic structure" which covers a large variety of musical styles. Structural segments are determined through the analysis of their similarities within the music piece, the coherence of their inner organization (“system-contrast" model) and their contextual relationship. A corpus of 383 pieces has been annotated according to this methodology and released to the scientific community.In terms of algorithmic contributions, this thesis concentrates in the first place on the estimation of structural boundaries. We formulate the segmentation process as the optimization of a cost function which is composed of two terms. The first one corresponds to the characterization of structural segments by means of audio criteria. The second one relies on the regularity of the target structure with respect to a “structural pulsation period". In this context, we compare several regularity constraints and study the combination of audio criteria through fusion.Secondly, we consider the estimation of structural labels as a probabilistic finite-state automaton selection process : in this scope, we propose an auto-adaptive criterion for model selection, applied to a description of the tonal content. We also propose a labeling method derived from the system-contrast model.We evaluate several systems for structural segmentation of music based on these approaches in the context of national and international evaluation campaigns (Quaero, MIREX). Additional diagnostic is finally presented to complement this work.

A precisão e acurácia dos métodos mais utilizados atualmente de processamento de dados de espectroscopia por Ressonância Magnética (MRS), baseados na Transformada de Fourier (FT), requerem supressão apropriada (o que está longe de ser trivial) e aquisições longas para a obtenção de alta resolução espectral. Além disso, a FT tem dificuldades quando faltam dados no domínio de tempo, como, por exemplo, pela redução do tempo de aquisição, e consequente número de pontos adquiridos. Isto pode ocorrer, também, por artefatos na aquisição ou, ainda, seja pela exclusão intencional dos primeiros pontos do sinal para a eliminação de ressonâncias largas que estão distorcendo a linha de base no domínio da frequência. Neste estudo, propomos a utilização do Método de Diagonalização na Base de Krylov (KBDM) como uma alternativa a FT para algumas de suas limitações. O método ajusta sinais de experimentos de Free Induction Decay (FID) por uma soma de funções harmônicas complexas, amortecidas exponencialmente, permitindo uma fácil manipulação dos seus parâmetros de caracterização. O KBDM é numericamente mais efetivo para análise de sinais truncados e tem diversos recursos que possibilitam remover picos de forma mais eficiente, como por exemplo, o pico residual da água. Além disso, foi introduzida a possibilidade de quantificação de dados de MRS com o método. Para avaliar a sensibilidade, eficiência e reprodutibilidade do método para quantificar e analisar sinais truncados, foi proposto fazer simulações de espectros clínicos e experimentos em phantoms que representassem o ambiente metabólico do cérebro, para MRS de próton de diferentes níveis de ruídos e para pequenas variações do N-acetil aspartato (NAA). Com estes estudos pôde se comprovar a viabilidade do método para processar dados de MRS e verificar seu potencial na complementação das técnicas atualmente empregadas, especialmente quando uma resolução espectral e temporal maior que o limite imposto pela Relação de Incerteza do formalismo de Fourier é necessária. Além disso, uma desejável facilidade de manipulação de picos específicos (por exemplo, exclusão e quantificação) é proporcionada pelo método. Como perspectivas animadoras deste trabalho esperamos a introdução do KBDM como uma técnica eficiente e coadjuvante ao Imageamento de Ressonância Magnética funcional (fMRI), auxiliando estudos de funções cerebrais, em sequências de MRS para identificar uma rápida variação das linhas associadas as atividades metabólicas dos cérebros. / The precision and accuracy of the most widely used methods to perform Magnetic Resonance Spectroscopy (MRS) data processing based on the Fourier Transform (FT), require appropriate suppression (which is far from trivial) and long acquisitions to obtain high spectral resolution. Furthermore, FT poses difficulty when there are missing data in the time domain. This occurs because of reduction of the acquisition time and consequently also in the number of acquired points, or because of artifacts during acquisition, or even intentional exclusion of the first signal points for the elimination of broad resonances that are producing the distorted baseline in the frequency domain. In this study, we propose the use of the Krylov Basis Diagonalization Method (KBDM) formalism as an alternative to some of FT limitations. The method adjusts signals of Free Induction Decay (FID) experiments with a sum of complex harmonic functions, exponentially damped, allowing easy manipulation of its characterization parameters. The KBDM is numerically more effective for truncated signal analysis and has several features that make it possible to remove peaks more efficiently, such as the residual water peak. Moreover, we introduced the possibility of quantification of MRS data with the described method. To evaluate the sensitivity, efficiency and reproducibility of the method for quantifying and analyzing truncated signals, and through the clinical spectra simulations and experiments in phantoms that would represent the brain metabolic environment, we proposed to perform proton MRS at different noise levels and with small variations of N- acetyl aspartate (NAA) metabolite. These studies allowed to prove the feasibility of the method to process MRS data and verified its potential in complementing techniques currently employed, especially when a greater temporal and spectral resolution is required, more than the limit imposed by the Uncertainty Relation of FT formalism. Furthermore, it is also a desirable effortless tool of handling specific peaks (e.g., exclusion and quantification). Exciting prospects from this work include the introduction of KBDM as an efficient and adjuvant technique to functional Magnetic Resonance Imaging (fMRI), for studying the brain functions, in MRS sequence to identify rapid variation in spectroscopic lines associated to metabolic activities in the brain.

La coexistence de deux formats : cinéma à 24 images/s et vidéo à<br />25 images/s, implique l'accélération ou le ralentissement de la<br />bande-son lors du transfert d'un format vers l'autre. Ceci<br />provoque une modification temporelle du signal sonore, et par<br />conséquent une modification spectrale avec altération du timbre.<br />Les studios de post-production audiovisuelle souhaitent compenser<br />cet effet par l'application d'une transformation sonore adéquate.<br /><br />L'objectif de ce travail est de fournir à l'industrie<br />audiovisuelle un système permettant de pallier la modification de<br />timbre engendrée par le changement de vitesse de lecture. Ce<br />système se compose d'une part d'un algorithme de traitement et<br />d'autre part d'une machine sur lequel il est implanté.<br />L'algorithme est conçu et développé pour répondre aux contraintes<br />liées à la qualité sonore et à la compatibilité multicanal. La<br />machine, baptisée HARMO, est conçue spécifiquement par la société<br />GENESIS sur la base de processeurs de signaux numériques, et doit<br />répondre à la contrainte de temps-réel. Cet aspect "valorisation"<br />conduit à intégrer dans le projet les contraintes de coût et de<br />délai de réalisation.<br /><br />Un état de l'art basé sur une bibliographie quasi-exhaustive<br />aboutit à une classification originale des méthodes de dilatation<br />et de transposition existantes. Ceci nous amène à distinguer et à<br />étudier les méthodes classiques temporelles et fréquentielles, et<br />à introduire les méthodes temps-fréquence. Cette classification<br />est à la base de plusieurs méthodes innovantes :<br /><br />1. deux méthodes temps-fréquence dont l'analyse est adaptée à l'audition,<br /><br />2. deux méthodes couplées qui associent les avantages des méthodes temporelles et fréquentielles,<br /><br />3. une méthode temporelle basée sur une amélioration des méthodes existantes.<br /><br />Les algorithmes sont évalués grâce à une banque de sons-test<br />spécifiquement élaborée pour mettre en évidence les défauts<br />caractéristiques des algorithmes. Notre choix final s'est porté<br />sur l'approche temporelle, que nous optimisons par l'adjonction de<br />critères de segmentation basés sur l'autocorrélation normalisée et<br />la détection de transitoires. Cet algorithme s'intègre dans un<br />logiciel qui a été structuré pour un fonctionnement temps-réel et<br />multicanal sur le système HARMO.

This thesis, within the subfield of computer science known as computer vision, deals with the use of scale-space analysis in early low-level processing of visual information. The main contributions comprise the following five subjects: The formulation of a scale-space theory for discrete signals. Previously, the scale-space concept has been expressed for continuous signals only. We propose that the canonical way to construct a scale-space for discrete signals is by convolution with a kernel called the discrete analogue of the Gaussian kernel, or equivalently by solving a semi-discretized version of the diffusion equation. Both the one-dimensional and two-dimensional cases are covered. An extensive analysis of discrete smoothing kernels is carried out for one-dimensional signals and the discrete scale-space properties of the most common discretizations to the continuous theory are analysed. A representation, called the scale-space primal sketch, which gives a formal description of the hierarchical relations between structures at different levels of scale. It is aimed at making information in the scale-space representation explicit. We give a theory for its construction and an algorithm for computing it. A theory for extracting significant image structures and determining the scales of these structures from this representation in a solely bottom-up data-driven way. Examples demonstrating how such qualitative information extracted from the scale-space primal sketch can be used for guiding and simplifying other early visual processes. Applications are given to edge detection, histogram analysis and classification based on local features. Among other possible applications one can mention perceptual grouping, texture analysis, stereo matching, model matching and motion. A detailed theoretical analysis of the evolution properties of critical points and blobs in scale-space, comprising drift velocity estimates under scale-space smoothing, a classification of the possible types of generic events at bifurcation situations and estimates of how the number of local extrema in a signal can be expected to decrease as function of the scale parameter. For two-dimensional signals the generic bifurcation events are annihilations and creations of extremum-saddle point pairs. Interpreted in terms of blobs, these transitions correspond to annihilations, merges, splits and creations. Experiments on different types of real imagery demonstrate that the proposed theory gives perceptually intuitive results. / <p>QC 20120119</p>

Les émetteurs-récepteurs actuels tendent à devenir multi-standards c’est-àdireque plusieurs standards de communication peuvent cohabiter sur la même puce. Lespuces sont donc amenées à traiter des signaux de formes très différentes, et les composantsanalogiques subissent des contraintes de conception de plus en plus fortes associées au supportdes différentes normes. Les auto-interférences, c’est à dire les interférences généréespar le système lui-même, sont donc de plus en plus présentes, et de plus en plus problématiquesdans les architectures actuelles. Ces travaux s’inscrivent dans le paradigmede la « radio sale » qui consiste à accepter une pollution partielle du signal d’intérêtet à réaliser, par l’intermédiaire d’algorithmes, une atténuation de l’impact de ces pollutionsauto-générées. Dans ce manuscrit, on s’intéresse à différentes auto-interférences(phénomène de "spurs", de "Tx leakage", ...) dont on étudie les modèles numériques etpour lesquelles nous proposons des stratégies de compensation. Les algorithmes proposéssont des algorithmes de traitement du signal adaptatif qui peuvent être vus comme des« algorithmes de soustraction de bruit » basés sur des références plus ou moins précises.Nous dérivons analytiquement les performances transitionnelles et asymptotiques théoriquesdes algorithmes proposés. On se propose également d’ajouter à nos systèmes unesur-couche originale qui permet d’accélérer la convergence, tout en maintenant des performancesasymptotiques prédictibles et paramétrables. Nous validons enfin notre approchesur une puce dédiée aux communications cellulaires ainsi que sur une plateforme de radiologicielle. / Radio frequency transceivers are now massively multi-standards, which meansthat several communication standards can cohabit in the same environment. As a consequence,analog components have to face critical design constraints to match the differentstandards requirements and self-interferences that are directly introduced by the architectureitself are more and more present and detrimental. This work exploits the dirty RFparadigm : we accept the signal to be polluted by self-interferences and we develop digitalsignal processing algorithms to mitigate those aforementioned pollutions and improve signalquality. We study here different self-interferences and propose baseband models anddigital adaptive algorithms for which we derive closed form formulae of both transientand asymptotic performance. We also propose an original adaptive step-size overlay toimprove transient performance of our method. We finally validate our approach on a systemon chip dedicated to cellular communications and on a software defined radio.

Aquesta tesi doctoral versa sobre el processament digital de la veu cantada, més concretament, sobre l'anàlisi, transformació i síntesi d'aquets tipus de veu en el domini espectral, amb especial èmfasi en aquelles tècniques rellevants per al desenvolupament d'aplicacions musicals.La tesi presenta nous procediments i formulacions per a la descripció i transformació d'aquells atributs específicament vocals de la veu cantada. La tesis inclou, entre d'altres, algorismes per l'anàlisi i la generació de desordres vocals como ara rugositat, ronquera, o veu aspirada, detecció i modificació de la freqüència fonamental de la veu, detecció de nasalitat, conversió de veu cantada a melodia, detecció de cops de veu, mutació de veu cantada, i transformació de veu a instrument; exemplificant alguns d'aquests algorismes en aplicacions concretes. / Esta tesis doctoral versa sobre el procesado digital de la voz cantada, más concretamente, sobre el análisis, transformación y síntesis de este tipo de voz basándose e dominio espectral, con especial énfasis en aquellas técnicas relevantes para el desarrollo de aplicaciones musicales.La tesis presenta nuevos procedimientos y formulaciones para la descripción y transformación de aquellos atributos específicamente vocales de la voz cantada. La tesis incluye, entre otros, algoritmos para el análisis y la generación de desórdenes vocales como rugosidad, ronquera, o voz aspirada, detección y modificación de la frecuencia fundamental de la voz, detección de nasalidad, conversión de voz cantada a melodía, detección de los golpes de voz, mutación de voz cantada, y transformación de voz a instrumento; ejemplificando algunos de éstos en aplicaciones concretas. / This dissertation is centered on the digital processing of the singing voice, more concretely on the analysis, transformation and synthesis of this type of voice in the spectral domain, with special emphasis on those techniques relevant for music applications. The thesis presents new formulations and procedures for both describing and transforming those attributes of the singing voice that can be regarded as voice specific. The thesis includes, among others, algorithms for rough and growl analysis and transformation, breathiness estimation and emulation, pitch detection and modification, nasality identification, voice to melody conversion, voice beat onset detection, singing voice morphing, and voice to instrument transformation; being some of them exemplified with concrete applications.

Η υλοποίηση πρακτικών ασύρματων συστημάτων επικοινωνίας δεδομένων στην τεχνολογία UWB παρουσιάζει ιδιαίτερες προκλήσεις, κυρίως λόγω της χαμηλής ισχύος εκπομπής και της πολύ σύντομης διάρκειας των παλμών που χρησιμοποιούνται, οι οποίοι θα πρέπει να στέλνονται με πολύ μεγάλες ταχύτητες για την επίτευξη των επιθυμητών ρυθμών μετάδοσης. Το κανάλι μετάδοσης είναι ιδιαίτερα επιλεκτικό ως προς την συχνότητα και εξαιρετικά πυκνό και πλούσιο σε πολυοδικές συνιστώσες με αρκετά μεγάλες καθυστερήσεις. Αυτές οι συνιστώσες μπορούν να ανιχνευθούν και να συλλεχθούν χρησιμοποιώντας κατάλληλες δομές δεκτών RAKE, οι οποίοι τις συνθέτουν ώστε να μεγιστοποιηθεί η ενέργεια του ωφέλιμου σήματος, αυξάνοντας την απόδοση του συστήματος. Οι δομές αυτές παρουσιάζουν την καλύτερη απόδοση σε τέτοια συστήματα, αλλά έχουν μεγάλη υπολογιστική πολυπλοκότητα, καθώς για την ικανοποιητική απόδοση του συστήματος πρέπει να συνδυάσουν πολλές συνιστώσες, δεδομένης και της χαμηλής ισχύος εκπομπής της τεχνολογίας. Συνεπώς, για την υλοποίηση ενός πρακτικού και αποδοτικού συστήματος, σημαντικό ζήτημα αποτελεί ο τρόπος επιλογής και συνδυασμού των συνιστωσών μέσω ενός αλγορίθμου που θα χρησιμοποιεί τον μικρότερο δυνατό αριθμό δακτύλων. Στόχοι της διατριβής ήταν η μελέτη της τεχνολογίας UWB, η διερεύνηση των παραμέτρων των παλμικών UWB συστημάτων, η μελέτη και εξομοίωση μοντέλων του καναλιού, η κατανόηση των οποίων είναι απαραίτητη για την αποτελεσματική ανίχνευση του σήματος και τον σχεδιασμό των αλγορίθμων ψηφιακής επεξεργασίας του σήματος, η διερεύνηση δεκτών RAKE καθώς και εναλλακτικών δομών, οι εξομοιώσεις πομποδέκτη παλμικού UWB σε επίπεδο συστήματος με έμφαση στον RAKE και τον εκτιμητή καναλιού, η διερεύνηση παραμέτρων και τεχνικών για την υλοποίηση σε υλικό και τέλος η ανάπτυξη, ο σχεδιασμός και υλοποίηση μιας πρακτικής δομής δέκτη με RAKE αποδιαμορφωτή και εκτιμητή καναλιού που συνδυάζει χαμηλή πολυπλοκότητα και ικανοποιητική απόδοση. Παρουσιάζονται και συγκρίνονται τρεις νέες διαφορετικές προσεγγίσεις σχεδίασης, οι οποίες βασίζονται σε προτεινόμενο υβριδικό αλγόριθμο (HPS) για την μείωση της πολυπλοκότητας του RAKE και δίνονται αποτελέσματα που αφορούν στην αξιοποίηση του υλικού και στις επιδόσεις του συστήματος. Tα αποτελέσματα παρουσιάζουν το trade-off ανάμεσα στην συλλογή ενέργειας, την απόδοση του δέκτη και την πολυπλοκότητά του. Η αποτελεσματικότητα των προτεινόμενων αρχιτεκτονικών επαληθεύεται μέσω ειδικής πλατφόρμας αναδιατασσόμενου υλικού στην οποία υλοποιήθηκε η σχεδίαση. / Τhe implementation of practical wireless data communications systems for the UWB technology is very challenging due to the use of low-power ns-duration pulses which have to be sent in a high-frequency in order to achieve the desirable data rates. The UWB channel is highly frequency selective and it is characterized by dense and rich multipath propagation and large multipath delay spreads in some cases. A RAKE receiver can be employed in order to exploit multipath diversity and effectively capture the desired signal energy which is dispersed over the various multipath components, helping to mitigate fading. However, the particular nature of UWB results in very low-energy paths which, in conjunction with high multipath diversity, leads to a RAKE receiver that must exploit a large number of MPCs in order to optimize the received SNR. Thus, for the implementation of a low-complexity system it is important to define a novel method for the selection and combining of MPCs and develop an algorithm that is able to utilize a minimum number of fingers in the RAKE structure. Our work was focused in the study of UWB technology, the investigation of the parameters of IR-UWB systems, the study and understanding of the channel models which is necessary for the design of practical and efficient DSP algorithms, the investigation of RAKE type receivers as well as other alternative structures, the system-level simulations of the IR-UWB transceiver with emphasis given to the algorithms for the RAKE demodulator and channel estimator, the investigation of the parameters and techniques for the implementation of the system in hardware and finally, the development, design, and implementation of a practical receiver structure that includes a RAKE demodulator and a channel estimator and combines low complexity and satisfactory performance. The ultimate goal of this work is the presentation and investigation of the proposed channel estimator and (MRC)-RAKE receiver architecture which is based on a proposed novel hybrid algorithm called HPS. Three different design approaches aiming to a practical system implementation in an FPGA are proposed and compared and system/algorithm performance, hardware utilization results are provided. The obtained results demonstrate the trade-off between energy capture, performance and receiver complexity. The effectiveness of the proposed architectures is verified on a special FPGA platform which was used for the implementation of the receiver structure.

Les émetteurs-récepteurs actuels tendent à devenir multi-standards c’est-àdireque plusieurs standards de communication peuvent cohabiter sur la même puce. Lespuces sont donc amenées à traiter des signaux de formes très différentes, et les composantsanalogiques subissent des contraintes de conception de plus en plus fortes associées au supportdes différentes normes. Les auto-interférences, c’est à dire les interférences généréespar le système lui-même, sont donc de plus en plus présentes, et de plus en plus problématiquesdans les architectures actuelles. Ces travaux s’inscrivent dans le paradigmede la « radio sale » qui consiste à accepter une pollution partielle du signal d’intérêtet à réaliser, par l’intermédiaire d’algorithmes, une atténuation de l’impact de ces pollutionsauto-générées. Dans ce manuscrit, on s’intéresse à différentes auto-interférences(phénomène de "spurs", de "Tx leakage", ...) dont on étudie les modèles numériques etpour lesquelles nous proposons des stratégies de compensation. Les algorithmes proposéssont des algorithmes de traitement du signal adaptatif qui peuvent être vus comme des« algorithmes de soustraction de bruit » basés sur des références plus ou moins précises.Nous dérivons analytiquement les performances transitionnelles et asymptotiques théoriquesdes algorithmes proposés. On se propose également d’ajouter à nos systèmes unesur-couche originale qui permet d’accélérer la convergence, tout en maintenant des performancesasymptotiques prédictibles et paramétrables. Nous validons enfin notre approchesur une puce dédiée aux communications cellulaires ainsi que sur une plateforme de radiologicielle. / Radio frequency transceivers are now massively multi-standards, which meansthat several communication standards can cohabit in the same environment. As a consequence,analog components have to face critical design constraints to match the differentstandards requirements and self-interferences that are directly introduced by the architectureitself are more and more present and detrimental. This work exploits the dirty RFparadigm : we accept the signal to be polluted by self-interferences and we develop digitalsignal processing algorithms to mitigate those aforementioned pollutions and improve signalquality. We study here different self-interferences and propose baseband models anddigital adaptive algorithms for which we derive closed form formulae of both transientand asymptotic performance. We also propose an original adaptive step-size overlay toimprove transient performance of our method. We finally validate our approach on a systemon chip dedicated to cellular communications and on a software defined radio.

This dissertation describes the development, design, implementation and evaluation of a data acquisition system, with the main aim of using it for data collection on a laboratory pilot ball mill. An open-source prototype hardware platform was utilised in the implementation of the data acquisition function, however, with limitations. An analogue signal conditioning card has been successfully developed to interface the analogue signals to the dual domain ADC module. Model-based software development was used to design and develop the algorithms to control the DAS acquisition process, but with limited capabilities. A GUI application has been developed and used for the collection and storage of the raw data on the host system. The DAS prototype was calibrated and collected data successfully through all the channels; however, the input signal bandwidth was limited to 2Hz. / Electrical and Mining Engineering / M. Tech. (Electrical Engineering)

O sinal EEG, cujas informações descrevem o comportamento elétrico do córtex cerebral, apesar de ser bastante utilizado para diagnósticos, principalmente de patologias como epilepsia, no Brasil ainda não é usual o monitoramento contínuo em ambiente de UTI em hospitais públicos. Diante disso, e partindo do pressuposto que estudos sobre o EEG normal, registrado em pessoas sem problemas neurológicos, são escassos, a criação de uma base de registros de EEG normal e análise quantitativa da mesma se faz necessária para que, por meio dos resultados obtidos, padrões normais possam ser estabelecidos e por meio deles a identificação de parâmetros patológicos se torne mais eficaz. Nesse projeto foi elaborada uma base de dados de EEG, com total de 100 registros válidos, advindos de voluntários normais e saudáveis. E a partir desses registros a situação de vigília e olhos fechados foi analisada sob o aspecto de três quantificadores distintos, sendo eles, Porcentagem de Contribuição de Potência (PCP), Frequência Mediana (FM) e Coerência, ambos avaliando o sinal no domínio da frequência. A fim de obter comparações para os resultados obtidos pela análise dos dados do EEG normal, foram utilizados 128 registros de EEG em situação de coma, com diferentes tipos de etiologias e desfechos. Os ritmos que apresentaram maiores distinções entre normal e coma foram Delta e Alfa, principalmente para o quantificador FM. Notou-se que o PCP avaliou características de potência e portanto sintetizou as informações de energia de cada ritmo cerebral tanto em EEG normal quanto em EEG coma. Já FM traz informações de valores de frequências em que há maior concentração de potência, e por fim o quantificador coerência informa o grau de semelhança entre o hemisfério direito e o esquerdo do cérebro. Sendo assim não foi possível afirmar qual dos quantificadores apresentou melhores resultados, visto que cada um trata-se de uma características distintas. / The EEG signal, whose information describes the electrical behavior of the cerebral cortex, although it is widely used for diagnoses, mainly of pathologies such as epilepsy, in Brazil it is still not usual to monitor the ICU environment in public hospitals. Considering this, and assuming that studies on normal EEG, registered in people without neurological problems, are scarce, the creation of a base of normal EEG registers and quantitative analysis of it is necessary so that, through the obtained results, Normal patterns can be established and through them, the identification of pathological parameters becomes more effective. In this project, an EEG database was developed, with 100 valid records from normal and healthy volunteers. In addition, from these records, the waking and closed eyes situation was analyzed under the aspect of three distinct quantifiers, being: Power Contribution Percentage (PCP), Median Frequency (FM) and Coherence, both evaluating the signal in the frequency domain. In order to obtain comparisons for the results obtained by the analysis of the normal EEG data, 128 EEG records were used in coma, with different types of etiologies and outcomes. The rhythms that presented the highest distinctions between normal and coma were Delta and Alpha, mainly for the FM quantifier. It was noted that PCP evaluated power characteristics and therefore synthesized the energy information of each brain rhythm in both normal EEG and EEG coma. Already FM brings information of values of frequencies in which there is greater concentration of power, and finally the quantifier coherence informs the degree of similarity between the right and left hemisphere of the brain. Thus, it was not possible to say which of the quantifiers presented better results, since each one is a distinct characterization. / Dissertação (Mestrado)