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Dissertations / Theses on the topic 'Adaptive Filters'

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Published: 4 June 2021
Last updated: 29 July 2024

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1

Ryan, James Gregory Carleton University Dissertation Engineering Electrical. "Subband adaptive filters." Ottawa, 1992.

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2

Eshghi, Mohammad. "Highly parallel transversal adaptive filter." Ohio : Ohio University, 1988. http://www.ohiolink.edu/etd/view.cgi?ohiou1182785469.

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3

Suhasini, Subba Rao Tata. "Wavelets and adaptive filters." Thesis, University of Bristol, 2001. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.364840.

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4

Chamon, Luiz Fernando de Oliveira. "Combinations of adaptive filters." Universidade de São Paulo, 2015. http://www.teses.usp.br/teses/disponiveis/3/3142/tde-14072016-143633/.

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Adaptive filtering has grown to become a fundamental topic in signal processing, increasingly attracting attention from the community. Important factors in this popularization were their low computational complexity and model-free nature, adapting even to nonstationary characteristics of the systems and/or signals under study. Nevertheless, many adaptive algorithms introduce trade-offs, for instance, between convergence rate, nonstationary signals tracking, and steady-state error, which can hinder their use in practical applications. Furthermore, some adaptive filters can become unstable when word length is reduced and/or the input data are highly correlated. Recently, combination of adaptive filters was put forward as a solution for such issues. This approach consists in combining a pool of filters by means of a supervisor that attempts to make the overall system at least as good (usually in the mean-square sense) as the best filter in the set. Examples of these structures have been shown to successfully solve this problem, although well-known limitations remain to be addressed. Moreover, due to the relative novelty of this topic, developments in combination of adaptive filters are difficult to accommodate into a common theoretical framework. This work studies combination of adaptive filters and addresses the aforementioned issue by (i) classifying the existing combinations and proposing a taxonomy that exposes the similarities and differences in their forms; (ii) proposing new combinations; (iii) devising a general framework for studying combinations of adaptive filters and using such framework in performance analyses.
Filtragem adaptativa vem ganhando destaque desde seu surgimento tornando-se um tópico de estudo fundamental em processamento de sinais. A versatilidade de dispensarem total conhecimento das propriedades estatísticas dos sinais, aliada à simplicidade computacional de seus métodos, foram importantes fatores em sua consagração. Apesar disto, muitos filtros adaptativos apresentam compromissos envolvendo, por exemplo, velocidade de convergência, rastreamento de sinais não-estacionários e erro em regime, que podem dificultar sua aplicação na prática. Ademais, alguns algoritmos adaptativos são instáveis quando suas entradas são altamente correlacionados e/ou a precisão dos cálculos é reduzida. Uma solução recente para estes problemas é o uso de combinações de filtros adaptativos. Esta abordagem baseia-se em combinar um conjunto de filtros por meio de um supervisor que procura fazer com que o sistema global seja pelo menos tão bom (em geral no sentido quadrático médio) quanto o melhor filtro do conjunto. Exemplos destas estruturas já mostraram a eficácia deste método, apesar de ainda existirem reconhecida limitações. Além disso, em se tratando de um tópico relativamente recente, os desenvolvimentos na área de combinação de filtros adaptativos não possuem uma estrutura teórica unificada. Este trabalho propõe abordar estas questões (i) classificando as combinações existentes e criando uma taxonomia que explicite semelhanças e diferenças entre elas; (ii) introduzindo novas combinações; e (iii) desenvolvendo uma forma unificada de descrever combinações de filtros adaptativos e usando-a em análises de desempenho.
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5

Lopes, Wilder Bezerra. "Geometric-algebra adaptive filters." Universidade de São Paulo, 2016. http://www.teses.usp.br/teses/disponiveis/3/3142/tde-22092016-143525/.

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This document introduces a new class of adaptive filters, namely Geometric- Algebra Adaptive Filters (GAAFs). Those are generated by formulating the underlying minimization problem (a least-squares cost function) from the perspective of Geometric Algebra (GA), a comprehensive mathematical language well-suited for the description of geometric transformations. Also, differently from the usual linear algebra approach, Geometric Calculus (the extension of Geometric Algebra to differential calculus) allows to apply the same derivation techniques regardless of the type (subalgebra) of the data, i.e., real, complex-numbers, quaternions etc. Exploiting those characteristics, among others, a general leastsquares cost function is posed, from which two types of GAAFs are designed. The first one, called standard, provides a generalization of regular adaptive filters for any subalgebra of GA. From the obtained update rule, it is shown how to recover the following least-mean squares (LMS) adaptive filter variants: real-entries LMS, complex LMS, and quaternions LMS. Mean-square analysis and simulations in a system identification scenario are provided, showing almost perfect agreement for different levels of measurement noise. The second type, called pose estimation, is designed to estimate rigid transformations { rotation and translation - in n-dimensional spaces. The GA-LMS performance is assessed in a 3-dimensional registration problem, in which it is able to estimate the rigid transformation that aligns two point clouds that share common parts.
Este documento introduz uma nova classe de filtros adaptativos, entitulados Geometric-Algebra Adaptive Filters (GAAFs). Eles s~ao projetados via formulação do problema de minimização (uma função custo de mínimos quadrados) do ponto de vista de álgebra geométrica (GA), uma abrangente linguagem matemática apropriada para a descrição de transformações geométricas. Adicionalmente, diferente do que ocorre na formulação com álgebra linear, cálculo geométrico (a extensão de álgebra geométrica que possibilita o uso de cálculo diferencial) permite aplicar as mesmas técnicas de derivação independentemente do tipo de dados (subálgebra), isto é, números reais, números complexos, quaternions etc. Usando essas e outras características, uma função custo geral de mínimos quadrados é proposta, da qual dois tipos de GAAFs são gerados. O primeiro, chamado standard, generaliza filtros adaptativos da literatura concebidos sob a perspectiva de subálgebras de GA. As seguintes variantes do filtro least-mean squares (LMS) s~ao obtidas como casos particulares: LMS real, LMS complexo e LMS quaternions. Uma análise mean-square é desenvolvida e corroborada por simulações para diferentes níveis de ruído de medição em um cenário de identificação de sistemas. O segundo tipo, chamado pose estimation, é projetado para estimar transformações rígidas - rotação e translação { em espaços n-dimensionais. A performance do filtro GA-LMS é avaliada em uma aplicação de alinhamento tridimensional na qual ele estima a tranformação rígida que alinha duas nuvens de pontos com partes em comum.
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6

FACEROLI, SILVANA TEREZINHA. "STABILIZATION OF ADAPTIVE IIR FILTERS." PONTIFÍCIA UNIVERSIDADE CATÓLICA DO RIO DE JANEIRO, 1998. http://www.maxwell.vrac.puc-rio.br/Busca_etds.php?strSecao=resultado&nrSeq=8628@1.

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CONSELHO NACIONAL DE DESENVOLVIMENTO CIENTÍFICO E TECNOLÓGICO
De modo geral, as análises de estabilidade em filtragem digital adaptativa referem-se a possíveis instabilidades nos algoritmos de adaptação. Entretanto, a estabilização do algoritmo é apenas parte do problema, visto que também deve ser considerada a estabilização das estruturas recursivas onde operam estes algoritmos. Este trabalho discute a estabilização das estruturas recursivas de filtragem digital utilizadas em filtros IIR adaptativos, apresentando soluções para seu emprego em tempo real. A estabilização da estrutura, somada a do algoritmo, tem o objetivo de estabilizar a operação do filtro como um todo. O estudo inicia pela análise dos principais algoritmos para filtros IIR adaptativos, enfatizando o aspecto estabilidade. Desta forma, pode-se atribuir eventuais instabilidades ocorridas na seqüência do trabalho apenas à estrutura, evitando confundi-las com problemas do algoritmo. Na seqüência, é proposto um novo algoritmo que pretende estabilizar as estruturas recursivas dos filtros IIR digitais através de um modelo com estrutura adaptável. O método é baseado na variação percentual da energia do erro, calculada em tempo real. Finalmente, são apresentados alguns exemplos, indicando a viabilidade do método proposto. Eles ainda mostram benefícios adicionais no seu uso no aspecto velocidade de convergência e redução de eventuais polarizações de parâmetros do processo.
The stability analysis in adaptative digital filters is generally referred to possible instabilities in the adaptation algorthms. However, the algorithm stabilization is only part of the problem, because the stabilization of the recursive structures, where this algorithms operate, should be also considered. This work discusses the stabilization of digital filtering recursive structures that are used in adaptative IIR filters, presenting solutions to its use real time. The stabilization of the structure, added to the stabilization of the algorithm, has the objective of stabilizing the whole filter operation. The study begins with the analysis of the main algorithms for adaptive IIR filters, emphasizing the stability aspect. In this way, instabilities wventually occurred in the sequence of the work could be atributed only to the structure, avoiding any confusion with algorithm´s errors. After that, a new algorithm, that intend to stabilize the recursive structures of the digital IIR filters with a adaptable structure model, is proposed. The method is based on the error energy percentage variation, with real time calculation. Finally, some examples are presented, indicating the viability of the proposed method, showing additional benefits in the convergence speed and reduction of the polarization of the process parameters.
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Che, Ujang Che Ahmad Bukhari Bin. "Quaternion-valued nonlinear adaptive filters." Thesis, Imperial College London, 2012. http://hdl.handle.net/10044/1/9601.

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Advances in vector sensor technology have created a need for adaptive nonlinear signal processing in the quaternion domain. The main concern of this thesis lies in the issue of analyticity of quaternion-valued nonlinear functions. The Cauchy-Riemann-Fueter (CRF) conditions determine the analyticity in the quaternion domain which proved too strict to be of any practical use. In order to circumvent this problem, split-quaternion nonlinear functions which are analytic componentwise are commonly employed. However, these functions do not fully capture the correlations between dimensions and are not suitable for real-world applications. To address this, the use of fully quaternion nonlinear functions in the derivation of a completely new class of algorithms which takes into consideration the non-commutative aspect of quaternion product is proposed. These fully quaternion functions satisfy the local analyticity condition (LAC) that guarantees the first-order differentiability of the function. This provides a unifying framework for the derivation of gradient based learning algorithms in the quaternion domain which are shown to have the same generic form as their real- and complex-valued counterparts. Unlike existing approaches, this new class of algorithms derived is suitable for the processing of signals with strong component correlations and is further extended to the recurrent neural network (RNN) architecture. Novel algorithms are also derived to improve the computational complexity of quaternion-valued adaptive filters which could be easily extended to incorporate nonlinear functions. A rigorous mathematical analysis provides a basis for the understanding of the convergence and steady-state performance of the proposed algorithms. Simulations over a range of synthetic and real-world signals support the approach taken in the thesis.
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8

Druckmüllerová, Hana. "Application of Adaptive Filters in Processing of Solar Corona Images." Doctoral thesis, Vysoké učení technické v Brně. Fakulta strojního inženýrství, 2014. http://www.nusl.cz/ntk/nusl-234232.

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Fotografování sluneční koróny patří mezi nejobtížnější úlohy astrofotografie a zároveň je jednou z klíčových metod pro studium koróny. Tato práce přináší ucelený souhrn metod pro pozorování sluneční koróny pomocí snímků. Práce obsahuje nutnou matematickou teorii, postup pro zpracování snímků a souhrn adaptivních filtrů pro vizualizaci koronálních struktur v digitálních obrazech. Dále přináší návrh nových metod určených především pro obrazy s vyšším obsahem šumu, než je běžné u obrazů bílé koróny pořízených během úplných zatmění Slunce, např. pro obrazy pořízené pomocí úzkopásmových filtrů. Fourier normalizing-radial-graded filter, který byl navržen v rámci této práce, je založen na aproximaci hodnot pixelů a jejich variability pomocí trigonometrických polynomů s využitím dalších vlastností obrazu.
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9

Hevey, Stephen J. "Adaptive Control using IIR Lattice Filters." Thesis, Virginia Tech, 1998. http://hdl.handle.net/10919/36736.

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This work is a study of a hybrid adaptive controller that blends fixed feedback control and adaptive feedback control techniques. This type of adaptive controller removes the requirement that information about the disturbance is known apriori. Additionally, the control structure is implemented in such a way that as long as the adaptive controller is stable during adaptation, the system consisting of the controller and plant remain stable. The objective is to design and implement an adaptive controller that damps the structural vibrations induced in a multi-modal structure. The adaptive controller utilizes an adaptive infinite impulse response lattice filter for improved damping over the fixed feedback controller alone. An adaptive finite impulse response LMS filter is also implemented for comparison of the ability for both algorithms to reject harmonic, narrow bandwidth and wide bandwidth disturbances. It is demonstrated that the lattice filter algorithm performs slightly better than the LMS filter algorithm in all three disturbance cases. The lattice filter also requires less than half the order of the LMS filter to get the same performance.
Master of Science
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10

Salminen, Daniel. "Adaptive filters applied on radar signals." Thesis, Uppsala universitet, Signaler och System, 2013. http://urn.kb.se/resolve?urn=urn:nbn:se:uu:diva-210087.

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This master thesis has been performed at SAAB AB in Järfälla, Sweden.A radar warning receiver must alert the user when someone highlights it with radarsignals. Radar signals used today varies and has a wide frequency band. In order todetect all possible radar signals the radar warning receiver must have a widebandwidth. This results in that the noise power will be high in the radar warningreceiver and weak radar signals will be hard to detect or even undetected.The aim of the thesis work was to investigate the possibility to suppress the noise inthe received radar signals. Unfortunately we do not know the frequency of thereceived radar signals, since the frequency has been decided by the threat radar. Wehave used adaptive filters, which adapts it band-pass to the received radar signal. Theadaptive filters must converge quickly to the state it reduces the noise and passes theradar signals since radar pulses can be very short in the time domain. We also wantto achieve a high SNR gain that is a measurement of how well the adaptive filterreduces the noise.We have investigated two adaptive algorithms, the recursive least square (RLS)algorithm and the least mean square (LMS) algorithm. We found out that the LMSalgorithm was more suitable for noise cancellation in radar applications due to its lowcomplexity and stability compared to RLS algorithm. The LMS algorithm gave SNRgains in the span 14-20 dB for different radar pulses.
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11

Goh, Su Lee. "Complex-valued nonlinear neural adaptive filters." Thesis, Imperial College London, 2006. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.427921.

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Sundaralingam, Sathiaseelan. "Evolving optimal IIR and adaptive filters." Thesis, University of Glasgow, 1999. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.300977.

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13

Gómez, Pablo Emilio Jojoa. "Um algoritmo acelerador de parâmetros." Universidade de São Paulo, 2003. http://www.teses.usp.br/teses/disponiveis/3/3142/tde-17122003-163354/.

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No campo do processamento digital de sinais e em especial da filtragem adaptativa, procura-se continuamente algoritmos que sejam rápidos e simples. Neste contexto, este trabalho apresenta o estudo de novos algoritmos de tempo discreto denominados algoritmos aceleradores (completo, regressivo e progressivo), obtidos a partir da discretização de um algoritmo de tempo contínuo baseado no ajuste da segunda derivada (aceleração) da estimativa dos parâmetros. Destes algoritmos optou-se por estudar mais aprofundadamente os algoritmos aceleradores progressivo e regressivo, devido respectivamente a sua menor complexidade computacional e ao seu desempenho. Para este estudo e análise foram escolhidos como base de comparação os algoritmos LMS e NLMS. Isto porque estes algoritmos estão entre os mais usados e, assim como os algoritmos aceleradores, podem ser obtidos a partir da discretização de algoritmos de tempo contínuo através dos métodos de Euler progressivo e regressivo respectivamente. A análise do algoritmo progressivo mostrou que seu desempenho é inferior ao do algoritmo LMS. Visando diminuir a complexidade computacional do algoritmo acelerador regressivo, foi obtido um novo algoritmo: o versão g. Assim a análise focou-se no algoritmo acelerador regressivo versão g, o qual apresentou um desempenho bom quando comparado no desajuste e no tracking com o algoritmo NLMS, mostrando um melhor compromisso entre velocidade de convergência e variância das estimativas. Este bom desempenho foi comprovado por análises teóricas, por simulações e através da aplicação deste algoritmo na equalização de um canal variante no tempo.
In the digital signal processing field and specially in adaptive filtering, there is a constant search for algorithms both simple and with good performance. This work presents new discrete-time algorithms called accelerating algorithms (APCM and ARg), obtained through the discretization of a continuous-time algorithm that uses the second derivate (acceleration) to adjust the parameter estimates. We provide theoretical analyses for both algorithms, finding expressions for the mean and mean-square errors in the parameter estimates. In addition, we compare the performance of the accelerating algorithms with LMS and NLMS. The analysis of the APCM algorithm showed that its performance is inferior to that of the LMS algorithm. On the other hand, the ARg algorithm presented good performance when compared in terms of misadjustment and tracking with the NLMS algorithm, showing a better compromise between convergence speed and variance of the estimates. This better performance was proven by theoretical analyses, by simulations and through the application of this algorithm to the equalization of a time-variant channel.
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Huang, Yuchen. "Adaptive Notch Filter." PDXScholar, 1994. https://pdxscholar.library.pdx.edu/open_access_etds/4802.

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The thesis presents a new adaptive notch filter (ANF) algorithm that is more accurate and efficient and has a faster convergent rate than previous ANF algorithms. In 1985, Nehorai designed an infinite impulse response (UR) ANF algorithm that has many advantages over previous ANF algorithms. It requires a minimal number of parameters with constrained poles and zeros. It has higher stability and sharper notches than any ANF algorithm until now. Because of the special filter structure and the recursive prediction error (RPE) method, however, the algorithm is very sensitive to the initial estimate of the filter coefficient and its covariance. Furthermore, convergence to the true filter coefficient is not guaranteed since the error-performance surface of the filter has its global minimum lying on a fairly flat region. We propose a new ANF algorithm that overcomes the convergence problem. By choosing a smaller notch bandwidth control parameter that makes the error-performance surface less flat, we can more easily detect a global minimum. We also propose a new convergence criterion to be used with the algorithm and a self-adjustment feature to reset the initial estimate of the filter coefficient and its covariance. This results in guaranteed convergence with more accurate results and more efficient computations than previous ANF algorithms.
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15

Zakaria, Gaguk. "Switching adaptive filter structures for improved performance." Thesis, This resource online, 1993. http://scholar.lib.vt.edu/theses/available/etd-07212009-040356/.

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McWhorter, Francis L. "Novel structures for very fast adaptive filters." Ohio : Ohio University, 1990. http://www.ohiolink.edu/etd/view.cgi?ohiou1173322289.

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17

White, Michael Stephen. "Evolutionary optimization of recursive-lattice adaptive filters." Thesis, Royal Holloway, University of London, 1996. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.338803.

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Lymar, Daria S. "Coupled-magnetic filters with adaptive inductance cancellation." Thesis, Massachusetts Institute of Technology, 2005. http://hdl.handle.net/1721.1/33293.

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Thesis (M. Eng.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 2005.
Includes bibliographical references (p. 95-96).
Conventional filter circuits suffer from a number of limitations, including performance degradation due to capacitor parasitic inductance and the size and cost of magnetic elements. Coupled-magnetic filters have been developed that provide increased filter order with a single magnetic component, but also suffer from parasitic inductance in the filter shunt path due to imperfectly-controlled coupling of the magnetics. This document proposes a new approach to coupled-magnetic filters that overcomes these limitations. Filter sensitivity to variations in coupling is overcome by adaptively tuning the coupling of the magnetic circuit with feedback based on the sensed filter output ripple. This active coupling control enables much greater robustness to manufacturing and environmental variations than is possible in the conventional coupled-magnetic approach, while preserving its advantages. Moreover, the proposed technique also adaptively cancels the deleterious effects of capacitor parasitic inductance, thereby providing much higher filter performance than is achievable in conventional designs. The new technique is experimentally demonstrated in a dc/dc power converter application and is shown to provide high performance.
by Daria S. Lymar.
M.Eng.
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McWhorter, Francis LeRoy. "Novel structures for very fast adaptive filters." Ohio University / OhioLINK, 1990. http://rave.ohiolink.edu/etdc/view?acc_num=ohiou1173322289.

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Song, Sewoong 1954. "Structural sub-band decomposition for adaptive filters." Thesis, The University of Arizona, 1990. http://hdl.handle.net/10150/291831.

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The theory of structural sub-band decomposition of an FIR filters is extended to adaptive filters. It is shown that this sub-band decomposition is equivalent to the transform of input data by orthogonal matrices, of which the Walsh-Hadamard Transform (WHT) is a special case. Thus, the proposed method is a generalization of the transform domain adaptive filtering (TDAF) using WHT, which is already known to enhance the convergence speed of adaptive filters. Furthermore, our method yields one possible hardware implementation of the fast WHT. The convergence behavior of the proposed sub-band adaptive filters is simulated using the Least Mean Squares (LMS) and Recursive Least Squares (RLS) algorithms. The results show the faster convergence speed of the proposed adaptive filters compared to conventional adaptive filters.
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Thomas, James W. "Efficient narrow-band notch filter." Master's thesis, This resource online, 1994. http://scholar.lib.vt.edu/theses/available/etd-02022010-020151/.

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22

Lai, Ching-An. "Global optimization algorithms for adaptive infinite impulse response filters." [Gainesville, Fla.] : University of Florida, 2002. http://purl.fcla.edu/fcla/etd/UFE0000558.

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23

Love, Lonnie J. "Adaptive impedance control." Diss., Georgia Institute of Technology, 1995. http://hdl.handle.net/1853/15911.

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Pasquato, Lorenzo. "Adaptive filtering with balanced model truncation." Thesis, University of Westminster, 2000. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.251702.

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Fischer, Peter Michael. "Adaptive optimization techniques for context-aware information filters." Zürich : ETH, 2006. http://e-collection.ethbib.ethz.ch/show?type=diss&nr=16671.

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Darlington, Paul. "Applications of adaptive filters in active noise control." Thesis, University of Southampton, 1987. https://eprints.soton.ac.uk/52264/.

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The active reduction of acoustic noise is achieved by the addition of a cancelling acoustic signal to the unwanted sound. Successful definition of the cancelling signal amounts to a system identification problem. Recent advances in adaptive signal processing have allowed this problem to be tackled using adaptive filters, which offer significant advantages over conventional solutions. This thesis is devoted to the study of the extension of adaptive noise cancelling techniques, which were developed in the electrical signal conditioning context, to the control of acoustic systems. An analysis is presented of the behaviour of the Widrow-Hoff LMS adaptive noise canceller with a linear filter in its control loop. `Equivalent Transfer Function' descriptors of the system are derived which are linear, time invariant for certain classes of input signal, typical of acoustical applications. This allows the influence of the control loop filter upon the adaptive filter to be analysed using conventional techniques, including the exact specification of the stability limits imposed by the control loop filter. The active control of plane waves propagating axially in a hardwalled duct is used as a motivating model problem. All of the electrical, electroacoustic and acoustic transfer functions in this application may be modelled by a single equivalent control loop filter, allowing the behaviour of the adaptively controlled system to be investigated using the equivalent transfer function approach. The equivalent transfer functions are extended to cover techniques for compensating the adaptive filter for the damaging consequences of the control loop filter. This allows an analytical approach to the design of compensating filters and suggests some formal compensator design rules. The model problem also motivates the study of the effects of feedback around an LMS adaptive filter. Such feedback is shown to be typical of many acoustical applications of the adaptive cancelling technique. The time variance and non-linearity introduced into the system equations, which preclude generalised analysis, are illustrated. An alternative stochastic gradient algorithm for controlling adaptive filters in the presence of feedback is presented. The theoretical work presented in this thesis is supported by experimental results, demonstrating the design and performance of compensated adaptive acoustic noise controllers.
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Zhai, Yiming. "Adaptive log domain filters using floating gate transistors." College Park, Md. : University of Maryland, 2004. http://hdl.handle.net/1903/2132.

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Thesis (M.S.) -- University of Maryland, College Park, 2004.
Thesis research directed by: Dept. of Electrical and Computer Engineering . Title from t.p. of PDF. Includes bibliographical references. Published by UMI Dissertation Services, Ann Arbor, Mich. Also available in paper.
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Fertig, Louis B. "Dual forms for constrained adaptive filtering." Diss., Georgia Institute of Technology, 1998. http://hdl.handle.net/1853/15642.

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Daniel, Timothy Seth. "The effects of precision on the fast, recursive least-squares transversal filters for adaptive filtering." Thesis, This resource online, 1990. http://scholar.lib.vt.edu/theses/available/etd-03242009-040454/.

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Owens, Peter. "Advanced signal processing of high resolution electrocardiograms." Thesis, University of Sussex, 1997. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.361399.

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31

Challa, Deepak Kumar. "Performance improvement of adaptive filters for echo cancellation applications." Diss., Rolla, Mo. : University of Missouri-Rolla, 2007. http://scholarsmine.umr.edu/thesis/pdf/Challa_09007dcc803c4dfe.pdf.

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Thesis (M.S.)--University of Missouri--Rolla, 2007.
Vita. The entire thesis text is included in file. Title from title screen of thesis/dissertation PDF file (viewed December 3, 2007) Includes bibliographical references (p. 65-66).
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Anwar, Fallatah. "Enhancement of Speech Auditory Brainstem Responses Using Adaptive Filters." Thèse, Université d'Ottawa / University of Ottawa, 2012. http://hdl.handle.net/10393/23272.

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Several adaptive filters were investigated to enhance speech auditory brainstem responses (speech ABR). The objective was to shorten the long recording time currently needed by the standard coherent averaging method to obtain acceptable performance, which has limited the clinical adoption of speech ABR. Five algorithms were implemented: Wiener Filter (WF), Steepest Descent (SD), Adaptive Noise Cancellation (ANC) based on Least-Mean-Square error (LMS) and normalized LMS error (nLMS), and a multi-adaptive cascade combination of SD and LMS. The performance of the adaptive filters was assessed on speech ABR data gathered from several subjects and compared with coherent averaging using the overall Signal-to-Noise Ratio (SNR), the local SNR around the fundamental frequency and the first formant, and Mean-Square-Error (MSE) in the time and frequency domains. The adaptive filters could reduce the time needed, by at least one order of magnitude, for obtaining comparable signal quality as that obtained with coherent averaging.
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33

Efe, Murat. "Adaptive approaches to manoeuvering target tracking." Thesis, University of Sussex, 1998. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.263867.

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34

Middleditch, Steven. "Microwave non linear adaptive filters and delay line oscillators." Thesis, University of Leeds, 2007. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.444040.

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35

Nordebo, Sven. "Noise cancelling using spatial filters in broadband adaptive beamformers." Licentiate thesis, Luleå tekniska universitet, 1993. http://urn.kb.se/resolve?urn=urn:nbn:se:ltu:diva-17070.

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36

Oddiraju, Swetha. "Improving performance for adaptive filtering with voice applications." Diss., Columbia, Mo. : University of Missouri-Columbia, 2007. http://hdl.handle.net/10355/6271.

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Thesis (M.S.)--University of Missouri-Columbia, 2007.
The entire dissertation/thesis text is included in the research.pdf file; the official abstract appears in the short.pdf file (which also appears in the research.pdf); a non-technical general description, or public abstract, appears in the public.pdf file. Title from title screen of research.pdf file (viewed on September 29, 2008) Includes bibliographical references.
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37

Zhou, Yi. "Improved analysis and design of efficient adaptive transversal filtering algorithms with particular emphasis on noise, input and channel modeling." Click to view the E-thesis via HKUTO, 2006. http://sunzi.lib.hku.hk/hkuto/record/B36653536.

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Zhou, Yi, and 周翊. "Improved analysis and design of efficient adaptive transversal filtering algorithms with particular emphasis on noise, input andchannel modeling." Thesis, The University of Hong Kong (Pokfulam, Hong Kong), 2006. http://hub.hku.hk/bib/B36653536.

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39

Carvalho, Christopher Alan. "Estimation of a wideband fading HF channel using modified adaptive filters." Ohio : Ohio University, 1993. http://www.ohiolink.edu/etd/view.cgi?ohiou1174936462.

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40

Zhang, Jie. "Blind adaptive cyclic filtering and beamforming algorithms /." *McMaster only, 2001.

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41

Zhao, Yubin [Verfasser]. "Adaptive Particle Filters for Wireless Indoor Target Tracking / Yubin Zhao." Berlin : Freie Universität Berlin, 2014. http://d-nb.info/1062950186/34.

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42

Moat, Trevor P. B. M. "Orthogonal adaptive digital filters with applications to acoustic system identification." Thesis, National Library of Canada = Bibliothèque nationale du Canada, 1998. http://www.collectionscanada.ca/obj/s4/f2/dsk2/tape17/PQDD_0025/MQ27022.pdf.

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43

Ma, Li 1975. "Document processing for adaptive page segmentation using order statistic filters." Thesis, McGill University, 2001. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=31264.

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Page segmentation is one of the important and basic research subjects of document analysis. Traditionally, there are two major kinds of page segmentation approaches. One is the top-down approach and the other is the bottom-up approach. Though these two approaches are been used till now, they are not effective for processing documents with high geometrical complexity and the process of splitting document needs iterative operations which is time consuming. The Modified Fractal Signature (MFS) approach which was presented in recent years can overcome the above weaknesses, however it needs to calculate modified fractal signature which makes the theory and the algorithm very complex. In this thesis, we present two new page segmentation approaches (one is the Maximum Order Statistic Filter (MaxOSF) approach, the other is the Median Order Statistic Filter (MexOSF) approach) based on the order statistic filter (OSF) which is more direct and much simpler. We use the MedOSF to remove the salt-pepper noise of the document and use the MaxOSF to do the page segmentation. In practice, they not only can adaptively process the documents with high geometrical complexity, but also save a lot of computing time.
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Pfann, Eugen. "Design and analysis of oversampled #sigma# #delta# adaptive LMS filters." Thesis, University of Strathclyde, 2003. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.273397.

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Smith, Cameron. "Restoration and registration of digital images using LMS adaptive filters." Thesis, University of the West of Scotland, 1997. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.360228.

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Ferreira, Júnior Ronaldo Sebastião. "Unscented transform performance assessment of adaptive lcmv filters and radioaltimeters." reponame:Repositório Institucional da UnB, 2015. http://dx.doi.org/10.26512/2015.5.D.22025.

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Dissertação (mestrado)—Universidade de Brasília, Faculdade de Tecnologia, Departamento de Engenharia Elétrica, Programa de Pós-graduação em Engenharia Elétrica, 2015.
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Nas últimas décadas técnicas em arranjos de antenas, filtragem adaptativa e processamento de sinais têm recebido grande atenção, por sua versatilidade, possibilidade de aplicação em sistemas embarcados, RADARes, SONARes e afins. Várias técnicas de filtragem adaptativa e beamforming têm sido desenvolvidas desde a década de 1960. Desde então, a complexidade de diversos sistemas e modelos estocásticos vem crescendo de uma forma exponencial, para isto é necessário a adoção de novas técnicas de simulação como a Transformada da Incerteza, de modo que as tradicionais técnicas de simulação como a de Monte Carlo sejam auxiliadas ou substituídas, a fim de obter celeridade no lançamentos de novos produtos e tecnologias no mercado. A filtragem de posto reduzido possibilita a maximização e a otimização da performance da adaptação de filtros adaptativos, além de reduzir a redundância dos sinais recebidos, por meio da redução da dimensão do sinal recebido do arranjo de antenas, com isto, facilita-se o armazenamento de sinais recebidos para pós-processamento e afins. Este trabalho tem como foco a verificação e estudo da performance da Transformada da Incerteza para a simulação de filtros de posto reduzido e radioaltímetros.
In the last decades, antenna arrays techniques, adaptive filtering and signal processing have been in great focus due to its versatility, embedded systems applications, RADARs, SONARs, etc. Various adaptive filtering and beamforming techniques have been developed since the 1960's, and along with those, the performance assessment complexity of stochastic systems simulations has been increasing in a exponential rate. It is mandatory to adopt and develop new simulational techniques, like the Unscented Transform, in order to aid or replace the traditional Monte Carlo simulation, in order to give celerity to the development time of new products and technologies for the market. The reduced rank filtering allows a faster adaptation time for adaptive filters and the elimination of redundant information of an antenna array, optimizing the raw storage for post processing and treatment. The focus of this work is to evaluate the Unscented Transform performance assessment over reduced rank filters and radioaltimeters.
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Holley, Richard D. "Time dependent adaptive filters for interference cancellation in CDMA systems." Thesis, Virginia Tech, 1994. http://hdl.handle.net/10919/40656.

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Interference is a major problem in modern wireless communications systems. No longer are background noise and average power loss the limiting factors in system capacity corruption of the available spectrum by multiple access and nearby interference provides the upper limit to system capacity. If the exponential growth of commercial wireless communications is to continue, systems must effectively deal with the increasingly crowded and corrupted spectrum.

Direct Sequence Spread Spectrum modulation (DS-SS) combined with Time Dependent Processing represents a valid approach to meeting the needs of future communications systems. Traditionally, the exploitation of cyclostationarity in digital communications signals has been reserved for the hostile communication environments faced by the military. However, the advent of cost-effective, high-speed DSP chips and associated processing hardware have made Time Dependent Processing a viable commercial technology.

This thesis presents several forms of the Time Dependent Adaptive Filter (TDAF) which are able to fully exploit the cyclostationarity and high degree of spectral correlation in certain DS-SS signals. It is shown that these optimal TDAFs are able to combat interference from noise, multipath, signals with dissimilar modulation, and signals with similar modulation (multiple access interference). Performance gains are achieved without a knowledge of the specific type of interference and depend solely on the high degree of spectral correlation in DS-SS signals. It is shown that properly designed DS-SS CDMA systems that utilize the TDAF can achieve spectral efficiencies which are within 10% of FDM/TDM systems.

Furthermore, these systems reta~n the benefits of wideband modulation and universal frequency reuse traditionally associated with CDMA systems. The net result is a tremendous increase in system user capacity and signal reception quality.
Master of Science

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48

Eshghi, Mohammad. "A parallel binary structured LMS algorithm for transversal adaptive filters /." The Ohio State University, 1994. http://rave.ohiolink.edu/etdc/view?acc_num=osu1487848891514071.

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49

Clifford, Colin Walter Giles. "Adaptive motion analysis in machine and biological vision." Thesis, University College London (University of London), 1997. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.263269.

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50

Lim, Teng Joon. "System identification using novel adaptive filter structures." Thesis, University of Cambridge, 1995. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.364388.

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