Dissertationen zum Thema „Statistical and digital signal processing“
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Wu, Tsan-Ming. „Statistical impulse reponse modeling and dereverberation for room acoustics“. Diss., Georgia Institute of Technology, 2000. http://hdl.handle.net/1853/14932.
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黃俊賢 und Chun-yin Vong. „Performance study of uniform sampling digital phase-locked loopsfor [Pi]/4-differentially encoded quaternary phase-shift keying“. Thesis, The University of Hong Kong (Pokfulam, Hong Kong), 1998. http://hub.hku.hk/bib/B31221816.
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Chan, Francis Chun Ngai Electrical Engineering & Telecommunications Faculty of Engineering UNSW. „Statistical methods on detecting superpositional signals in a wireless channel“. Awarded by:University of New South Wales. School of Electrical Engineering and Telecommunications, 2006. http://handle.unsw.edu.au/1959.4/30596.
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The objective of the thesis is concerned on the problem of detecting superpositional signals in a wireless channel. In many wireless systems, an observed signal is commonly represented as a linear combination of the transmitted signal with the interfering signals dispersed in space and time. These systems are generally known as the interference-limited systems. The mathematical model of these systems is generally referred as a superpositional model. A distinguished characteristic of signal transmission in a time-varying wireless channel is that the channel process is not known a priori. Reliable signal reception inherently requires exploiting the structure of the interfering signals under channel uncertainty. Our goal is to design computational efficient receivers for various interference-limited systems by using advanced statistical signal processing techniques. The thesis consists of four main parts. Firstly, we have proposed a novel Multi-Input Multi-Output (MIMO) signal detector, known as the neighbourhood exploring detector (NED). According to the maximum likelihood principle, the space time MIMO detection problem is equivalent to a NP-hard combinatorial optimization problem. The proposed detector is a sub-optimal maximum likelihood detector which eliminates exhaustive multidimensional searches. Secondly, we address the problem of signal synchronization for Global Positioning System (GPS) in a multipath environment. The problem of multipath mitigation constitutes a joint estimation of the unknown amplitudes, phases and time delays of the linearly combined signals. The complexity of the nonlinear joint estimation problem increases exponentially with the number of signals. We have proposed two robust GPS code acquisition systems with low complexities. Thirdly, we deal with the problem of multipath mitigation in the spatial domain. A GPS receiver integrated with the Inertial Navigation System (INS) and a multiple antenna array is considered. We have designed a software based GPS receiver which effectively estimates the directions of arrival and the time of arrival of the linearly combined signals. Finally, the problem of communications with unknown channel state information is investigated. Conventionally, the information theoretical communication problem and the channel estimation problem are decoupled. However the training sequence, which facilitates the estimation of the channel, reduces the throughput of the channel. We have analytically derived the optimal length of the training sequence which maximizes the mutual information in a block fading channel.
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Le, Faucheur Xavier Jean Maurice. „Statistical methods for feature extraction in shape analysis and bioinformatics“. Diss., Georgia Institute of Technology, 2010. http://hdl.handle.net/1853/33911.
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The presented research explores two different problems of statistical data analysis.
In the first part of this thesis, a method for 3D shape representation, compression and smoothing is presented. First, a technique for encoding non-spherical surfaces using second generation wavelet decomposition is described. Second, a novel model is proposed for wavelet-based surface enhancement. This part of the work aims to develop an efficient algorithm for removing irrelevant and noise-like variations from 3D shapes. Surfaces are encoded using second generation wavelets, and the proposed methodology consists of separating noise-like wavelet coefficients from those contributing to the relevant part of the signal. The empirical-based Bayesian models developed in this thesis threshold wavelet coefficients in an adaptive and robust manner. Once thresholding is performed, irrelevant coefficients are removed and the inverse wavelet transform is applied to the clean set of wavelet coefficients. Experimental results show the efficiency of the proposed technique for surface smoothing and compression.
The second part of this thesis proposes using a non-parametric clustering method for studying RNA (RiboNucleic Acid) conformations. The local conformation of RNA molecules is an important factor in determining their catalytic and binding properties. RNA conformations can be characterized by a finite set of parameters that define the local arrangement of the molecule in space. Their analysis is particularly difficult due to the large number of degrees of freedom, such as torsion angles and inter-atomic distances among interacting residues. In order to understand and analyze the structural variability of RNA molecules, this work proposes a methodology for detecting repetitive conformational sub-structures along RNA strands. Clusters of similar structures in the conformational space are obtained using a nearest-neighbor search method based on the statistical mechanical Potts model. The proposed technique is a mostly automatic clustering algorithm and may be applied to problems where there is no prior knowledge on the structure of the data space, in contrast to many other clustering techniques. First, results are reported for both single residue conformations- where the parameter set of the data space includes four to seven torsional angles-, and base pair geometries. For both types of data sets, a very good match is observed between the results of the proposed clustering method and other known classifications, with only few exceptions. Second, new results are reported for base stacking geometries. In this case, the proposed classification is validated with respect to specific geometrical constraints, while the content and geometry of the new clusters are fully analyzed.
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Kjellson, Angelica. „Sound Source Localization and Beamforming for Teleconferencing Solutions“. Thesis, Umeå universitet, Institutionen för matematik och matematisk statistik, 2014. http://urn.kb.se/resolve?urn=urn:nbn:se:umu:diva-89707.
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In teleconferencing the audio quality is key to conducting successful meetings. The conference room setting imposes various challenges on the speech signal processing, such as noise and interfering signals, reverberation, or participants positioned far from the telephone unit. This work aims at improving the received speech signal of a conference telephone by implementing sound source localization and beamforming. The implemented microphone array signal processing techniques are compared to the performance of an existing multi-microphone solution and evaluated under various conditions using a planar uniform circular array. Recordings of test-sequences for the evaluation were performed using a custom-built array mockup. The implemented algorithms did not show good enough performance to motivate the increased computational complexity compared to the existing solution. Moreover, an increase in number of microphones used was concluded to have little or no effect on the performance of the methods. The type of microphone used was, however, concluded to have impact on the performance and a subjective listening evaluation indicated a preference for omnidirectional microphones which is recommended to investigate further.God ljudkvalitet är en grundsten för lyckade telefonmöten. Miljön i ett konferens-rum medför ett flertal olika utmaningar för behandlingen av mikrofonsignalerna: det kan t.ex. vara brus och störningar, eller att den som talar befinner sig långt från telefonen. Målet med detta arbete är att förbättra den talsignal som tas upp av en konferenstelefon genom att implementera lösningar för lokalisering av talaren och riktad ljudupptagning med hjälp av ett flertal mikrofoner. De implementerade metoderna jämförs med en befintlig lösning och utvärderas under olika brusscenarion för en likformig cirkulär mikrofonkonstellation. För utvärderingen användes testsignaler som spelades in med en specialbyggd enhet. De implementerade algoritmerna kunde inte uppvisa en tillräcklig förbättring i jämförelse med den befintliga lösningen för att motivera den ökade beräkningskomplexitet de skulle medföra. Dessutom konstaterades att en fördubbling av antalet mikrofoner gav liten eller ingen förbättring på metoderna. Vilken typ av mikrofon som användes konstaterades däremot påverka resultatet och en subjektiv utvärdering indikerade en preferens för de rundupptagande mikrofonerna, en skillnad som föreslås undersökas vidare.
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Frankford, Mark Thomas. „EXPLORATION OF MIMO RADAR TECHNIQUES WITH A SOFTWARE-DEFINED RADAR“. The Ohio State University, 2011. http://rave.ohiolink.edu/etdc/view?acc_num=osu1306526246.
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Luo, Chenchi. „Non-uniform sampling: algorithms and architectures“. Diss., Georgia Institute of Technology, 2012. http://hdl.handle.net/1853/45873.
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Modern signal processing applications emerging in telecommunication and instrumentation industries have placed an increasing demand for ADCs with higher speed and resolution. The most fundamental challenge in such a progress lies at the heart of the classic signal processing: the Shannon-Nyquist sampling theorem which stated that when sampled uniformly, there is no way to increase the upper frequency in the signal spectrum and still unambiguously represent the signal except by raising the sampling rate. This thesis is dedicated to the exploration of the ways to break through the Shannon-Nyquist sampling rate by applying non-uniform sampling techniques.
Time interleaving is probably the most intuitive way to parallel the uniform sampling process in order to achieve a higher sampling rate. Unfortunately, the channel mismatches in the TIADC system make the system an instance of a recurrent non-uniform sampling system whose non-uniformities are detrimental to the performance of the system and need to be calibrated. Accordingly, this thesis proposed a flexible and efficient architecture to compensate for the channel mismatches in the TIADC system. As a key building block in the calibration architecture, the design of the Farrow structured adjustable fractional delay filter has been investigated in detail. A new modified Farrow structure is proposed to design the adjustable FD filters that are optimized for a given range of bandwidth and fractional delays. The application of the Farrow structure is not limited to the design of adjustable fractional delay filters. It can also be used to implement adjustable lowpass, highpass and bandpass filters as well as adjustable multirate filters. This thesis further extends the Farrow structure to the design of filters with adjustable polynomial phase responses.
Inspired by the theory of compressive sensing, another contribution of this thesis is to use randomization as a means to overcome the limit of the Nyquist rate. This thesis investigates the impact of random sampling intervals or jitters on the power spectrum of the sampled signal. It shows that the aliases of the original signal can be well shaped by choosing an appropriate probability distribution of the sampling intervals or jitters such that aliases can be viewed as a source of noise in the signal power spectrum. A new theoretical framework has been established to associate the probability mass function of the random sampling intervals or jitters with the aliasing shaping effect. Based on the theoretical framework, this thesis proposes three random sampling architectures, i.e., SAR ADC, ramp ADC and level crossing ADC, that can be easily implemented based on the corresponding standard ADC architectures. Detailed models and simulations are established to verify the effectiveness of the proposed architectures. A new reconstruction algorithm called the successive sine matching pursuit has also been proposed to recover a class of spectrally sparse signals from a sparse set of non-uniform samples onto a denser uniform time grid so that classic signal processing techniques can be applied afterwards.
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Vargas, Paredero David Eduardo. „Transmit and Receive Signal Processing for MIMO Terrestrial Broadcast Systems“. Doctoral thesis, Universitat Politècnica de València, 2016. http://hdl.handle.net/10251/66081.
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[EN] Multiple-Input Multiple-Output (MIMO) technology in Digital Terrestrial Television (DTT) networks has the potential to increase the spectral efficiency and improve network coverage to cope with the competition of limited spectrum use (e.g., assignment of digital dividend and spectrum demands of mobile broadband), the appearance of new high data rate services (e.g., ultra-high definition TV - UHDTV), and the ubiquity of the content (e.g., fixed, portable, and mobile). It is widely recognised that MIMO can provide multiple benefits such as additional receive power due to array gain, higher resilience against signal outages due to spatial diversity, and higher data rates due to the spatial multiplexing gain of the MIMO channel. These benefits can be achieved without additional transmit power nor additional bandwidth, but normally come at the expense of a higher system complexity at the transmitter and receiver ends. The final system performance gains due to the use of MIMO directly depend on physical characteristics of the propagation environment such as spatial correlation, antenna orientation, and/or power imbalances experienced at the transmit aerials. Additionally, due to complexity constraints and finite-precision arithmetic at the receivers, it is crucial for the overall system performance to carefully design specific signal processing algorithms.
This dissertation focuses on transmit and received signal processing for DTT systems using MIMO-BICM (Bit-Interleaved Coded Modulation) without feedback channel to the transmitter from the receiver terminals. At the transmitter side, this thesis presents investigations on MIMO precoding in DTT systems to overcome system degradations due to different channel conditions. At the receiver side, the focus is given on design and evaluation of practical MIMO-BICM receivers based on quantized information and its impact in both the in-chip memory size and system performance. These investigations are carried within the standardization process of DVB-NGH (Digital Video Broadcasting - Next Generation Handheld) the handheld evolution of DVB-T2 (Terrestrial - Second Generation), and ATSC 3.0 (Advanced Television Systems Committee - Third Generation), which incorporate MIMO-BICM as key technology to overcome the Shannon limit of single antenna communications. Nonetheless, this dissertation employs a generic approach in the design, analysis and evaluations, hence, the results and ideas can be applied to other wireless broadcast communication systems using MIMO-BICM.[ES] La tecnología de múltiples entradas y múltiples salidas (MIMO) en redes de Televisión Digital Terrestre (TDT) tiene el potencial de incrementar la eficiencia espectral y mejorar la cobertura de red para afrontar las demandas de uso del escaso espectro electromagnético (e.g., designación del dividendo digital y la demanda de espectro por parte de las redes de comunicaciones móviles), la aparición de nuevos contenidos de alta tasa de datos (e.g., ultra-high definition TV - UHDTV) y la ubicuidad del contenido (e.g., fijo, portable y móvil). Es ampliamente reconocido que MIMO puede proporcionar múltiples beneficios como: potencia recibida adicional gracias a las ganancias de array, mayor robustez contra desvanecimientos de la señal gracias a la diversidad espacial y mayores tasas de transmisión gracias a la ganancia por multiplexado del canal MIMO. Estos beneficios se pueden conseguir sin incrementar la potencia transmitida ni el ancho de banda, pero normalmente se obtienen a expensas de una mayor complejidad del sistema tanto en el transmisor como en el receptor. Las ganancias de rendimiento finales debido al uso de MIMO dependen directamente de las características físicas del entorno de propagación como: la correlación entre los canales espaciales, la orientación de las antenas y/o los desbalances de potencia sufridos en las antenas transmisoras. Adicionalmente, debido a restricciones en la complejidad y aritmética de precisión finita en los receptores, es fundamental para el rendimiento global del sistema un diseño cuidadoso de algoritmos específicos de procesado de señal. Esta tesis doctoral se centra en el procesado de señal, tanto en el transmisor como en el receptor, para sistemas TDT que implementan MIMO-BICM (Bit-Interleaved Coded Modulation) sin canal de retorno hacia el transmisor desde los receptores. En el transmisor esta tesis presenta investigaciones en precoding MIMO en sistemas TDT para superar las degradaciones del sistema debidas a diferentes condiciones del canal. En el receptor se presta especial atención al diseño y evaluación de receptores prácticos MIMO-BICM basados en información cuantificada y a su impacto tanto en la memoria del chip como en el rendimiento del sistema. Estas investigaciones se llevan a cabo en el contexto de estandarización de DVB-NGH (Digital Video Broadcasting - Next Generation Handheld), la evolución portátil de DVB-T2 (Second Generation Terrestrial), y ATSC 3.0 (Advanced Television Systems Commitee - Third Generation) que incorporan MIMO-BICM como clave tecnológica para superar el límite de Shannon para comunicaciones con una única antena. No obstante, esta tesis doctoral emplea un método genérico tanto para el diseño, análisis y evaluación, por lo que los resultados e ideas pueden ser aplicados a otros sistemas de comunicación inalámbricos que empleen MIMO-BICM.
[CAT] La tecnologia de múltiples entrades i múltiples eixides (MIMO) en xarxes de Televisió Digital Terrestre (TDT) té el potencial d'incrementar l'eficiència espectral i millorar la cobertura de xarxa per a afrontar les demandes d'ús de l'escàs espectre electromagnètic (e.g., designació del dividend digital i la demanda d'espectre per part de les xarxes de comunicacions mòbils), l'aparició de nous continguts d'alta taxa de dades (e.g., ultra-high deffinition TV - UHDTV) i la ubiqüitat del contingut (e.g., fix, portàtil i mòbil). És àmpliament reconegut que MIMO pot proporcionar múltiples beneficis com: potència rebuda addicional gràcies als guanys de array, major robustesa contra esvaïments del senyal gràcies a la diversitat espacial i majors taxes de transmissió gràcies al guany per multiplexat del canal MIMO. Aquests beneficis es poden aconseguir sense incrementar la potència transmesa ni l'ample de banda, però normalment s'obtenen a costa d'una major complexitat del sistema tant en el transmissor com en el receptor. Els guanys de rendiment finals a causa de l'ús de MIMO depenen directament de les característiques físiques de l'entorn de propagació com: la correlació entre els canals espacials, l'orientació de les antenes, i/o els desequilibris de potència patits en les antenes transmissores. Addicionalment, a causa de restriccions en la complexitat i aritmètica de precisió finita en els receptors, és fonamental per al rendiment global del sistema un disseny acurat d'algorismes específics de processament de senyal. Aquesta tesi doctoral se centra en el processament de senyal tant en el transmissor com en el receptor per a sistemes TDT que implementen MIMO-BICM (Bit-Interleaved Coded Modulation) sense canal de tornada cap al transmissor des dels receptors. En el transmissor aquesta tesi presenta recerques en precoding MIMO en sistemes TDT per a superar les degradacions del sistema degudes a diferents condicions del canal. En el receptor es presta especial atenció al disseny i avaluació de receptors pràctics MIMO-BICM basats en informació quantificada i al seu impacte tant en la memòria del xip com en el rendiment del sistema. Aquestes recerques es duen a terme en el context d'estandardització de DVB-NGH (Digital Video Broadcasting - Next Generation Handheld), l'evolució portàtil de DVB-T2 (Second Generation Terrestrial), i ATSC 3.0 (Advanced Television Systems Commitee - Third Generation) que incorporen MIMO-BICM com a clau tecnològica per a superar el límit de Shannon per a comunicacions amb una única antena. No obstant açò, aquesta tesi doctoral empra un mètode genèric tant per al disseny, anàlisi i avaluació, per la qual cosa els resultats i idees poden ser aplicats a altres sistemes de comunicació sense fils que empren MIMO-BICM.
Vargas Paredero, DE. (2016). Transmit and Receive Signal Processing for MIMO Terrestrial Broadcast Systems [Tesis doctoral no publicada]. Universitat Politècnica de València. https://doi.org/10.4995/Thesis/10251/66081
TESIS
Premiado
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Zanetti, Ricardo Antonio 1978. „Separação de eventos sísmicos por métodos de decomposição de sinais“. [s.n.], 2013. http://repositorio.unicamp.br/jspui/handle/REPOSIP/259285.
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Orientadores: João Marcos Travassos Romano, Leonardo Tomazeli DuarteDissertação (mestrado) - Universidade Estadual de Campinas, Faculdade de Engenharia Elétrica e de Computação
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Mestrado
Telecomunicações e Telemática
Mestre em Engenharia Elétrica
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Van, den Broeck Samuel. „Optique statistique appliquée à la granulométrie submicronique : simulation d'un signal gaussien lorentzien“. Rouen, 1998. http://www.theses.fr/1998ROUES020.
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Ce travail a pour objet de simuler le signal généré par des particules lors d'une diffusion quasi élastique de la lumière. Le but est de disposer d'un outil numérique susceptible de reproduire diverses configurations expérimentales et de pouvoir évaluer de nouvelles méthodes d'analyses du signal enregistré. La simulation proposée génère les temps d'arrivée des photoélectrons issues d'un photomultiplicateur. Nous nous sommes limités ici au cas connu théoriquement d'un ensemble de particules sphériques monodispersées. Les différentes étapes du calcul sont : 1. Génération d'un signal gaussien lorentzien à partir d'un bruit blanc filtre. Ce signal simule le champ diffusé par les particules ; 2. Génération d'un processus de poisson non homogène à partir du signal gaussien lorentzien. Ce processus simule la réponse de la photocathode à l'intensité diffusée. Le nombre de points maximum généré par fichier est de 524288 (2#1#9). La validité de la simulation est testée en comparant des statistiques de premier et de deuxième ordre du champ diffusé par les particules avec leurs valeurs initiales. La densité de probabilité des intervalles de temps des photoélectrons et la fonction d'autocorrélation du nombre de photocoups sont comparées favorablement avec leurs expressions théoriques respectives.
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Pavan, Flávio Renê Miranda. „Sobre a desconvolução multiusuário e a separação de fontes“. Universidade de São Paulo, 2016. http://www.teses.usp.br/teses/disponiveis/3/3142/tde-22092016-103501/.
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Os problemas de separação cega de fontes e desconvolução cega multiusuário vêm sendo intensamente estudados nas últimas décadas, principalmente devido às inúmeras possibilidades de aplicações práticas. A desconvolução multiusuário pode ser compreendida como um problema particular de separação de fontes em que o sistema misturador é convolutivo, e as estatísticas das fontes, que possuem alfabeto finito, são bem conhecidas. Dentre os desafios atuais nessa área, cabe destacar que a obtenção de soluções adaptativas para o problema de separação cega de fontes com misturas convolutivas não é trivial, pois envolve ferramentas matemáticas avançadas e uma compreensão aprofundada das técnicas estatísticas a serem utilizadas. No caso em que não se conhece o tipo de mistura ou as estatísticas das fontes, o problema é ainda mais desafiador. Na área de Processamento Estatístico de Sinais, soluções vêm sendo propostas para resolver casos específicos. A obtenção de algoritmos adaptativos eficientes e numericamente robustos para realizar separação cega de fontes, tanto envolvendo misturas instantâneas quanto convolutivas, ainda é um desafio. Por sua vez, a desconvolução cega de canais de comunicação vem sendo estudada desde os anos 1960 e 1970. A partir de então, várias soluções adaptativas eficientes foram propostas nessa área. O bom entendimento dessas soluções pode sugerir um caminho para a compreensão aprofundada das soluções existentes para o problema mais amplo de separação cega de fontes e para a obtenção de algoritmos eficientes nesse contexto. Sendo assim, neste trabalho (i) revisitam-se a formulação dos problemas de separação cega de fontes e desconvolução cega multiusuário, bem como as relações existentes entre esses problemas, (ii) abordam-se as soluções existentes para a desconvolução cega multiusuário, verificando-se suas limitações e propondo-se modificações, resultando na obtenção de algoritmos com boa capacidade de separação e robustez numérica, e (iii) relacionam-se os critérios de desconvolução cega multiusuário baseados em curtose com os critérios de separação cega de fontes.Blind source separation and blind deconvolution of multiuser systems have been intensively studied over the last decades, mainly due to the countless possibilities of practical applications. Blind deconvolution in the multiuser case can be understood as a particular case of blind source separation in which the mixing system is convolutive, and the sources, which exhibit a finite alphabet, have well known statistics. Among the current challenges in this area, it is worth noting that obtaining adaptive solutions for the blind source separation problem with convolutive mixtures is not trivial, as it requires advanced mathematical tools and a thorough comprehension of the statistical techniques to be used. When the kind of mixture or source statistics are unknown, the problem is even more challenging. In the field of statistical signal processing, solutions aimed at specific cases have been proposed. The development of efficient and numerically robust adaptive algorithms in blind source separation, for either instantaneous or convolutive mixtures, remains an open challenge. On the other hand, blind deconvolution of communication channels has been studied since the 1960s and 1970s. Since then, various types of efficient adaptive solutions have been proposed in this field. The proper understanding of these solutions can suggest a path to further understand the existing solutions for the broader problem of blind source separation and to obtain efficient algorithms in this context. Consequently, in this work we (i) revisit the problem formulation of blind source separation and blind deconvolution of multiuser systems, and the existing relations between these problems, (ii) address the existing solutions for blind deconvolution in the multiuser case, verifying their limitations and proposing modifications, resulting in the development of algorithms with proper separation performance and numeric robustness, and (iii) relate the kurtosis based criteria of blind multiuser deconvolution and blind source separation.
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Zhao, Wentao. „Genomic applications of statistical signal processing“. [College Station, Tex. : Texas A&M University, 2008. http://hdl.handle.net/1969.1/ETD-TAMU-2952.
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Gomes, Marco Aurelio Cazarotto 1984. „Filtragem otima para melhorar o desempenho de estimadores DOA-ML“. [s.n.], 2009. http://repositorio.unicamp.br/jspui/handle/REPOSIP/261946.
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Orientador: Amauri LopesDissertação (mestrado) - Universidade Estadual de Campinas, Faculdade de Engenharia Eletrica e de Computação
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Resumo: Abordamos o problema de estimação de direção de chegada (DOA) de ondas planas usando um arranjo de sensores. Na literatura encontramos diversos estimadores para DOA, porém estamos considerando apenas os estimadores de Máxima Verossimilhança (ML) que geram candidatas à estimativa DOA e selecionam as melhores através do critério ML. Também estamos interessados em situações em que o espaçamento angular entre as fontes de sinal é pequeno e a relação sinal-ruído é baixa. Nesse caso temos uma degradação de desempenho associada ao efeito de limiar. Mostramos que este problema pode ser amenizado reduzindo o ruído presente na matriz de covariância dos dados recebidos (snapshots) utilizada para a seleção das candidatas. Propomos então modificar o processo de seleção de candidatas, utilizando uma nova matriz de covariância dos snapshots, calculada após uma filtragem ótima dos dados através de um filtro FIR multibanda. Propomos também modificar a função custo ML para adequá-la às dimensões da matriz de covariância filtrada e para isso apresentamos 3 opções de modificação. As simulações mostram que nossa proposta tem melhor desempenho que os métodos conhecidos, reduzindo significativamente a relação sinal-ruído de limiar.
Abstract: We approached the estimation of direction of arrival (DOA) of plane waves using an array of sensors. In the literature there are several DOA estimators, but we considered only the maximum likelihood (ML) estimators that generate candidates for DOA estimation and select the best one through an ML criterion. We also considered situations where the signal sources are spatially closely spaced and the signal-to-noise ratio is low. In these cases a performance degradation associated with the threshold effect occur. We demonstrated that we can improve the estimation performance by reducing the noise in the received data covariance matrix used to select the candidates. Then we proposed to modify the selection process using a new data covariance matrix, computed after an optimum multiband FIR filtering of the received data. We also proposed to modify the ML cost function to adapt it to the dimensions of the new covariance matrix and we considered 3 alternatives of modification. Some simulations showed that our proposal has better performance than known DOA methods, significantly reducing the threshold SNR.
Mestrado
Telecomunicações e Telemática
Mestre em Engenharia Elétrica
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Vollgraf, Roland. „Unsupervised learning methods for statistical signal processing“. [S.l.] : [s.n.], 2006. http://opus.kobv.de/tuberlin/volltexte/2007/1488.
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Eng, Frida. „Non-Uniform Sampling in Statistical Signal Processing“. Doctoral thesis, Linköping : Department of Electrical Engineering, Linköpings universitet, 2007. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-8480.
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Bornn, Luke. „Statistical solutions for and from signal processing“. Thesis, University of British Columbia, 2008. http://hdl.handle.net/2429/5345.
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With the wide range of fields engaging in signal processing research, many
methods do not receive adequate dissemination across disciplines due to
differences in jargon, notation, and level of rigor. In this thesis, I attempt to
bridge this gap by applying two statistical techniques originating in signal
processing to fields for which they were not originally intended. Firstly,
I employ particle filters, a tool used for state estimation in the physics
signal processing world, for the task of prior sensitivity analysis and cross
validation in Bayesian statistics. Secondly, I demonstrate the application of
support vector forecasters, a tool used for forecasting in the machine learning
signal processing world, to the field of structural health monitoring.
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Sallee, Philip Andrew. „Statistical methods for image and signal processing /“. For electronic version search Digital dissertations database. Restricted to UC campuses. Access is free to UC campus dissertations, 2004. http://uclibs.org/PID/11984.
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Huber, Stefan. „Voice Conversion by modelling and transformation of extended voice characteristics“. Electronic Thesis or Diss., Paris 6, 2015. https://accesdistant.sorbonne-universite.fr/login?url=https://theses-intra.sorbonne-universite.fr/2015PA066750.pdf.
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La Conversion de la Voix (VC) vise à transformer les caractéristiques de la voix d’un locuteur source de manière qu’il sera perçu comme étant prononcé par un locuteur cible. Le principe de la VC est de définir des fonctions du transposition pour la conversion de la voix de l’un locuteur source à la voix de l’un locuteur cible. Les fonctions de transformation de VC systèmes "State-Of-The-Art" (START) adapte instantanément aux caractéristiques de la voix source. Cependant, la qualité est pas encore suffisant. Des améliorations considérables sont nécessaires que les techniques VC peuvent être utilisés dans un environnement industriel professionnel. L’objectif de cette thèse est d’augmenter la qualité de la conversion de la voix pour faciliter son applicabilité industrielle dans une mesure raisonnable. Les propriétés de base de différentes START algorithmes de la conversion de la voix sont discutés sur leurs avantages intrinsèques et ses déficits. Basé sur des évaluations expérimentales avec un GMM VC système la conclusion est que la plupart des systèmes VC START qui reposent sur des modèles statistiques sont, en raison de l’effet en moyenne de la régression linéaire, moins appropriées pour atteindre un score du similitude assez élevé avec le haut-parleur cible requise pour l’utilisation industrielle. Les contributions établies pendant de ce travail de thèse se trouvent dans les moyens étendus à a) modéliser l’excitation du source glottique, b) modéliser des descripteurs de la voix en utilisant un nouveau système de parole basée sur un modèle élargie de source-filtre, et c) avancer une nouveau système VC de l’Ircam en le combinant avec les contributions de a) et b)Voice Conversion (VC) aims at transforming the characteristics of a source speaker’s voice in such a way that it will be perceived as being uttered by a target speaker. The principle of VC is to define mapping functions for the conversion from one source speaker’s voice to one target speaker’s voice. The transformation functions of common State-Of-The-Art (START) VC system adapt instantaneously to the characteristics of the source voice. While recent VC systems have made considerable progress over the conversion quality of initial approaches, the quality is nevertheless not yet sufficient. Considerable improvements are required before VC techniques can be used in an professional industrial environment. The objective of this thesis is to augment the quality of Voice Conversion to facilitate its industrial applicability to a reasonable extent. The basic properties of different START algorithms for Voice Conversion are discussed on their intrinsic advantages and shortcomings. Based on experimental evaluations of one GMM-based State-Of-The-Art VC approach the conclusion is that most VC systems which rely on statistical models are, due to averaging effect of the linear regression, less appropriate to achieve a high enough similarity score to the target speaker required for industrial usage. The contributions established throughout this thesis work lie in the extended means to a) model the glottal excitation source, b) model a voice descriptor set using a novel speech system based on an extended source-filter model, and c) to further advance IRCAM’s novel VC system by combining it with the contributions of a) and b)
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Farag, Emad N. „VLSI low-power digital signal processing“. Thesis, National Library of Canada = Bibliothèque nationale du Canada, 1997. http://www.collectionscanada.ca/obj/s4/f2/dsk3/ftp04/nq22199.pdf.
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Ekstam, Ljusegren Hannes, und Hannes Jonsson. „Parallelizing Digital Signal Processing for GPU“. Thesis, Linköpings universitet, Programvara och system, 2020. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-167189.
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Because of the increasing importance of signal processing in today's society, there is a need to easily experiment with new ways to process signals. Usually, fast-performing digital signal processing is done with special-purpose hardware that are difficult to develop for. GPUs pose an alternative for fast performing digital signal processing. The work in this thesis is an analysis and implementation of a GPU version of a digital signal processing chain provided by SAAB. Through an iterative process of development and testing, a final implementation was achieved. Two benchmarks, both comprised of 4.2 M test samples, were made to compare the CPU implementation with the GPU implementation. The benchmark was run on three different platforms: a desktop computer, a NVIDIA Jetson AGX Xavier and a NVIDIA Jetson TX2. The results show that the parallelized version can reach several magnitudes higher throughput than the CPU implementation.
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Feiste, Kurt Alan. „Merged arithmetic for digital signal processing /“. Digital version accessible at:, 1999. http://wwwlib.umi.com/cr/utexas/main.
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Xu, Cuichun. „Statistical processing on radar, sonar, and optical signals /“. View online ; access limited to URI, 2008. http://0-digitalcommons.uri.edu.helin.uri.edu/dissertations/AAI3328735.
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Ljungqvist, Martin. „Bayesian Decoding for Improved Random Access in Compressed Video Streams“. Thesis, Linköping University, Department of Science and Technology, 2005. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-297.
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A channel change in digital television is usually conducted at a reference frame, which are sent at certain intervals. A higher compression ratio could however be obtained by sending reference frames at arbitrary long intervals. This would on the other hand increase the average channel change time for the end user. This thesis investigates various approaches for reducing the average channel change time while using arbitrary long intervals between reference frames, and presents an implementation and evaluation of one of these methods, called Baydec.
The approach of Baydec for solving the channel switch problem is to statistically estimate what the original image looked like, starting with an incoming P-frame and estimate an image between the original and current image. Baydec gathers statistical data from typical video sequences and calculates expected likelihood for estimation. Further on it uses the Simulated Annealing search method to maximise the likelihood function.
This method is more general than the requirements of this thesis. It is not only applicable to channel switches between video streams, but can also be used for random access in general. Baydec could also be used if an I-frame is dropped in a video stream.
However, Baydec has so far shown only theoretical result, but very small visual improvements. Baydec produces images with better PSNR than without the method in some cases, but the visual impression is not better than for the motion compensated residual images. Some examples of future work to improve Baydec is also presented.
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Kuchler, Ryan J. „Theory of multirate statistical signal processing and applications“. Monterey, Calif. : Springfield, Va. : Naval Postgraduate School ; Available from National Technical Information Service, 2005. http://library.nps.navy.mil/uhtbin/hyperion/05Sep%5FKuchler%5FPhD.pdf.
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Vigoda, Benjamin William 1973. „Continuous-time analog circuits for statistical signal processing“. Thesis, Massachusetts Institute of Technology, 2003. http://hdl.handle.net/1721.1/62962.
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Thesis (Ph. D.)--Massachusetts Institute of Technology, School of Architecture and Planning, Program in Media Arts and Sciences, 2003.Vita.
Includes bibliographical references (p. 205-209).
This thesis proposes an alternate paradigm for designing computers using continuous-time analog circuits. Digital computation sacrifices continuous degrees of freedom. A principled approach to recovering them is to view analog circuits as propagating probabilities in a message passing algorithm. Within this framework, analog continuous-time circuits can perform robust, programmable, high-speed, low-power, cost-effective, statistical signal processing. This methodology will have broad application to systems which can benefit from low-power, high-speed signal processing and offers the possibility of adaptable/programmable high-speed circuitry at frequencies where digital circuitry would be cost and power prohibitive. Many problems must be solved before the new design methodology can be shown to be useful in practice: Continuous-time signal processing is not well understood. Analog computational circuits known as "soft-gates" have been previously proposed, but a complementary set of analog memory circuits is still lacking. Analog circuits are usually tunable, rarely reconfigurable, but never programmable. The thesis develops an understanding of the convergence and synchronization of statistical signal processing algorithms in continuous time, and explores the use of linear and nonlinear circuits for analog memory. An exemplary embodiment called the Noise Lock Loop (NLL) using these design primitives is demonstrated to perform direct-sequence spread-spectrum acquisition and tracking functionality and promises order-of-magnitude wins over digital implementations. A building block for the construction of programmable analog gate arrays, the "soft-multiplexer" is also proposed.
by Benjamin Vigoda.
Ph.D.
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Vallet, Pascal. „Random matrices and applications to statistical signal processing“. Thesis, Paris Est, 2011. http://www.theses.fr/2011PEST1055/document.
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Dans cette thèse, nous considérons le problème de la localisation de source dans les grands réseaux de capteurs, quand le nombre d'antennes du réseau et le nombre d'échantillons du signal observé sont grands et du même ordre de grandeur. Nous considérons le cas où les signaux source émis sont déterministes, et nous développons un algorithme de localisation amélioré, basé sur la méthode MUSIC. Pour ce faire, nous montrons de nouveaux résultats concernant la localisation des valeurs propres des grandes matrices aléatoires gaussiennes complexes de type information plus bruitIn 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
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Palladini, Alessandro <1981>. „Statistical methods for biomedical signal analysis and processing“. Doctoral thesis, Alma Mater Studiorum - Università di Bologna, 2009. http://amsdottorato.unibo.it/1358/1/palladini_alessandro_tesi.pdf.
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Statistical modelling and statistical learning theory are two powerful analytical frameworks for analyzing signals and developing efficient processing and classification algorithms. In this thesis, these frameworks are applied for modelling and processing biomedical signals in two different contexts: ultrasound medical imaging systems and primate neural activity analysis and modelling.
In the context of ultrasound medical imaging, two main applications are explored: deconvolution of signals measured from a ultrasonic transducer and automatic image segmentation and classification of prostate ultrasound scans.
In the former application a stochastic model of the radio frequency signal measured from a ultrasonic transducer is derived. This model is then employed for developing in a statistical framework a regularized deconvolution procedure, for enhancing signal resolution.
In the latter application, different statistical models are used to characterize images of prostate tissues, extracting different features. These features are then uses to segment the images in region of interests by means of an automatic procedure based on a statistical model of the extracted features. Finally, machine learning techniques are used for automatic classification of the different region of interests.
In the context of neural activity signals, an example of bio-inspired dynamical network was developed to help in studies of motor-related processes in the brain of primate monkeys. The presented model aims to mimic the abstract functionality of a cell population in 7a parietal region of primate monkeys, during the execution of learned behavioural tasks.
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Palladini, Alessandro <1981>. „Statistical methods for biomedical signal analysis and processing“. Doctoral thesis, Alma Mater Studiorum - Università di Bologna, 2009. http://amsdottorato.unibo.it/1358/.
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Statistical modelling and statistical learning theory are two powerful analytical frameworks for analyzing signals and developing efficient processing and classification algorithms. In this thesis, these frameworks are applied for modelling and processing biomedical signals in two different contexts: ultrasound medical imaging systems and primate neural activity analysis and modelling.
In the context of ultrasound medical imaging, two main applications are explored: deconvolution of signals measured from a ultrasonic transducer and automatic image segmentation and classification of prostate ultrasound scans.
In the former application a stochastic model of the radio frequency signal measured from a ultrasonic transducer is derived. This model is then employed for developing in a statistical framework a regularized deconvolution procedure, for enhancing signal resolution.
In the latter application, different statistical models are used to characterize images of prostate tissues, extracting different features. These features are then uses to segment the images in region of interests by means of an automatic procedure based on a statistical model of the extracted features. Finally, machine learning techniques are used for automatic classification of the different region of interests.
In the context of neural activity signals, an example of bio-inspired dynamical network was developed to help in studies of motor-related processes in the brain of primate monkeys. The presented model aims to mimic the abstract functionality of a cell population in 7a parietal region of primate monkeys, during the execution of learned behavioural tasks.
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Kwan, Ching Chung. „Digital signal processing techniques for on-board processing satellites“. Thesis, University of Surrey, 1990. http://epubs.surrey.ac.uk/754893/.
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In on-board processing satellite systems in which FDMA/SCPC access schemes are employed. transmultiplexers are required for the frequency demultiplexing of the SCPC signals. Digital techniques for the implementation of the transmultiplexer for such application were examined in this project. The signal processing in the transmultiplexer operations involved many parameters which could be optimized in order to reduce the hardware complexity whilst satisfying the level of performance required of the system. An approach for the assessment of the relationship between the various parameters and the system performance was devised. which allowed hardware requirement of practical system specifications to be estimated. For systems involving signals of different bandwidths a more flexible implementation of the trans multiplexer is required and two computationally efficient methods. the DFT convolution and analysis/synthesis filter bank. were investigated. These methods gave greater flexibility to the input frequency plan of the transmultiplexer. at the expense of increased computational requirements. Filters were then designed to exploit specific properties of the flexible transmultiplexer methods. resulting in considerable improvement in their efficiencies. Hardware implementation of the flexible transmultiplexer was considered and an efficient multi-processor architecture in combination with parallel processing software algorithms for the signal processing operations were designed. Finally. an experimental model of the payload for a land-mobile satellite system proposal. T -SAT. was constructed using general-purpose digital signal processors and the merits of the on-board processing architecture was demonstrated.
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DI, NUNZIO LUCA. „Reconfigurable digital architecture for high speed digital signal processing“. Doctoral thesis, Università degli Studi di Roma "Tor Vergata", 2010. http://hdl.handle.net/2108/1295.
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Low cost microprocessors and DSPs are optimized to perform arithmetic and logic
operations on data having a xed size, typically 16,32 or 64 bit. On the other hand,
their e ciency decreases when data shorter respect than their native wordlength
are processed (more clock cycles per operation are required). Recently di erent
solutions have been proposed to overcome this problem. Among those, the ones
based on a main processor with a Recon gurable Unit used as hardware accelerator
are the most interesting in terms of performance and
exibility. Typically those
architectures are similar to very small FPGA; they consist in arrays of Look-Up
Tables (LUTs) interconnected by pass transistors networks.
This work proposes a new Recon gurable Accelerator called ADAPTO (Adderbased
Dynamic Architecture for Processing Tailored Operators). The main di erent
between ADAPTO and the others Recon gurable Units proposed in literature is the
reduced hardware complexity in terms of silicon area. This feature give the possibility
to integrate ADAPTO in embedded low cost microprocessors and DSPs (Digital
Signal Processors), in fact, for these kind of processors, the area occupation and
therefore the cost is a very critical aspect. The ADAPTO Unit supports both hardware
recon guration and instruction execution in the same processor clock cycle.
These goals have been obtained with the multicontext approach using a recon gurable
unit based on full adders, instead LUTs. As discussed in this work this choice
allows to the multicontext technique a reduced wasting of hardware resources.
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Wang, Limin. „The ECG signal processing by ADSP-21062 digital signal processor“. Morgantown, W. Va. : [West Virginia University Libraries], 1999. http://etd.wvu.edu/templates/showETD.cfm?recnum=840.
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Thesis (M.S.)--West Virginia University, 1999.Title from document title page. Document formatted into pages; contains vi, 110 p. : ill. (some col.) Includes abstract. Includes bibliographical references (p. 66-68).
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Nordström, Jesper. „Real time digital signal processing using Matlab“. Thesis, Uppsala universitet, Signaler och System, 2017. http://urn.kb.se/resolve?urn=urn:nbn:se:uu:diva-332075.
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Increased usage of electronic devices and the fast development of microprocessors has increased the usage of digital filters ahead of analog filters. Digital filters offer great benefits over analog filters in that they are inexpensive, they can be reprogrammed easily and they open up whole new range of possibilities when it comes to Internet of things. This thesis describes development of a program that can sample music from the computer's microphone input, filter it inside the program with user built filters and reconstruct the music to the computer's headphone output meaning that the music can be played from the speakers. All of this is to happen in real time. The program is developed for students studying at the department of ``Signals and Systems" and the program is supposed the be one of the educational tools to make sense of signals and filtering. The program works well and filters the sound with satisfying results. It is easy to create filters and filter the signal. Since it is music that is filtered constructing perfect filters with minimum ripple, minimum or linear phase is quite difficult to achieve. The program could be improved by improving the user interface, making the environment more interactive and less difficult to construct good filters. Some improvements could also be made to the implementation; as of now the program might run a bit slow on startup on slower computers.
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Yang, Shijun. „Smart receiver using baseband digital signal processing“. Thesis, National Library of Canada = Bibliothèque nationale du Canada, 2000. http://www.collectionscanada.ca/obj/s4/f2/dsk1/tape4/PQDD_0017/MQ48478.pdf.
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Papaspiridis, Alexandros. „Digital signal processing techniques for gene prediction“. Thesis, Imperial College London, 2012. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.590037.
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The purpose of this research is to apply existing Digital Signal Processing techniques to DNA sequences, with the objective of developing improved methods for gene prediction. Sections of DNA sequences are analyzed in the frequency domain and frequency components that distinguish intron regions are identified (21t/lOA). Novel detectors are created using digital filters and auto correlation, capable of identifying the location of intron regions in a sequence. The resulting signal from these detectors is used as a dynamic threshold in existing gene detectors, resulting in an improved accuracy of 12% and 25% respectively. Finally, DNA sequences are analyzed in terms of their amino acid composition, and new gene prediction algorithms are introduced.
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Roome, Stephen John. „The industrial application of digital signal processing“. Thesis, City University London, 1989. http://openaccess.city.ac.uk/7405/.
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This thesis describes an investigation into the application of digital signal processing techniques to the solution of industrial signal processing problems. The investigation took the form of three case studies chosen to illustrate the variety of possible applications. The first was the computer simulation of a digital microwave communications link which utilised narrowband FM modulation and partial response techniques. In order to ensure that the behaviour of the simulation reliably matched that of the modelled system it was found necessary to have a sound theoretical background, implementation using good software engineering methodology together with methodical testing and validation. The second case study was a comprehensive investigation of adaptive noise cancelling systems concentrating on issues important for practical implementation of the technique: stability and convergence of the adaptation algorithm; misadjustment noise and effects due to realizability constraints. It was found that theoretical predictions of the systems behaviour were in good agreement with the results of computer simulation except for the level of output misadjustment noise. In order to make the mathematics of the LMS algorithm tractable it was assumed that the input data formed a series of uncorrelated vectors. It was found that this assumption is only appropriate for the prediction of misadjustment noise when the reference input is uncorrelated. The final case study concerned the automatic detection and assessment of pressing faults on gramophone records for quality assurance purposes. A pattern recognition technique for identifying the signals due to gramophone record defects and a numerical method for assessing the perceived severity of the defects were developed empirically. Prototype equipment was designed, built and tested in extended field trials. The equipment was shown to be superior to previous equipment developed using analogue signal processing techniques. These case studies demonstrate that digital signal processing is a powerful and widely applicable technique for the solution of industrial signal processing problems. Solutions may be theoretical or obtained by experiment or simulation. The strengths and weaknesses of each approach are illustrated.
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Esparcia, Alcázar Anna Isabel. „Genetic programming for adaptive digital signal processing“. Thesis, University of Glasgow, 1998. http://theses.gla.ac.uk/4780/.
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Wells, Ian. „Digital signal processing architectures for speech recognition“. Thesis, University of the West of England, Bristol, 1995. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.294705.
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Yang, Shijun Carleton University Dissertation Engineering Electronics. „Smart receiver using baseband digital signal processing“. Ottawa, 1999.
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Vrcelj, Bojan Vaidyanathan P. P. „Multirate signal processing concepts in digital communications /“. Diss., Pasadena, Calif. : California Institute of Technology, 2004. http://resolver.caltech.edu/CaltechETD:etd-06252003-115639.
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Musoke, David. „Digital image processing with the Motorola 56001 digital signal processor“. Scholarly Commons, 1992. https://scholarlycommons.pacific.edu/uop_etds/2236.
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This report describes the design and testing of the Image56 system, an IBM-AT based system which consists of an analog video board and a digital board. The former contains all analog and video support circuitry to perform real-time image processing functions. The latter is responsible for performing non real-time, complex image processing tasks using a Motorola DSP56001 digital signal processor. It is supported by eight image data buffers and 512K words of DSP memory (see Appendix A for schematic diagram).
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Okullo-Oballa, Thomas Samuel. „Systolic realization of multirate digital filters“. Thesis, [Hong Kong] : University of Hong Kong, 1988. http://sunzi.lib.hku.hk/hkuto/record.jsp?B12433998.
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Ali-Bakhshian, Mohammad. „Digital processing of analog information adopting time-mode signal processing“. Thesis, McGill University, 2013. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=114237.
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As CMOS technologies advance to 22-nm dimensions and below, constructing analog circuits in such advanced processes suffers many limitations, such as reduced signal swings, sensitivity to thermal noise effects, loss of accurate switching functions, to name just a few. Time-Mode Signal Processing (TMSP) is a technique that is believed to be well suited for solving many of these challenges. It can be defined as the detection, storage, and manipulation of sampled analog information using time-mode variables. One of the important advantages of TMSP is the ability to realize analog functions using digital logic structures. This technique has a long history of application in electronics; however, due to lack of some fundamental functions, the use of TM variables has been mostly limited to intermediate stage processing and it has been always associated with voltage/current-to-time and time-to-voltage/current conversion. These conversions necessitate the inclusion of analog blocks that contradict the digital advantage of TMSP. In this thesis, an intensive research has been presented that provides an appropriate foundation for the development of TMSP as a general processing tool. By proposing the new concept of delay interruption, a completely new asynchronous approach for the manipulation of TM variables is suggested. As a direct result of this approach, practical techniques for storage, addition and subtraction of time-mode variables are presented. To Extend the digital implementation of TMSP to a wider range of applications, the comprehensive design of a unity gain dual-path time-to-time integrator (accumulator) is demonstrated. This integrator is then used to implement a digital second-order delta-sigma modulator. Finally, to demonstrate the advantage of TMSP, a very low power and compact tunable interface for capacitive sensors is presented that is composed of a number of delay blocks associated with typical logic gates. All the proposed theories are supported by experimental results and post-layout simulations.The emphasis on the digital construction of the proposed circuits has been the first priority of this thesis. Having the building blocks implemented with a digital structure, provides the feasibility of a simple, synthesizable, and reconfigurable design where affordable circuit calibrations can be adopted to remove the effects of process variations.Les technologies CMOS progressant vers les procédés 22 nm et au delà, la abrication des circuits analogiques dans ces technologies se heurte a de nombreuses limitations. Entre autres limitations on peut citer la réduction d'amplitude des signaux, la sensibilité aux effets du bruit thermique et la perte de fonctions précises de commutation. Le traitement de signal en mode temps (TMSP pour Time-Mode Signal Processing) est une technique que l'on croit être bien adapté pour résoudre un grand nombre de problèmes relatifs a ces limitations. TMSP peut être défini comme la détection, le stockage et la manipulation de l'information analogique échantillonnée en utilisant des quantités de temps comme variables. L'un des avantages importants de TMSP est la capacité à réaliser des fonctions analogiques en utilisant des structures logiques digitales. Cette technique a une longue histoire en terme d'application en électronique. Cependant, en raison du manque de certaines fonctions fondamentales, l'utilisation de variables en mode temps a été limitée à une utilisation comme étape intermédiaire dans le traitement d'un signal et toujours dans le contexte d'une conversion tension/courant-temps et temps-tension/courant. Ces conversions nécessitent l'inclusion de blocs analogiques qui vont a l'encontre de l'avantage numérique des TMSP. Cette thèse fournit un fondement approprié pour le développement de TMSP comme outil général de traitement de signal. En proposant le concept nouveau d'interruption de retard, une toute nouvelle approche asynchrone pour la manipulation de variables en mode temps est suggéré. Comme conséquence directe de cette approche, des techniques pratiques pour le stockage, l'addition et la soustraction de variables en mode temps sont présentées. Pour étendre l'implémentation digitale de TMSP à une large gamme d'applications, la conception d'un intégrateur (accumulateur) à double voie temps- à -temps est démontrée. cet intégrateur est ensuite utilisé pour implémenter un modulateur delta-sigma de second ordre.Enfin, pour démontrer l'avantage de TMSP, une Interface de très basse puissance, compacte et réglable pour capteurs capacitifs est présenté. Cette interface est composé d'un certain nombre de blocs de retard associés à des portes logiques typiques. Toutes les théories proposées sont soutenues par des résultats expérimentaux et des simulations post-layout. L'implémentation digitale des circuits proposés a été la première priorité de cette thèse. En effet, une implémentation des bloc avec des structures digitales permet des conceptions simples, synthétisable et reconfigurables où des circuits de calibration très abordables peuvent être adoptées pour éliminer les effets des variations de process.
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Noor, Fazal. „Inverse and Eigenspace decomposition algorithms for statistical signal processing“. Thesis, McGill University, 1993. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=39489.
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In this work, a number of advances are described which we feel lead to better understanding and solution of the eigenvalue and inverse eigenvalue problems for Hermitian Toeplitz matrices. First, a unitary matrix is derived which transforms a Hermitian Toeplitz matrix into a real Toeplitz plus Hankel matrix. Some properties of this transformation are also presented. Second, we solve the inverse eigenvalue problem for Hermitian Toeplitz matrices. Specifically, we present a method for the construction of a Hermitian Toeplitz matrix from an arbitrary set of real eigenvalues. The procedure utilizes the discrete Fourier transform to first construct a real symmetric negacyclic matrix from the specified eigenvalues. The algorithm presented is computationally efficient. Finally, we derive a new order recursive algorithm and modify Trench's algorithm, both for eigenvalue decomposition. The former development is of mathematical interest; whereas, the latter is clearly of practical interest. The modifications proposed to Trench's algorithm are to employ noncontiguous intervals and to include a procedure to detect multiple eigenvalues. The goals of the modification are to improve the rate of convergence. The modified algorithm presented utilizes three root searching techniques: the Pegasus method, the modified Rayleigh quotient iteration with bisection shifts (MRQI-B), and the MRQI with Pegasus shifts (MRQI-P). Simulation results are provided for large matrices of orders 50, 100, 200, and 500. Application of the algorithms to Pisarenko's harmonic decomposition, an important signal processing problem, is presented. Fortran programs of the modified method are also provided.
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Hill, S. „Applications of statistical learning theory to signal processing problems“. Thesis, University of Cambridge, 2003. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.604048.
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The dissertation focuses on the applicability of Support Vector Regression (SVR) in signal processing contexts. This is shown to be particularly well-suited to filtering in alpha-stable noise environments, and a further slight modification is proposed to this end. The main work in this dissertation on SVR is on the application to audio filtering based on perceptual criteria. This appears an ideal solution to the problem due to the fact that the loss function typically used by perceptual audio filtering practitioners incorporates a region of zero loss, as does SVR. SVR is extended to the problem of complex-valued regression, for application in the audio filtering problem to the frequency domain. This is with regions of zero loss that are both square and circular, and the circular case is extended to the problem of vector-valued regression. Three experiments are detailed with a mix of both good and poor results, and further refinements are proposed. Polychotomous, or multi-category classification is then studied. Many previous attempts are reviewed, and compared. A new approach is proposed, based on a geometrical structure. This is shown to overcome many of the problems identified with previous methods in addition to being very flexible and efficient in its implementation. This architecture is also derived, for just the three-class case, using a complex-valued kernel function. The general architecture is used experimentally in three separate implementations to give a demonstration of the overall approach. The methodology is shown to achieve results comparable to those of many other methods, and to include many of them as special cases. Further possible refinements are proposed which should drastically reduce optimisation times for so-called 'all-together' methods.
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Chan, Tsang Hung. „Digital signal processing in optical fibre digital speckle pattern interferometry“. HKBU Institutional Repository, 1996. http://repository.hkbu.edu.hk/etd_ra/269.
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Orcutt, Edward Kerry 1964. „Correlation filters for time domain signal processing“. Thesis, The University of Arizona, 1989. http://hdl.handle.net/10150/277215.
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This study proposes employing new filters in various configurations for use in digital communication systems. We believe that significant improvements in such performance areas as transmission rate and synchronization may be achieved by incorporating these filters into digital communications receivers. Recently reported in the literature, these filters may offer advantages over the matched filter which allow enhancements in data rates, ISI tolerance, and synchronization. To make full use of the benefits of these filters, we introduce the concept of parallel signal transmission over a single channel. We also examine the effects of signal set selection and noise on performance.
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Lopes, Wenderson Nascimento. „Investigação do conteúdo harmônico do sinal de emissão acústica na dressagem de rebolos de óxido de alumínio com dressador de ponta única“. Universidade Estadual Paulista (UNESP), 2018. http://hdl.handle.net/11449/154406.
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Coordenação de Aperfeiçoamento de Pessoal de Nível Superior (CAPES)
A dressagem é uma operação muito importante para o processo de retificação. O objetivo desta é recondicionar o rebolo para restabelecer suas características de corte perdidas devido ao desgaste produzido após sucessivos passes. Sistemas de monitoramento que utilizam emissão acústica (EA) têm sido propostos para correlacionar os sinais com diversas condições da ferramenta. Este estudo traz uma nova abordagem de processamento de sinais de EA com o objetivo de identificar o momento correto para finalizar a dressagem, o que é essencial em um sistema de controle automático. A partir dos sinais de EA, coletados em testes de dressagem de rebolo de óxido de alumínio com dressador de ponta única, a análise espectral foi realizada por meio da Densidade Espectral de Potência (PSD, Power Spectral Density), selecionando-se bandas de frequências que melhor caracterizam o processo. O parâmetro estatístico "counts" foi aplicado ao sinal original não filtrado e filtrado nas bandas selecionadas para identificar a condição da ferramenta e, por sua vez, para a implementação de um sistema de monitoramento. Os resultados mostraram uma relação expressiva entre as condições de corte da ferramenta e os sinais processados nas bandas selecionadas. Houve uma grande diferença dos sinais filtrados nas bandas selecionadas e sinais não filtrados, refletindo que os filtrados foram mais eficientes em termos de automação de processos.
Dressing is an important operation for the grinding process. Its goal is to recondition the wheel tool to re-establish its cutting characteristics, owing to the wear produced after successive passes. Monitoring systems that use acoustic emission (AE) have been studied to correlate the signals with several tool conditions. This study brings a new approach of processing AE signals with the purpose of identifying the correct moment to stop the dressing, which is essential in an automatic control system. From the AE signals collected in dressing tests with aluminium oxide grinding wheel and single-point dresser, spectral analysis was made through power spectral density, selecting frequencies bands that best characterize the process. The statistical parameter ‘counts’ was applied to the raw signal unfiltered and filtered in the selected bands in order to identify the tool condition and, in turn, towards a monitoring system implementation. Results showed an expressive relation between tool cutting conditions and processed signals in the selected bands. There was a great disparity of the filtered signals in the selected bands and signals unfiltered, reflecting that the filtered ones were more efficient in terms of process automation.
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Lei, Chi-un, und 李志遠. „VLSI macromodeling and signal integrity analysis via digital signal processing techniques“. Thesis, The University of Hong Kong (Pokfulam, Hong Kong), 2011. http://hub.hku.hk/bib/B45700588.
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Martinsson, Jesper. „Statistical tools for ultrasonic analysis of dispersive fluids“. Licentiate thesis, Luleå : Luleå University of Technology, 2006. http://epubl.ltu.se/1402-1757/2006/17/.
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Ng, Chiu-wa, und 吳潮華. „Bit-stream signal processing on FPGA“. Thesis, The University of Hong Kong (Pokfulam, Hong Kong), 2009. http://hub.hku.hk/bib/B41633842.
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