Thèses sur le sujet « Applications de traitement du signal »
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Vallet, Pascal. « Matrices aléatoires et applications au traitement statistique du signal ». Phd thesis, Université Paris-Est, 2011. http://pastel.archives-ouvertes.fr/pastel-00674747.
Texte intégralHamie, Ali. « Traitement statistique du signal : applications en biologie et économie ». Thesis, Université Grenoble Alpes (ComUE), 2016. http://www.theses.fr/2016GREAS012/document.
Texte intégralIn this thesis, we focus on developing mathematical tools to treat a range of biological and economic signals. First, we propose the Dynalet transform for non-symmetrical biological relaxation signals. This transform is considered as an alternative to the Fourier transform and the wavelet transform. The applicability of the new approximation approach is illustrated on real data. Then, for spectrometric biological signals, we correct the baseline using a penalized expectile regression. Thus, the proposed applications show that our proposed regression is more efficient than the quantile regression. Then to remove random noise, we adapt to spectrometric data a new denoising method that combine wavelets, soft thresholding rule and PLS components. Finally, note that the biological signals may be often regarded as functional data. On one hand, we develop a functional local likelihood aiming to perform a supervised classification of curves. On the other hand, we estimate the regression operator with positive responses, by minimizing the mean squared relative error. Moreover, The asymptotic distributions of our estimator are established and their efficiency is illustrated on a simulation study and on a spectroscopic and economic data set
Nguyen, Quang-Thang. « Contributions to Statistical Signal Processing with Applications in Biomedical Engineering ». Télécom Bretagne, 2012. http://www.telecom-bretagne.eu/publications/publication.php?idpublication=13290.
Texte intégralThis PhD thesis presents some contributions to Statistical Signal Processing with applications in biomedical engineering. The thesis is separated into two parts. In the first part, the detection of protein interface hotspots ¿ the residues that play the most important role in protein interaction ¿ is considered in the Machine Learning framework. The Random Forests is used as the classifier. A new family of protein hotspot descriptors is also introduced. These descriptors are based exclusively on the primary one-dimensional amino acid sequence. No information on the three dimensional structure of the protein or the complex is required. These descriptors, capturing the protein frequency characteristics, make it possible to get an insight into how the protein primary sequence can determine its higher structure and its function. In the second part, the RDT (Random Distortion Testing) robust hypothesis testing is considered. Its application to signal detection is shown to be resilient to model mismatch. We propose an extension of RDT in the sequential decision framework, namely Sequential RDT. Three classical signal deviation/distortion detection problems are reformulated and cast into the RDT framework. Using RDT and Sequential RDT, we investigate the detection of AutoPEEP (auto-Positive End Expiratory Pressure), a common ventilatory abnormality during mechanical ventilation. This is the first work of that kind in the state-of-the-art. Extension to the detection of other types of asynchrony is also studied and discussed. These early detectors of AutoPEEP and asynchrony are key elements of an automatic and continuous patient-ventilator interface monitoring framework
Nyamsi, Lonzenu Madeleine. « Prototypage d'architectures pour des applications de traitement numérique du signal ». Rennes 1, 2005. http://www.theses.fr/2005REN1S120.
Texte intégralNyamsi, Lonzenu Madeleine Quinton Patrice. « Prototypage d'architectures pour des applications de traitement numérique du signal ». [S.l.] : [s.n.], 2005. ftp://ftp.irisa.fr/techreports/theses/2005/nyamsi.pdf.
Texte intégralPapakonstantinou, Konstantinos. « Les applications du traitement du signal statistique à la localisation mobile ». Paris, Télécom ParisTech, 2010. http://www.theses.fr/2010ENST0041.
Texte intégralIn this work we attack the problem of mobile terminal (MT) location estimation in NLoS environments. Traditional localization methods are 2-step processes: In the 1st step a set of location-dependent parameters (LDP) is estimated. In the 2nd step, the MT location is estimated by finding the position that best fits the LDP estimates. For the 1st step we have developed a high-resolution low-complexity LDP estimation algorithm (4D Unitary ESPRIT) for MIMO-OFDM systems, to estimate the angles of arrival (AoA), the angles of departure (AoD), the delays (ToA) and the Doppler shifts (DS) of the multipath components (MPC). As far as the second step of localization is concerned, we developed several hybrid methods applicable to NLoS environments. In the NLoS localization problem, mapping the LDP estimates to the location of the MT is not trivial. To this end, we utilize static and dynamic geometrical channel models (eg. SBM). The 2 great advantages of the SBM-based methods are the identifiability even for cases when LDP estimates are available for only 2 MPC and the remarkable performance for cases when the channel is richer. Due to these great advantages, we consider SBM-based methods to be an appealing solution for the NLoS localization problem. Moreover, we have developed a direct location estimation (DLE) method for MIMO-OFDM systems. In contrast to traditional methods, DLE estimates the MT location directly from the received signal. Its main advantage is the enhanced accuracy at low to medium signal-to-noise ratio (SNR) and/or with small number of data samples, as demonstrated by our results
Daoudi, Khalid. « Généralisations des systèmes de fonctions itérées : applications au traitement du signal ». Paris 9, 1996. https://portail.bu.dauphine.fr/fileviewer/index.php?doc=1996PA090078.
Texte intégralBenjelloun, Touimi Abdellatif. « Traitement du signal audio dans le domaine code : techniques et applications ». Paris, ENST, 2001. http://www.theses.fr/2001ENST0018.
Texte intégralGRIMAL, BRUNO. « Synthese d'architectures autotestables dediees a des applications de traitement du signal ». Rennes 1, 1994. http://www.theses.fr/1994REN10173.
Texte intégralBenjelloun, Touimi Abdellatif. « Traitement du signal audio dans le domaine codé : techniques et applications / ». Paris : École nationale supérieure des télécommunications, 2001. http://catalogue.bnf.fr/ark:/12148/cb388319544.
Texte intégralPham, Gia-Thuy. « Applications des grandes matrices aléatoires aux traitements du signal de grandes dimensions ». Thesis, Paris Est, 2017. http://www.theses.fr/2017PESC1168/document.
Texte intégralLin, Tao. « Traitement du signal tensoriel à base d'ondelettes et applications aux images hyperspectrales ». Ecole centrale de Marseille, 2013. http://www.theses.fr/2013ECDM0002.
Texte intégralVallet, Pascal. « Random matrices and applications to statistical signal processing ». Thesis, Paris Est, 2011. http://www.theses.fr/2011PEST1055/document.
Texte intégralIn this thesis, we consider the problem of source localization in large sensor networks, when the number of antennas of the network and the number of samples of the observed signal are large and of the same order of magnitude. We also consider the case where the source signals are deterministic, and we develop an improved algorithm for source localization, based on the MUSIC method. For this, we fist show new results concerning the position of the eigen values of large information plus noise complex gaussian random matrices
Betser, Michaël. « Modélisation sinusoïdale et applications à l’indexation sonore ». Paris, ENST, 2008. http://pastel.archives-ouvertes.fr/pastel-00004089/fr/.
Texte intégralThe goal of the thesis is the analysis of audio signals using sinusoidal modeling. The first part of the thesis deals with the estimation of the sinusoidal parameters, and in particular with the methods based on the Fourier Transform. The advantages of this family of methods are a low algorithmic complexity and an ease of use. A complete state of the art of these methods is presented. Then, we describe the new estimators which have been developped during the thesis. In particular, we present two original methods allowing to estimate all the parameters of a sinusoid modulated both in amplitude and frequency. Their performances are shown to be better than the only quivalent method in the litterature, namely the quadratically interpolated fast Fourier transform (QIFFT). Audio indexing is a large domain whose purpose is to answer the needs for content access in the audio documents. In the second part of the thesis, we applied the sinusoidal modeling to two audio indexing tasks for which this modeling is particularly appropriate: pitch estimation and sound object detection. The two algorithms developped involve similar ideas: a matching of the sinusoidal peaks estimated in the audio stream with those of the reference sound object, and a likelihood measure of the matching
Woiselle, Arnaud. « Parcimonie et ses applications à des données de flot optique ». Paris 7, 2010. http://www.theses.fr/2010PA077201.
Texte intégralThis manuscript addresses inverse problems using 3D sparsifying transforms in the context of infrared video. We first survey the infrared imaging System and the problems associated such as noise and missing data, and make a short state of the art on industrial imaging processing technics Then we introduce the concept of sparsity and present common algorithms and representations used in this domain. The core of this research is the development of new 3D representations, which are well adapted to representing surfaces or filaments in a 3D volume. Two innovative transforms are proposed, and an existing one ~ the fast curvelet transform — is modified to better suit the applications by reducing its redundancy. These tools are used in many inverse problems such as denoising, inpainting and video deinterlacing, and prove to be very useful and competitive. Finally, we propose a few concrete applications of sparse transforms to infrared video processing, such as small targets detection or motion blur deconvolution
Ghisa, Laura. « Couplage de micro résonateurs. Applications aux fonctions optiques de traitement du signal ». Phd thesis, Université Rennes 1, 2008. http://tel.archives-ouvertes.fr/tel-00383746.
Texte intégralDans une première partie nous présentons un modèle matriciel pour décrire la propagation de la lumière dans des structures à anneaux couplés. L'utilisation de ce modèle se résume pour des structures dont on connaît les coefficients de couplage entre les guides.
Dans le cas de résonateurs constitués d'un milieu qui ne présente pas d'effets non-linéaires nous avons montré que les structures à anneaux couples permettent de réaliser la synthèse de filtres optiques.
Si les résonateurs sont le siège d'une non linéarité dispersive d'ordre 3 nous avons deux cas possibles en fonction du type d'excitation. Hors résonance nous avons montré que nous pouvons obtenir potentiellement des dispositifs /N+1/ stables en utilisant un système formé de /N/ anneaux couplés, ainsi que la tristabilité dans un seul anneau en utilisant un signal de commande. A résonance nous avons montré que l'on peut obtenir la fonction de remise en forme du signal pour des débits de 40 Gb/s.
La deuxième partie porte sur les travaux réalisés sur l'oscillateur optoélectronique. Dans le schéma classique d'un tel oscillateur la fonction de retard optique est obtenue en utilisant une ligne à fibre optique de longueur proportionnelle au retard nécessaire. Nous voulons remplacer cette ligne avec une ligne à retard à base de microrésonateurs. Nous présentons des procédés de fabrication de résonateurs sphériques et diverses méthodes de mesure de leurs facteurs de qualité, proportionnel au retard induit.
Coussy, Philippe. « Synthèse d'architectures pour les applications de traitement du signal et de l'image ». Habilitation à diriger des recherches, Université de Bretagne Sud, 2011. http://tel.archives-ouvertes.fr/tel-00663253.
Texte intégralMba, Mathieu Leonel. « Génération automatique de plate-forme matérielles distribuées pour des applications de traitement du signal ». Electronic Thesis or Diss., Sorbonne université, 2023. http://www.theses.fr/2023SORUS341.
Texte intégralLocal languages or mother tongues of individuals play an essential role in their fulfillment in their various socio-economic activities. African languages and specifically Cameroonian languages are exposed to disappearance in favor of foreign languages adopted as official languages after independence. This is why it is essential to digitalize and integrate them into the majority of dematerialized services for their sustainability. Speech recognition, widely used as a human-machine interface, can be not only a tool for integrating local languages into applications but also a tool for collecting and digitizing corpora. Embedded systems are the preferred environment for deploying applications that use this human-machine interface. This implies that it is necessary to take measures (through the reduction of the reaction time) to satisfy the real-time constraint very often met in this type of application. Two approaches exist for the reduction of the application's response time, namely parallelization and the use of efficient hardware architectures. In this thesis, we exploit a hybrid approach to reduce the response time of an application. We do this by parallelizing this application and implementing it on a reconfigurable architecture. An architecture whose implementation languages are known to be low-level. Moreover, given the multitude of problems posed by the implementation of parallel systems on reconfigurable architecture, there is a problem with design productivity for the engineer. In this thesis, to implement a real-time speech recognition system on an embedded system, we propose an approach for the productive implementation of parallel applications on reconfigurable architecture. Our approach exploits MATIP, a platform-based design tool, as an FPGA Overlay based on high-level synthesis. We exploit this approach to implement a parallel model of a feature extraction algorithm for the recognition of tonal languages (characteristic of the majority of Cameroonian languages). The experimentation of this implementation on isolated words of the Kóló language, in comparison to other implementations (software version and hardware IP), shows that our approach is not only productive in implementation time but also the obtained parallel application is efficient in processing time. This is the reason why we implemented XMATIP an extension of MATIP to make this approach compatible with hardware-software co-design and co-synthesis
Lumeau, Bernard. « Traitement spatial et analyse spectrale applications neurophysiologiques / ». Grenoble 2 : ANRT, 1987. http://catalogue.bnf.fr/ark:/12148/cb37607489r.
Texte intégralRenard, Nadine. « Traitement du signal tensoriel. Application à l'imagerie hyperspectrale ». Aix-Marseille 3, 2008. http://www.theses.fr/2008AIX30062.
Texte intégralThis thesis focus on developing new algebraic methods for hyperspectral applications. The proposed method are original because based on new data representation using third-order tensor. This data representation involves the use of multilinear algebra tools. The proposed methods are referred to as multiway or multimodal methods. TUCKER tensor decompositionbased methods jointly analyze the spatial and spectral modes using an alternating least squares algorithm. This thesis focus on two problematics specific to hyperspectral images. The first one concerns noise reduction. The considered additive noise is due to the acquisition system and degrades the target detection efficiency. A robust to noise detection technique is proposed by incorporating a multimodal Wiener filter. The spatial and spectral n-mode filters are estimated by minimizing the mean squared error between the desired and estimated tensors. The second problematic is the spectral dimension reduction. The curse of the dimensionality degrades the statistical estimation for the classification process. For this issue, the proposed multimodal reduction method reduces the spectral mode by linear transformation jointly to the lower spatial ranks approximation. This method extends the traditional dimension reduction methods. These two multimodal methods are respectively assessed in respect to their impact on detection and classification efficiency. These results highlight the interest of the spatial/spectral analysis in comparison to the traditional spectral analysis only and the hybrid ones which process sequentially the spectral and the spatial mode
Thierry, Eric. « Analyse de forme des signaux monodimensionnels : études théoriques et applications à la structure fine du signal électrique cardiaque ». Nice, 1989. http://www.theses.fr/1989NICE4285.
Texte intégralOlivero, Anaik. « Les multiplicateurs temps-fréquence : Applications à l’analyse et la synthèse de signaux sonores et musicaux ». Thesis, Aix-Marseille, 2012. http://www.theses.fr/2012AIXM4788/document.
Texte intégralAnalysis/Transformation/Synthesis is a generalparadigm in signal processing, that aims at manipulating or generating signalsfor practical applications. This thesis deals with time-frequencyrepresentations obtained with Gabor atoms. In this context, the complexity of a soundtransformation can be modeled by a Gabor multiplier. Gabormultipliers are linear diagonal operators acting on signals, andare characterized by a time-frequency transfer function of complex values, called theGabor mask. Gabor multipliers allows to formalize the conceptof filtering in the time-frequency domain. As they act by multiplying in the time-frequencydomain, they are "a priori'' well adapted to producesound transformations like timbre transformations. In a first part, this work proposes to model theproblem of Gabor mask estimation between two given signals,and provides algorithms to solve it. The Gabor multiplier between two signals is not uniquely defined and the proposed estimationstrategies are able to generate Gabor multipliers that produce signalswith a satisfied sound quality. In a second part, we show that a Gabor maskcontain a relevant information, as it can be viewed asa time-frequency representation of the difference oftimbre between two given sounds. By averaging the energy contained in a Gabor mask, we obtain a measure of this difference that allows to discriminate different musical instrumentsounds. We also propose strategies to automaticallylocalize the time-frequency regions responsible for such a timbre dissimilarity between musicalinstrument classes. Finally, we show that the Gabor multipliers can beused to construct a lot of sounds morphing trajectories,and propose an extension
Burel, Gilles. « Réseaux de neurones en traitement d'images : des modèles théoriques aux applications industrielles ». Brest, 1991. http://www.theses.fr/1991BRES2019.
Texte intégralComte, Jean-Christophe. « Modélisations théorique et électronique de systèmes de réaction-diffusion applications au traitement du signal ». Dijon, 2000. http://www.theses.fr/2000DIJOS022.
Texte intégralShirakawa, Alexandre Augusto. « Etude, synthèse et réalisation de filtres Baw pour applications mobiles ». Bordeaux 1, 2006. http://www.theses.fr/2006BOR13240.
Texte intégralThis manuscript presents the analysis, the synthesis, and the realisation of bulk acoustic wave (BAW) resonator filters. Bulk acoustic wave resonator technology is the only technology that is able to ally excellent filtering performances using a process compatible with silicon integrated circuits. Indeed, this technology enables the conception of very low lossy filters in a wide frequency range (from 500 MHz to 20 GHz) having a power handling capability comparable to ceramic filters. BAW filters present very reduced dimensions and can be fabricated on top of silicon substrates. Firstly, this manuscript presents a detailed study of the bulk acoustic wave technology that shows the multitude of parameters that must be considered on the filter synthesis. Secondly, a synthesis methodology of BAW filters based on a mathematical approach is developed and implemented on a software platform. Finally, in the framework of collaboration between the CEA-Leti and the IXL Microelectronics Laboratory, five bulk acoustic wave resonator filters have been realized. These filters are based on ladder, lattice and ladder-lattice topologies. They addressed the transmission (1. 92 – 1. 98 GHz) and the reception (2. 11 – 2. 17 GHz) of the 3rd generation W CDMA standard. The fabricated filters have confirmed the excellent performances of the BAW-SMR technology concerning these three topologies. The filters show the limitations on selectivity and the insertion losses. The combined ladder-lattice topology presents the best compromise between selectivity and isolation
Andriamisaina, Choukataly Caaliph. « Synthèse d'architecture multi-modes pour les applications du traitement du signal et de l'image ». Lorient, 2008. http://www.theses.fr/2008LORIS127.
Texte intégralEmerging standards lead an increasing demand for high performance, flexibility and low power of embedded systems. These requirements had led to the research on efficient devices to respect these constraints. To have a good trade-off between flexibility and performance, adding flexibility to dedicated architectures and/or improving software performances can be considered. In the literature, the architectures, which result from the addition of flexibility in dedicated architectures, are named multi-modes architectures. The multi-modes or multi-configurations architectures are specifically designed for a set of time-wise mutually exclusive applications (multi-standard applications). They are also used to reduce the reconfiguration time overhead in FPGAs and to implement loop accelerators. We propose a design methodology, based on the High-Level Synthesis (HLS) technique, dedicated to the multi-modes architectures generation. The high-level synthesis is a technique, which considerably increase the productivity by raising the level of abstraction of the input specification and by automating the design. In the context of HLS, we propose to modify the allocation, scheduling and binding steps in order to reduce the area cost of the generated architecture by limiting the performance and power consumption overhead. In order to obtain an area efficient multi-modes architecture, we propose to reduce the area cost of the datapath and the controller. The controller complexity reduction is obtained by increasing the resources similarity between the control steps of the different modes. Whereas, the datapath complexity reduction is obtained by the resources sharing between the modes, and by the consideration of the interconnections cost during this sharing. This approach was integrated in a design environment, dedicated to the multi-modes architectures generation. This environment is based on different tools: GAUT “Multi-modes”, CDFG2UCOM and STARGene “Multi-modes”. It takes as input a functional description specified in C/C++ language in order to obtain automatically a multi-modes architecture at register transfer level described in VHDL. A set of experiment has been done to show the effectiveness of our approach and its associated high-level synthesis tools. From these experiments, we notice that with our approach, we can greatly reduce (up to 60%) the area occupied by the different modes to implement. With this important area reduction, we also obtain a low performance (average of 5%) and power consumption (average of 16%) overhead
Stitou, Youssef. « Modélisation paramètrique 3-D fondée sur la décomposition de Wold : applications au filtrage de blocs sismiques 3-D ». Bordeaux 1, 2006. http://www.theses.fr/2006BOR13296.
Texte intégralLiu, Xuefeng. « Traitement du signal multidimensionnel par décomposition tensorielle et application ». Ecole centrale de Marseille, 2012. http://www.theses.fr/2012ECDM0002.
Texte intégralVincensini, Pascale. « Dispositifs à dispersion contrôlée : application au traitement du signal ». Grenoble INPG, 1988. http://www.theses.fr/1988INPG0013.
Texte intégralVincensini, Pascale. « Dispositifs à dispersion contrôlée application au traitement du signal / ». Grenoble 2 : ANRT, 1988. http://catalogue.bnf.fr/ark:/12148/cb37619138q.
Texte intégralVanloot, Daniel. « Contribution à l'étude et à la réalisation d'un dispositif de traitement du signal optimum pour les systèmes radiométriques : applications au radiomètre à méthode de zéro automatique ». Lille 1, 1987. http://www.theses.fr/1987LIL10053.
Texte intégralDevlaminck, Vincent. « Etude des bruits numériques dans les structures de filtres à coefficients fixes : applications à l'arithmétique classique et à l'arithmétique distribuée ». Lille 1, 1988. http://www.theses.fr/1988LIL10121.
Texte intégralAssous, Saïd. « Analyse temps-fréquence par la transformée en S et interprétation des signaux de fluximétrie laser Doppler : applications au diagnostic clinique ». Paris, ENSAM, 2005. http://www.theses.fr/2005ENAM0026.
Texte intégralThe laser Doppler flowmetry (LDF) technique is based on the spectral broadening of monochromatic light, that interacts with moving red blood cells in tissue. The power spectral density of the backscattered light can be processed to yield an estimate of microvascular tissue perfusion in the form of a signal. The primary objective of this work is to explore the information contained in this signal in order to understand the physiological phenomena which appear in the microcirculation and to present solutions of assistance in the diagnosis of corresponding pathologies. The decision problem may be divided into the stages of feature extraction, dimensionality reduction and pattern recognition. This work shows that decision performance depends largely upon the signal representation. The study relies on the S transform, a linear time-frequency representation which overcomes the short time Fourier transform fixed length window limitation and the phase notion in the wavelet transform for non stationary signal analysis. This transform provides a powerful framework for feature extraction, localizing the discriminant information in the laser Doppler flowmetry signal in time and in frequency. This new approach makes possible the analysis of the five characteristic frequencies contained in laser Doppler flowmetry signals and contributes to the apprehension of the signals of reactive hyperaemia. In addition, due to the high dimension of time-frequency representations, its success relies upon an appropriate form of dimensionality reduction. It is shown that the singular values decomposition provides an effective means of concentrating that information which is important, and discarding that which is irrelevant. Its singular vectors reflect the behavior spectral in the time of the various physiological activities present in the laser Doppler flowmetry signal
Le, Pennec Erwan. « Some (statistical) applications of Ockham's principle ». Habilitation à diriger des recherches, Université Paris Sud - Paris XI, 2013. http://tel.archives-ouvertes.fr/tel-00802653.
Texte intégralRaspaud, Martin. « Modèles spectraux hiérarchiques pour les sons et applications ». Bordeaux 1, 2007. https://zenodo.org/records/10477958.
Texte intégralDessein, Arnaud. « Méthodes Computationnelles en Géométrie de l'Information et Applications Temps Réel au Traitement du Signal Audio ». Phd thesis, Université Pierre et Marie Curie - Paris VI, 2012. http://tel.archives-ouvertes.fr/tel-00768524.
Texte intégralFail, Abderrahim. « Etude de capteurs-émetteurs et des procédés de télétransmission associés : applications à la rééducation des membres inférieurs ». Vandoeuvre-les-Nancy, INPL, 1990. http://www.theses.fr/1990INPL055N.
Texte intégralWang, Hongzhi. « Architectures reconfigurables à base d'opérateur CORDIC pour le traitement du signal : Applications aux récepteurs MIMO ». Phd thesis, Université Rennes 1, 2009. http://tel.archives-ouvertes.fr/tel-00446156.
Texte intégralDECHELLE, FRANCOIS. « Definition d'une station de developpement pour des applications de traitement du signal sur multi-processeurs ». Paris 6, 1989. http://www.theses.fr/1989PA066585.
Texte intégralSymul, Thomas. « Amplification paramétrique non linéaire : applications au traitement tout optique du signal et à l'information quantique ». Paris 6, 2001. http://www.theses.fr/2001PA066480.
Texte intégralWang, Hongzhi. « Architectures reconfigurables à base d’opérateur CORDIC pour le traitement du signal : applications aux récepteurs MIMO ». Rennes 1, 2009. http://www.theses.fr/2009REN1S028.
Texte intégralThis thesis addresses the realization of reconfigurable architectures to answer the communication system requirement like high performance on data rate and flexibility supporting the evolution of future systems. We are particularly interested in MIMO applications. The proposed architectures use CORDIC (COordinate Rotation DIgital Computing) operator as common operator. Also the static and partial reconfiguration of FPGA is performed in view of an optimization and dynamic adaptation of hardware resources according to communication system specifications. The number of CORDIC operators can be changed by static reconfiguration to adapt to different MIMO receivers and different required data rate. The interconnections between CORDIC operators can be changed by partial and dynamic FPGA reconfiguration to optimize the hardware resources. The impact of step number of CORDIC operator on BER and MSE is also studied. Three MIMO receivers are realized in this architecture: V-BLAST Square Root, MMSE and CMA. The use of these architectures allows reducing the hardware resources without BER and MSE degradation
Thioune, Abdoulaye. « Décomposition modale empirique et décomposition spectrale intrinsèque : applications en traitement du signal et de l’image ». Thesis, Paris Est, 2015. http://www.theses.fr/2015PESC1127/document.
Texte intégralIn this thesis, it is about a study on the time-frequency, time-scale analysis methods and more particularly on Empirical Mode Decomposition (EMD), by first a course on traditional methods from Fourier analysis to wavelets, including the multiresolution representation. The need for precision measurements both in time space and in frequency space has always been a major preoccupation. In fact, the Fourier transformation does not reconcile the frequency description and location in time. The Short-Term Fourier Transform (STFT) and its derivatives - including the spectrogram - have long been the most used in practical applications. It must be recognized that despite its many attractive aspects, these technics are naturally limited by the fact that they were ineffective for non-stationary signals analysis. The wavelet transform has been very successful in recent decades with the large number of its applications in signal and image processing. Despite its effectiveness in the representation and manipulation of signals, even non-stationary, a priori knowledge about the signal to be decomposed is necessary for an appropriate wavelet choice for each type of signal. The empirical mode decomposition (EMD) is a decomposition method of non-stationary or from non-linear systems signals, in an amount of modes, each mode being localized in frequency. This decomposition is associated with a Hilbert-Huang transformation (HHT) to locally extract instantaneous amplitude and instantaneous frequency. It is similar to the wavelet decomposition with the added benefit that constitutes its auto-adaptability. In the remainder of this work, we introduced a new decomposition method based on a spectral decomposition of an interpolation intrinsic operator. The new method called Spectral Decomposition Intrinsic (SID) is auto-adaptive and is more general than the basic principle of Empirical Mode Decomposition. The SID method can produce a dictionary of Spectral Proper Mode Functions (SPMF) that are similar to atoms in sparse representations
Doghmane, Noureddine. « Application des techniques de traitement du signal en spectroscopie Auger ». Grenoble 2 : ANRT, 1988. http://catalogue.bnf.fr/ark:/12148/cb37613301g.
Texte intégralWu, Jiasong. « Algorithmes rapides pour transformées orthogonales discrètes et leurs applications en traitement du signal et de l'image ». Rennes 1, 2011. http://www.theses.fr/2011REN1S153.
Texte intégralCette thèse se focalise principalement sur les algorithmes rapides dédiés à certaines transformations trigonométriques (MDCT, MCLT, DHT) et à des transformations de Hadamard (SCHT et CS-SCHT). Nous proposons des approches algorithmiques qui permettent de réduire leur complexité numérique et nous comparons leur performance à d’autres algorithmes de la littérature. Nous explorons aussi leurs applications potentielles en traitement du signal et de l’image. Les principaux résultats et contributions de cette thèse peuvent être résumés ainsi : 1. Algorithmes rapides pour les transformations orthogonales à recouvrement (1) Pour la MDCT monodimensionnelle, nous avons proposé une approche unifiée pour des algorithmes de MDCT combinant plusieurs subdivisions du nombre de points du signal à transformer. Il s’agit de méthodes agrégeant des algorithmes de décimation en fréquence et ou en temps pour un nombre de points composite en puissance de 2 et en puissance d’un nombre impair q (radix-2 decimation-in-frequency (DIF), radix-3 DIF algorithm, radix-3 decimation-in-time (DIT) algorithm et radix-q DIT algorithm). (2) Pour la MDCT en 2D, nous avons proposé un algorithme rapide basé sur la DCT de type II. (3) En ce qui concerne la MCLT monodimensionnelle, nous avons proposé une approche de mise en œuvre à partir de la transformée de Hartley généralisée de type II (GDHT-II). 2. Algorithmes rapides pour la transformée de Hartley discrète (GHT) (1) Pour la DHT monodimensionnelle, en considérant une séquence de longueur N subdivisée en trois sous séquences adjacentes de longueur N/3 dont on connaît les DHT, nous avons proposé un algorithme rapide et efficace de calcul de la DHT de la séquence. (2) Nous avons proposé deux algorithmes rapides pour la DHT en 2D. Le premier est un algorithme de type DIF où le nombre de points est composite et multiple impair d’une puissance de 2. Nous avons mis en œuvre cet algorithme pour la compression d’image notamment médicale. Le second, se fond sur une approche DIF pour un nombre de points qui est une puissance de trois. (3) Une extension en 3D a aussi été entreprise pour la GDHT, nous avons proposé un algorithme pour cette transformation en prenant comme point de départ l’algorithme DIF et en considérant un nombre de points en puissance de deux. Cette approche a été mise en œuvre pour le calcul de convolutions circulaires en 3D. 3. Algorithmes rapides pour la transformée complexe de Hadamard sur fenêtre glissante (SCHT). (1) Dans le cas monodimensionnel, nous avons proposé deux algorithmes, l’un du type « radix-2 DIS » et l’autre du type « radix-4 DIS » pour le calcul de la SCHT. (2) Pour le calcul de la transformation CS-SCHT, nous avons proposé trois approches rapides, incluant les méthodes « radix-2 DIS » et « radix-4 DIS » et le codage de gray (GCK). (3) Nous avons mis en œuvre les transformations SCHT et CS-SCHT dans le contexte de l’estimation spectrale et du filtrage adaptatif
Benazouz, Mohamed. « Dimensionnement des mémoires pour les applications de traitement de flux de données ». Paris 6, 2012. http://www.theses.fr/2012PA066067.
Texte intégralFlamary, Rémi. « Apprentissage statistique pour le signal : applications aux interfaces cerveau-machine ». Phd thesis, Université de Rouen, 2011. http://tel.archives-ouvertes.fr/tel-00687501.
Texte intégralMorfu, Savério. « Etude des défauts et perturbations dans les réseaux électroniques dissipatifs non linéaires : applications à la transmission et au traitement du signal ». Dijon, 2002. http://www.theses.fr/2002DIJOS015.
Texte intégralLatiri, Mohamed Anis. « Architecture et conception de récepteur reconfigurable à échantillonnage RF pour les applications mutistandard ». Paris, ENST, 2008. http://pastel.archives-ouvertes.fr/pastel-00004551/fr/.
Texte intégralThe fast development of wireless communication systems requires more flexible and cost effective radio architectures. The concept of "Software Defined Radio" where communication standards are chosen by reconfiguration of hardware is attractive but still unrealistic at present time, due to the high requirements imposed on the analog to digital converter. Different techniques for shifting the RF and analog circuit design complexity to digitally intensive domain were developed recently. These techniques are based on direct RF sampling and discrete-time analog signal processing and allow for a great flexibility and ease of reconfigurability, as well as a high level of hardware integration and a reduction of cost and power consumption. These concepts have been used in this thesis to develop a reconfigurable discrete-time radio receiver front-end. The circuit, mainly structured around two discrete-time analog signal processing stages, performs RF sampling, anti-alias filtering, frequency downconversion, decimation and lowpass filtering. The GSM900 and 802. 11g communication standards have been targeted in order to validate the flexibility and reconfigurability of the receiver. The circuit has been designed in 90nm CMOS technology and first measurement results demonstrated the functionality of the receiver. In addition, a fully passive second order discrete-time anti-alias filter has been described and included in the proposed receiver. This filter is intended to considerably improve the alias filter rejection and can be tuned to different RF frequency bands thanks to its reconfigurable structure and is hence suitable for true multi-standard operations
Flamary, Rémi. « Apprentissage statistique pour le signal : applications aux interfaces cerveau-machine ». Phd thesis, Rouen, 2011. http://www.theses.fr/2011ROUES044.
Texte intégralBrain Computer Interfaces (BCI) require the use of statistical learning methods for signal recognition. In this thesis we propose a general approach using prior knowledge on the problem at hand through regularization. To this end, we learn jointly the classifier and the feature extraction step in a unique optimization problem. We focus on the problem of sensor selection, and propose several regularization terms adapted to the problem. Our first contribution is a filter learning method called large margin filtering. It consists in learning a filtering maximizing the margin between samples of each classe so as to adapt to the properties of the features. In addition, this approach is easy to interpret and can lead to the selection of the most relevant sensors. Numerical experiments on a real life BCI problem and a 2D image classification show the good behaviour of our method both in terms of performance and interpretability. The second contribution is a general sparse multitask learning approach. Several classifiers are learned jointly and discriminant kernels for all the tasks are automatically selected. We propose some efficient algorithms and numerical experiments have shown the interest of our approach. Finally, the third contribution is a direct application of the sparse multitask learning to a BCI event-related potential classification problem. We propose an adapted regularization term that promotes both sensor selection and similarity between the classifiers. Numerical experiments show that the calibration time of a BCI can be drastically reduced thanks to the proposed multitask approach
Rosenberg, Nicole. « La suppression du fond en spectrométrie Auger : étude générale et applications ». Lyon 1, 1990. http://www.theses.fr/1990LYO10004.
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