O presente trabalho tem como objetivo fundamental a construção de um sistema de identificação capaz de diagnosticar em tempo real as diferentes configurações de escoamentos bifásicos horizontais. É importante ressaltar que o desenvolvimento deste know-how é capital para a operação eficaz de instalações de manipulação e ou transporte de fluidos multifásicos, e representa, hoje, um dos grandes desafios nas indústrias do petróleo e termonuclear. O princípio de funcionamento do sistema proposto baseia-se nos sinais captados por um sensor de pressão flutuante de resposta rápida, e no seu pósprocessamento com auxílio da transformada de Gabor e de uma rede neural convenientemente treinada. A implementação é tal que a operação de diagnóstico pode ser feita on-line, desde a aquisição dos sinais até o pósprocessamento. Resultados experimentais foram obtidos no circuito experimental do NETeF - Núcleo de Engenharia Térmica e Fluidos da USP - Universidade de São Paulo, para uma secção de testes horizo ntal com 12 m de comprimento e diâmetro interno de 30 mm. Em específico foram ensaiados os seguintes padrões de escoamento ar-água: estratificado liso, ondulado, intermitente, anular e a bolhas. Os resultados mostram que, dependendo dos limites de detecção pré-estabelecidos, todos o principais padrões de escoamento bifásico horizontal são identificados corretamente. / The fundamental objective of this work is the construction of an identification system capable of diagnosing in real time different configurations of horizontal two-phase flow patterns. It is important to emphasize that the development of this know-how is capital to the efficient operation of facilities for manipulation and transportation of multiphase fluids, and represents, today, one of the most important challenges in the oil and thermonuclear industries. The working principle of the proposed system is based on the signals acquired by a rapid response fluctuating pressure sensor, and on its post processing through Gabor Transform and on a conveniently trained artificial neural network. The implementation is accomplished in way that the diagnosis operation is performed on-line, from signal acquisition to post-processing. Experimental results were obtained on the experimental circuit at NETeF - Núcleo de Engenharia Térmica e Fluidos of USP - Universidade de São Paulo, at São Carlos, using a horizontal test section, with 12 m length and 30 mm internal diameter. Experiments were done with the following air-water flow patterns: stratified smooth, wavy, intermittent, annular, and bubbly. Results show that, depending on the preset detection limits, all the main horizontal two phase flow patterns were correctly identified.

This thesis addresses four topics in the area of applied harmonic analysis. First, we show that the affine densities of separable wavelet frames affect the frame properties. In particular, we describe a new relationship between the affine densities, frame bounds and weighted admissibility constants of the mother wavelets of pairs of separable wavelet frames. This result is also extended to wavelet frame sequences. Second, we consider affine pseudodifferential operators, generalizations of pseudodifferential operators that model wideband wireless communication channels. We find two classes of Banach spaces, characterized by wavelet and ridgelet transforms, so that inclusion of the kernel and symbol in appropriate spaces ensures the operator is Schatten p-class. Third, we examine the Schatten class properties of pseudodifferential operators. Using Gabor frame techniques, we show that if the kernel of a pseudodifferential operator lies in a particular mixed modulation space, then the operator is Schatten p-class. This result improves existing theorems and is sharp in the sense that larger mixed modulation spaces yield operators that are not Schatten class. The implications of this result for the Kohn-Nirenberg symbol of a pseudodifferential operator are also described. Lastly, Fourier integral operators are analyzed with Gabor frame techniques. We show that, given a certain smoothness in the phase function of a Fourier integral operator, the inclusion of the symbol in appropriate mixed modulation spaces is sufficient to guarantee that the operator is Schatten p-class.

O presente trabalho tem como objetivo fundamental a construção de um sistema de identificação capaz de diagnosticar em tempo real as diferentes configurações de escoamentos bifásicos horizontais. É importante ressaltar que o desenvolvimento deste know-how é capital para a operação eficaz de instalações de manipulação e ou transporte de fluidos multifásicos, e representa, hoje, um dos grandes desafios nas indústrias do petróleo e termonuclear. O princípio de funcionamento do sistema proposto baseia-se nos sinais captados por um sensor de pressão flutuante de resposta rápida, e no seu pósprocessamento com auxílio da transformada de Gabor e de uma rede neural convenientemente treinada. A implementação é tal que a operação de diagnóstico pode ser feita on-line, desde a aquisição dos sinais até o pósprocessamento. Resultados experimentais foram obtidos no circuito experimental do NETeF - Núcleo de Engenharia Térmica e Fluidos da USP - Universidade de São Paulo, para uma secção de testes horizo ntal com 12 m de comprimento e diâmetro interno de 30 mm. Em específico foram ensaiados os seguintes padrões de escoamento ar-água: estratificado liso, ondulado, intermitente, anular e a bolhas. Os resultados mostram que, dependendo dos limites de detecção pré-estabelecidos, todos o principais padrões de escoamento bifásico horizontal são identificados corretamente. / The fundamental objective of this work is the construction of an identification system capable of diagnosing in real time different configurations of horizontal two-phase flow patterns. It is important to emphasize that the development of this know-how is capital to the efficient operation of facilities for manipulation and transportation of multiphase fluids, and represents, today, one of the most important challenges in the oil and thermonuclear industries. The working principle of the proposed system is based on the signals acquired by a rapid response fluctuating pressure sensor, and on its post processing through Gabor Transform and on a conveniently trained artificial neural network. The implementation is accomplished in way that the diagnosis operation is performed on-line, from signal acquisition to post-processing. Experimental results were obtained on the experimental circuit at NETeF - Núcleo de Engenharia Térmica e Fluidos of USP - Universidade de São Paulo, at São Carlos, using a horizontal test section, with 12 m length and 30 mm internal diameter. Experiments were done with the following air-water flow patterns: stratified smooth, wavy, intermittent, annular, and bubbly. Results show that, depending on the preset detection limits, all the main horizontal two phase flow patterns were correctly identified.

La séparation aveugle de source consiste à estimer les signaux de sources uniquement à partir des mélanges observés. Le problème peut être séparé en deux catégories en fonction du modèle de mélange: mélanges instantanés, où le retard et la réverbération (effet multi-chemin) ne sont pas pris en compte, et des mélanges convolutives qui sont plus généraux mais plus compliqués. De plus, le bruit additif au niveaux des capteurs et le réglage sous-déterminé, où il y a moins de capteurs que les sources, rendent le problème encore plus difficile.Dans cette thèse, tout d'abord, nous avons étudié le lien entre deux méthodes existantes pour les mélanges instantanés: analyse des composants indépendants (ICA) et analyse des composant parcimonieux (SCA). Nous avons ensuite proposé une nouveau formulation qui fonctionne dans les cas déterminés et sous-déterminés, avec et sans bruit. Les évaluations numériques montrent l'avantage des approches proposées.Deuxièmement, la formulation proposés est généralisés pour les mélanges convolutifs avec des signaux de parole. En intégrant un nouveau modèle d'approximation, les algorithmes proposés fonctionnent mieux que les méthodes existantes, en particulier dans des scénarios bruyant et / ou de forte réverbération.Ensuite, on prend en compte la technique de décomposition morphologique et l'utilisation de parcimonie structurée qui conduit à des algorithmes qui peuvent mieux exploiter les structures des signaux audio. De telles approches sont testées pour des mélanges convolutifs sous-déterminés dans un scénario non-aveugle.Enfin, en bénéficiant du modèle NMF (factorisation en matrice non-négative), nous avons combiné l'hypothèse de faible-rang et de parcimonie et proposé de nouvelles approches pour les mélanges convolutifs sous-déterminés. Les expériences illustrent la bonne performance des algorithmes proposés pour les signaux de musique, en particulier dans des scénarios de forte réverbération. / Blind source separation (BSS) consists of estimating the source signals only from the observed mixtures. The problem can be divided into two categories according to the mixing model: instantaneous mixtures, where delay and reverberation (multi-path effect) are not taken into account, and convolutive mixtures which are more general but more complicated. Moreover, the additive noise at the sensor level and the underdetermined setting, where there are fewer sensors than the sources, make the problem even more difficult.In this thesis, we first studied the link between two existing methods for instantaneous mixtures: independent component analysis (ICA) and sparse component analysis (SCA). We then proposed a new formulation that works in both determined and underdetermined cases, with and without noise. Numerical evaluations show the advantage of the proposed approaches.Secondly, the proposed formulation is generalized for convolutive mixtures with speech signals. By integrating a new approximation model, the proposed algorithms work better than existing methods, especially in noisy and/or high reverberation scenarios.Then, we take into account the technique of morphological decomposition and the use of structured sparsity which leads to algorithms that can better exploit the structures of audio signals. Such approaches are tested for underdetermined convolutive mixtures in a non-blind scenario.At last, being benefited from the NMF model, we combined the low-rank and sparsity assumption and proposed new approaches for under-determined convolutive mixtures. The experiments illustrate the good performance of the proposed algorithms for music signals, especially in strong reverberation scenarios.

Visualization, processing and editing of audio, directly on a time-frequency surface, is the scope of this thesis. More precisely the scalogram produced by a Gabor Wavelet transform is used, which is a powerful alternative to traditional techinques where the wave form is the main visual aid and editting is performed by parametric filters. Reconstruction properties, scalogram design and enhancements as well audio manipulation algorithms are investigated for this audio representation.The scalogram is designed to allow a flexible choice of time-frequency ratio, while maintaining high quality reconstruction. For this mean, the Loglet is used, which is observed to be the most suitable filter choice.  Re-assignmentare tested, and a novel weighting function using partial derivatives of phase is proposed.  An audio interpolation procedure is developed and shown to perform well in listening tests.The feasibility to use the transform coefficients directly for various purposes is investigated. It is concluded that Pitch shifts are hard to describe in the framework while noise thresh holding works well. A downsampling scheme is suggested that saves on operations and memory consumption as well as it speeds up real world implementations significantly. Finally, a Scalogram 'compression' procedure is developed, allowing the caching of an approximate scalogram.

Tato práce se zabývá doplňováním chybějících dat do audio signálů a algoritmy řešícími problém založenými na řídké reprezentaci audio signálu. Práce se zaměřuje na některé algoritmy, které řeší doplňování chybějících dat do audio signálů pomocí řídké reprezentace signálů. Součástí práce je také návrh algoritmu, který používá řídkou reprezentaci signálu a také nízkou hodnost signálu ve spektrogramu audio signálu. Dále práce uvádí implementaci tohoto algoritmu v programu Matlab a jeho vyhodnocení.