No description available.

In this thesis, we consider a sampling design problem for a distributed parameter estimation system. The system contains a number of remotely located local sensors that can preprocess the observed signal and convey the processed data to a data fusion center to make the final estimate. Two issues are considered for this system. One is a sampling scheme design for a parameter estimation problem in a single context. The other is how to assign the appropriate number of sampling points to each of the sensors when a constraint on the total sample size is assumed. Here we propose to design this two issues by maximizing the criterion of Fisher's information or minimizing the Fisher's information loss . A sampling design procedure will be established and some numerical simulations will be also carried out for illustration purpose.

Thèse de doctorat : Electronique, Electrotechnique, Automatique. Traitement du signal : Strasbourg 1 : 2006. / Titre provenant de l'écran-titre. Bibliogr. 6 p.

The body motion of patients, during magnetic resonance (MR) imaging causes significant artifacts in the reconstructed image. Artifacts are manifested as a motion induced blur and ghost repetitions of the moving structures. which obscure vital anatomical and pathological detail. The techniques that have been proposed for suppressing motion artifacts fall into two major categories. Real-time techniques attempt to prevent the motion from corrupting the data by restricting the data acquisition times or motion of the patients, whereas the post-processing techniques use the information embedded in the corrupted data to restore the image. Most methods currently in widespread use belong to the real-time techniques, however with the advent of fast computing platforms and sophisticated signal processing algorithms, the emergence of post-processing techniques is clearly evident. The post-processing techniques usually demand an appropriate model of the motion. The restoration of the image requires that the motion parameters be determined in order to invert the data degradation process. Methods for the correction of translational motion have been studied extensively in the past. The subject of this thesis encompasses the rotational motion model and the effect of rotational motion on the collected MR data in the spatial frequency space (k-space), which is in general, more complicated than the translational model. Rotational motion artifacts are notably prevalent in MR images of head, brain and limbs. Post-processing techniques for the correction of rotational motion artifacts often involve interpolation and re-gridding of the acquired data in the k-space. These methods create significant data overlap and void regions. Therefore, in the past, proposed corrective techniques have been limited to suppression of artifacts caused by small angle rotations. This thesis presents a method of managing overlap regions, using weighted averaging of redundant data, in order to correct for large angle rotations. An iterative estimation technique for filling the data void regions has also been developed by the use of iterated application of projection operators onto constraint sets. These constraint sets are derived from the k-space data generated by the MR imager, and available a priori knowledge. It is shown that the iterative algorithm diverges at times from the required image, due to inconsistency among the constraint sets. It is also shown that this can be overcome by using soft. constraint sets and fuzzy projections. One of the constraints applied in the iterative algorithm is the finite support of the imaged object, marked by the outer boundary of the region of interest (ROI). However, object boundary extraction directly from the motion affected MR image can be difficult, specially if the motion function of the object is unknown. This thesis presents a new ROI extraction scheme based on entropy minimization in the image background. The object rotation function is usually unknown or unable to be measured with sufficient accuracy. The motion estimation algorithm proposed in this thesis is based on maximizing the similarity among the k-space data subjected to angular overlap. This method is different to the typically applied parameter estimation technique based on minimization of pixel energy outside the ROI, and has higher efficiency and ability to estimate rotational motion parameters in the midst of concurrent translational motion. The algorithms for ROI extraction, rotation estimation and data correction have been tested with both phantom images and spin echo MR images producing encouraging results.

Reproduction de : Thèse de doctorat : Acoustique : Le Mans : 2007. / Titre provenant de l'écran-titre. Bibliogr. p.203-209.

The body motion of patients, during magnetic resonance (MR) imaging causes significant artifacts in the reconstructed image. Artifacts are manifested as a motion induced blur and ghost repetitions of the moving structures. which obscure vital anatomical and pathological detail. The techniques that have been proposed for suppressing motion artifacts fall into two major categories. Real-time techniques attempt to prevent the motion from corrupting the data by restricting the data acquisition times or motion of the patients, whereas the post-processing techniques use the information embedded in the corrupted data to restore the image. Most methods currently in widespread use belong to the real-time techniques, however with the advent of fast computing platforms and sophisticated signal processing algorithms, the emergence of post-processing techniques is clearly evident. The post-processing techniques usually demand an appropriate model of the motion. The restoration of the image requires that the motion parameters be determined in order to invert the data degradation process. Methods for the correction of translational motion have been studied extensively in the past. The subject of this thesis encompasses the rotational motion model and the effect of rotational motion on the collected MR data in the spatial frequency space (k-space), which is in general, more complicated than the translational model. Rotational motion artifacts are notably prevalent in MR images of head, brain and limbs. Post-processing techniques for the correction of rotational motion artifacts often involve interpolation and re-gridding of the acquired data in the k-space. These methods create significant data overlap and void regions. Therefore, in the past, proposed corrective techniques have been limited to suppression of artifacts caused by small angle rotations. This thesis presents a method of managing overlap regions, using weighted averaging of redundant data, in order to correct for large angle rotations. An iterative estimation technique for filling the data void regions has also been developed by the use of iterated application of projection operators onto constraint sets. These constraint sets are derived from the k-space data generated by the MR imager, and available a priori knowledge. It is shown that the iterative algorithm diverges at times from the required image, due to inconsistency among the constraint sets. It is also shown that this can be overcome by using soft. constraint sets and fuzzy projections. One of the constraints applied in the iterative algorithm is the finite support of the imaged object, marked by the outer boundary of the region of interest (ROI). However, object boundary extraction directly from the motion affected MR image can be difficult, specially if the motion function of the object is unknown. This thesis presents a new ROI extraction scheme based on entropy minimization in the image background. The object rotation function is usually unknown or unable to be measured with sufficient accuracy. The motion estimation algorithm proposed in this thesis is based on maximizing the similarity among the k-space data subjected to angular overlap. This method is different to the typically applied parameter estimation technique based on minimization of pixel energy outside the ROI, and has higher efficiency and ability to estimate rotational motion parameters in the midst of concurrent translational motion. The algorithms for ROI extraction, rotation estimation and data correction have been tested with both phantom images and spin echo MR images producing encouraging results.

Compressed sensing has been a very active area of research and several elegant algorithms have been developed for the recovery of sparse signals in the past few years. However, most of these algorithms are either computationally expensive or make some assumptions that are not suitable for all real world problems. Recently, focus has shifted to Bayesian-based approaches that are able to perform sparse signal recovery at much lower complexity while invoking constraint and/or a priori information about the data. While Bayesian approaches have their advantages, these methods must have access to a priori statistics. Usually, these statistics are unknown and are often difficult or even impossible to predict. An effective workaround is to assume a distribution which is typically considered to be Gaussian, as it makes many signal processing problems mathematically tractable. Seemingly attractive, this assumption necessitates the estimation of the associated parameters; which could be hard if not impossible. In the thesis, we focus on this aspect of Bayesian recovery and present a framework to address the challenges mentioned above. The proposed framework allows Bayesian recovery of sparse signals but at the same time is agnostic to the distribution of the unknown sparse signal components. The algorithms based on this framework are agnostic to signal statistics and utilize a priori statistics of additive noise and the sparsity rate of the signal, which are shown to be easily estimated from data if not available. In the thesis, we propose several algorithms based on this framework which utilize the structure present in signals for improved recovery. In addition to the algorithm that considers just the sparsity structure of sparse signals, tools that target additional structure of the sparsity recovery problem are proposed. These include several algorithms for a) block-sparse signal estimation, b) joint reconstruction of several common support sparse signals, and c) distributed estimation of sparse signals. Extensive experiments are conducted to demonstrate the power and robustness of our proposed sparse signal estimation algorithms. Specifically, we target the problems of a) channel estimation in massive-MIMO, and b) Narrowband interference mitigation in SC-FDMA. We model these problems as sparse recovery problems and demonstrate how these could be solved naturally using the proposed algorithms.

Muitos métodos de redução de ruído se baseiam na possibilidade de representar o sinal original com um reduzido número de coeficientes de uma transformada, ou melhor, obtém-se um sinal com menos ruído pelo cancelamento dos coeficientes abaixo de um valor adequadamente estabelecido de magnitude. Deve-se supor que a contribuição do ruído se distribua de maneira uniforme por todos os coeficientes. Uma desvantagem destes métodos, quando aplicados a sinais de voz, é a distorção introduzida pela eliminação dos coeficientes de pequena magnitude, juntamente com a presença de sinais espúrios, como o “ruído musical" produzido por coeficientes ruidosos isolados que eventualmente ultrapassam o limiar. Para as transformadas usualmente empregadas, o histograma da distribuição dos coeficientes do sinal de voz possui um grande número de coeficientes próximos à origem. Diante disto, propomos uma nova função de “thresholding" concebida especialmente para redução de ruído em sinais de voz adicionados a AWGN (“Additive, White, and Gaussian Noise"). Esta função, chamada de SoftSoft, depende de dois valores de limiar: um nível inferior, ajustado para reduzir a distorção da voz, e um nível superior, ajustado para eliminar ruído. Os valores ótimos de limiar são calculados para minimizar uma estimativa do erro quadrático médio (MSE): diretamente, supondo conhecido o sinal original; indiretamente, usando uma função de interpolação para o MSE, levando a um método prático. A função SoftSoft alcança um MSE inferior ao que se obtém pelo emprego das conhecidas operações de “Soft" ou “Hard-thresholding", as quais dispõem apenas do limiar superior. Ainda que a melhoria em termos de MSE não seja muito expressiva, a melhoria da qualidade perceptual foi certificada tanto por um ouvinte quanto por uma medida perceptual de distorção (a distância log-espectral). / Many noise-reduction methods are based on the possibility of representing the clean signal as a reduced number of coefficients of a block transform, so that cancelling coefficients below a certain thresholding level will produce an enhanced reconstructed signal. It is necessary to assume that the clean signal has a sparse representation, while the noise energy is spread over all coefficients. The main drawback of those methods is the speech distortion introduced by eliminating small magnitude coefficients, and the presence of artifacts (“musical noise") produced by isolated noisy coefficients randomly crossing the thresholding level. Based on the observation that the speech coefficient histogram has many important coefficients close to origin, we propose a custom thresholding function to perform noise reduction in speech signals corrupted by AWGN. This function, called SoftSoft, has two thresholding levels: a lower level adjusted to reduce speech distortion, and a higher level adjusted to remove noise. The joint optimal values can be determined by minimizing the resulting mean square error (MSE). We also verify that this new thresholding function leads to a lower MSE than the well-known Soft and Hard-thresholding functions, which employ only a higher thresholding level. Although the improvement in terms of MSE is not expressive, a perceptual distortion measure (the log-spectral distance, LSD) is employed to prove the higher performance of the proposed thresholding scheme.

Muitos métodos de redução de ruído se baseiam na possibilidade de representar o sinal original com um reduzido número de coeficientes de uma transformada, ou melhor, obtém-se um sinal com menos ruído pelo cancelamento dos coeficientes abaixo de um valor adequadamente estabelecido de magnitude. Deve-se supor que a contribuição do ruído se distribua de maneira uniforme por todos os coeficientes. Uma desvantagem destes métodos, quando aplicados a sinais de voz, é a distorção introduzida pela eliminação dos coeficientes de pequena magnitude, juntamente com a presença de sinais espúrios, como o “ruído musical” produzido por coeficientes ruidosos isolados que eventualmente ultrapassam o limiar. Para as transformadas usualmente empregadas, o histograma da distribuição dos coeficientes do sinal de voz possui um grande número de coeficientes próximos à origem. Diante disto, propomos uma nova função de “thresholding” concebida especialmente para redução de ruído em sinais de voz adicionados a AWGN (“Additive, White, and Gaussian Noise”). Esta função, chamada de SoftSoft, depende de dois valores de limiar: um nível inferior, ajustado para reduzir a distorção da voz, e um nível superior, ajustado para eliminar ruído. Os valores ótimos de limiar são calculados para minimizar uma estimativa do erro quadrático médio (MSE): diretamente, supondo conhecido o sinal original; indiretamente, usando uma função de interpolação para o MSE, levando a um método prático. A função SoftSoft alcança um MSE inferior ao que se obtém pelo emprego das conhecidas operações de “Soft” ou “Hard-thresholding”, as quais dispõem apenas do limiar superior. Ainda que a melhoria em termos de MSE não seja muito expressiva, a melhoria da qualidade perceptual foi certificada tanto por um ouvinte quanto por uma medida perceptual de distorção (a distância log-espectral). / Many noise-reduction methods are based on the possibility of representing the clean signal as a reduced number of coefficients of a block transform, so that cancelling coefficients below a certain thresholding level will produce an enhanced reconstructed signal. It is necessary to assume that the clean signal has a sparse representation, while the noise energy is spread over all coefficients. The main drawback of those methods is the speech distortion introduced by eliminating small magnitude coefficients, and the presence of artifacts (“musical noise”) produced by isolated noisy coefficients randomly crossing the thresholding level. Based on the observation that the speech coefficient histogram has many important coefficients close to origin, we propose a custom thresholding function to perform noise reduction in speech signals corrupted by AWGN. This function, called SoftSoft, has two thresholding levels: a lower level adjusted to reduce speech distortion, and a higher level adjusted to remove noise. The joint optimal values can be determined by minimizing the resulting mean square error (MSE). We also verify that this new thresholding function leads to a lower MSE than the well-known Soft and Hard-thresholding functions, which employ only a higher thresholding level. Although the improvement in terms of MSE is not expressive, a perceptual distortion measure (the log-spectral distance, LSD) is employed to prove the higher performance of the proposed thresholding scheme.

Darbe atlikta eksponentinių-sinusinių modelių įvertinimo specialiu mažiausių kvadratų metodu analizė. Apžvelgti pagrindiniai signalo parametrai. Aprašyti signalo modeliai bei jų formos. Išnagrinėtas visuminių mažiausių kvadratų metodas bei jam alternatyvūs metodai: kovariacinis, Tufts-Kumaresan ir Pisarenko. Pateikti šių metodų matematiniai aprašymai. Signalų modelių parametrų analizei sukurtos MATLAB programos bei pateikti jų programiniai kodai. Skaitiniais eksperimentais ištirta, kaip kiekvienas iš metodų veikia, esant skirtingam signalo-triukšmo santykiui. Gauti rezultatai iliustruoti grafiškai. Remiantis sumodeliuotais rezultatais, suformuluotos išvados apie nagrinėjamų metodų galimybes. / The aim of this study is to explore exponential-sinusoidal signal model estimation by a special least squares method. The main signal parameters are considered. Signal models and their forms are described. The total least squares method as well as its alternatives – the covariance, Tufts-Kumaresan and Pisarenko methods – are analysed. The mathematical description of these methods is given. MATLAB–based programs to analyse signal model parameters are developed and their codes are presented. We investigated the performance of each of these methods for different signal noise ratio values. The results obtained are illustrated graphically. Conclusions about the method properties are drawn on the basis of simulation experiments.