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CUNHA, FERNANDO BRANDAO LOBATO. "ADAPTATION ALGORITHM OF IIR." PONTIFÍCIA UNIVERSIDADE CATÓLICA DO RIO DE JANEIRO, 1991. http://www.maxwell.vrac.puc-rio.br/Busca_etds.php?strSecao=resultado&nrSeq=9838@1.
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CONSELHO NACIONAL DE DESENVOLVIMENTO CIENTÍFICO E TECNOLÓGICOA partir da forma geral do algoritmo de adaptação, é proposto o uso de mais uma variável de projeto, denominada janela. Esta janela tem por objetivo melhorar as características de convergência de algoritmos, cujos parâmetros são partes de estruturas IIR. A introdução das janelas é justificada heuristicamente e seu desempenho é avaliado por meio de diversas simulações de identificação de sistemas. Os resultados obtidos indicam aumentos significativos na velocidade de convergência (cerca de uma ordem de grandeza mais rápido do que os algoritmos atualmente mais usados), na precisão das estimativas dos parâmetros do problema e na robustez dos novos algoritmos (menor número de pólos instáveis durante a adaptação). Estes resultados foram observados em ambientes estacionários e não estacionários, com e sem ruído de medida e com ordem de identificação suficiente ou não.
From the adaptation algorithm general form it is proposed the usage of another design variable, called Window. The goal of this Window is to improve the convergence characteristics of algorithms whose parameters are parts of IIR Structures. The introduction of the Window is heuristically justified and its performance is eventuated by several system identification simulations. The results achieved suggest significant increase in the convergence speed (about one order of magnitude faster than the currently most used algorithms), in the parameter estimation precision and in the new algorithm robusteness (fewer unstable poles during adaptation). These results were observed in sationary and non-stationary environments, with and without measurement noise and with sufficient identification order or not.
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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 TECNOLÓGICODe 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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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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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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Bentmar, Holgersson Sebastian. "Optimising IIR Filters Using ARM NEON." Thesis, Malmö högskola, Fakulteten för teknik och samhälle (TS), 2012. http://urn.kb.se/resolve?urn=urn:nbn:se:mau:diva-20110.
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ARMs processorserie Cortex-A9 har stöd för SIMD-instruktioner med hjälp av NEON MPE. Detta innebär att processorn kan använda sig av vektor-instruktioner som kan utföra operationer på ett flertal element med varje instruktion. Målet med bruk av NEON MPE är att öka prestandan, men då man försöker optimera en speciell IIR-filtertyp som kallas för "biquads" kan man stöta på problem. Problemen med NEON-optimering av "biquads" beror på att endast fem operationer krävs för varje iteration och att behandling av IIR-filter kräver att man behandlar en sampel i taget eftersom varje behandlat sampels värde beror på tidigare behandlade samplar. Rapporten ger en kort beskrivning och genomgång av hur IIR-filter och NEON-optimering fungerar.För att analysera NEON-optimering av biquad-filter skapas fyra olika implementationer av en audioeffekt. De fyra implementationerna jämför prestandan hos flyttalsaritmetik, fixpunkts-aritmetik och NEON-optimering samt en version som implementerar både fixpunktsaritmetik och NEON-optimering. Problemen med optimering av biquad-filter med hjälp av NEON-instruktioner löses genom parallell behandling av ljudkanalerna. Eftersom kanalerna är självständiga kan man fördubbla prestanda genom att utföra varje operation på såväl höger- som vänsterkanal. Vidare prestandaförbättring ges även då effektiviteten hos minnesoperationer förbättras och med hjälp av fixpunkts-behandling.Resultaten visar att fixpunktsversionen som använder sig av NEON-instruktioner är snabbast, men flyttalsversionen med NEON-instruktioner är bara marginellt långsammare och dessutom enklare att implementera. Användandet av NEON-instruktioner förbättrar prestandan med mellan 1,7-2,8 gånger i de fall som testas.The ARM Cortex-A9 CPU has a SIMD extension called NEON MPE. It allows for vector instructions that can perform operations on multiple elements in a single instruction. Whilst this usually improves performance, certain IIR filters called biquads pose problems as only five operations are necessary per sample and every iteration is dependent on the result of the previous result. A brief overview is given for IIR filters, the NEON extension and fixed-point processing.In order to analyse optimisation of biquad filters, an audio effect with four different implementations is produced, comparing results with/without fixed-point processing and with/without NEON optimisation. The problems introduced by the use of biquad filters are solved by running multiple channels in parallel. As the audio channels are independent, two samples can be calculated in parallel, which approximately doubles peformance. Further performance improvement is provided by improved memory operation efficiency and the use of fixed-point processing.The results show that the fixed-point NEON implementation is the fastest, however the floating-point NEON implementation is marginally slower but simpler to write. The use of NEON MPE improves performance by between 1.7 to 2.8 times in this case.
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McNally, Owen C. "VLSI systems for high performance IIR filter." Thesis, Queen's University Belfast, 1991. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.333291.
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Ferro, Humberto Fioravante. "Hybrid convex combinations for IIR system identification." Universidade de São Paulo, 2016. http://www.teses.usp.br/teses/disponiveis/3/3142/tde-04072016-103658/.
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The low complexity of IIR adaptive filters (AFs) is specially appealing to realtime applications but some drawbacks have been preventing their widespread use so far. For gradient based IIR AFs, adverse operational conditions cause convergence problems in system identification scenarios: underdamped and clustered poles, undermodelling or non-white input signals lead to error surfaces where the adaptation nearly stops on large plateaus or get stuck at sub-optimal local minima that can not be identified as such a priori. Furthermore, the non-stationarity in the input regressor brought by the filter recursivity and the approximations made by the update rules of the stochastic gradient algorithms constrain the learning step size to small values, causing slow convergence. In this work, we propose IIR performance enhancement strategies based on hybrid combinations of AFs that achieve higher convergence rates than ordinary IIR AFs while keeping the stability.A baixa complexidade dos filtros adaptativos (FAs) IIR é atrativa para aplicações em tempo real, mas certos inconvenientes têm impedido sua ampla utilização até agora. Para os FAs baseados no gradiente descendente, condições operacionais adversas suscitam problemas de convergência em cenários de identificação de sistemas: pólos subamortecidos ou agrupados, submodelagem ou sinais correlacionados originam superfícies de erro onde a adaptação desacelera em grandes planícies ou para em mínimos locais sub-ótimos que não podem ser identificados como tais a priori. Além disso, a não-estacionaridade do regressor de entrada causada pela recursividade do filtro e as aproximações feitas pelas regras de atualização dos algoritmos de gradiente estocástico restringem o passo de aprendizado a valores pequenos, retardando a convergência. Neste trabalho, propomos estratégias de aprimoramento de desempenho baseadas em combinações híbridas e estáveis de FAs que alcançam taxas de convergências mais altas do que FAs IIR comuns.
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Hall, M. C. "Adaptive IIR filter algorithms for real-time applications." Thesis, University of Liverpool, 1987. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.234800.
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Krukowski, Artur. "Flexible IIR digital filter design and multipath realisation." Thesis, University of Westminster, 1999. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.322993.
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Guttman, Michael. "Sampled-data IIR filtering via time-mode signal processing." Thesis, McGill University, 2010. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=86770.
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In this work, the design of sampled-data infinite impulse response filters based on time-mode signal processing circuits is presented. Time-mode signal processing (TMSP), defined as the processing of sampled analog information using time-difference variables, has become one of the more popular emerging technologies in circuit design. As TMSP is still relatively new, there is still much development needed to extend the technology into a general signal-processing tool. In this work, a set of general building block will be introduced that perform the most basic mathematical operations in the time-mode. By arranging these basic structures, higher-order time-mode systems, specifically, time-mode filters, will be realized. Three second-order time-mode filters (low-pass, band-reject, high-pass) are modeled using MATLAB, and simulated in Spectre to verify the design methodology. Finally, a damped integrator and a second-order low-pass time-mode IIR filter are both implemented using discrete components.Dans ce mémoire, la conception de filtres de données-échantillonnées ayant une réponse impulsionnelle infinie basée sur le traitement de signal en mode temporel est présentée. Le traitement de signal dans le domaine temporel (TSDT), définie comme étant le traitement d'information analogique échantillonnée en utilisant des différences de temps comme variables, est devenu une des techniques émergentes de conception de circuits des plus populaires. Puisque le TSDT est toujours relativement récent, il y a encore beaucoup de développements requis pour étendre cette technologie comme un outil de traitement de signal général. Dans cette recherche, un ensemble de blocs d'assemblage capable de réaliser la plupart des opérations mathématiques dans le domaine temporel sera introduit. En arrangeant ces structures élémentaires, des systèmes en mode temporel d'ordre élevé, plus spécifiquement des filtres en mode temporel, seront réalisés. Trois filtres de deuxième ordre dans le domaine temporel (passe-bas, passe-bande et passe-haut) sont modélisés sur MATLAB et simulé sur Spectre afin de vérifier la méthodologie de conception. Finalement, un intégrateur amorti et un filtre passe-bas IIR de deuxième ordre en mode temporel sont implémentés avec des composantes discrètes.
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Filho, Thomas Edson Filgueiras. "Sobre a velocidade de convergência da filtragem adaptativa IIR." Universidade de São Paulo, 2008. http://www.teses.usp.br/teses/disponiveis/3/3142/tde-09022009-181757/.
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Filtros adaptativos com resposta ao impulso infinita (IIR) podem substituir com vantagens aqueles com respostas ao impulso finitas (FIR). Entre estas vantagens está o seu reduzido número de parâmetros que leva a uma menor complexidade computacional na obtenção de respostas similares. Porém, a utilização de filtros adaptativos IIR apresenta alguns problemas práticos, sendo o mais destacado sua convergência lenta. Este problema aparece principalmente quando algoritmos baseados no gradiente são utilizados para a adaptação dos coeficientes do filtro. A abordagem baseada na teoria de realização balanceada de sistemas, previamente utilizada para se analizar filtros com entrada branca, se mostrou uma ferramenta útil para entender o que faz com que um filtro convirja lentamente. Este método já foi aplicado com sucesso na análise de filtros adaptativos IIR com entrada branca nas configurações de identificação e de identificação inversa. Neste trabalho aplicaremos este mesmo método para o caso de entrada não branca. Será mostrado que a configuração de identificação inversa é um caso particular da configuração de identificação com entrada não-branca, podendo ambas serem tratadas conjuntamente. Também será mostrado que o sistema que controla as propriedades de convergência não é mais o sistema desconhecido que se está tentando identificar, e sim um sistema relacionado a este e a densidade espectral da entrada. No caso de entrada branca, esta análise levou ao algoritmo de aproximações sucessivas, o qual, utilizando um bloco auxiliar, tenta fazer com que o filtro adaptativo enxergue um um sistema de rápida convergência. Será apresentada uma generalização deste algoritmo para o caso de entrada não-branca, inclusive serão apontadas limitações do mesmo quanto a valores dos passos de adaptação. Simulações numéricas serão usadas para ilustrar todos os resultados obtidos.Adaptive filters with infinite impulse response (IIR) can replace with advantages the ones with finite impulse response (FIR). One of these advantages is the reduced number of parameters which leads to a smaller computational complexity giving similar responses. However, the use of adaptive IIR filters has some pratical issue, being the most prominent its slow convergence. This issue is mainly seem when gradient descent algorithms are applied to adptate the filters coefficients. The approach bassed on the balanced realization of systems, previously used to analyze the convergence speed of adaptive IIR filter with white input, has seemed to be a useful tool in understanding what causes the slow convergence in a filter. This approach was sucessful aplliedto the analyzes of the identification and inverse identification configurations. In this work we will aplly this same approach to the non-white input case. It will be shown that the invese identification configuration is a special case of the identification configuration with non-white input, so both can be addressed together. It will also be shown that the convergence properties are no more set by the caracteristics of the unknown system, but by the caracteristics of a system related to it and the input spectral density function. In the white input case, the results of this analysis were used to propose the sucessive approximations algorithm, which uses an auxiliary block trying to make the adaptive filter sees a system with faster convergence. A more general form of this algorithm that includes the non-white input case will be presented, and some drawbacks regarding the adptation stepsize will be pointed out. Numerical simulations will be used to illustrate all the obtained results.
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Burt, Phillip Mark Seymour. "Um estudo sobre filtros IIR adaptativos com aplicação a uma estrutura polifásica." Universidade de São Paulo, 1997. http://www.teses.usp.br/teses/disponiveis/3/3142/tde-14122017-134500/.
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Neste trabalho faz-se um estudo sobre filtros IIR adaptativos e é apresentada uma estrutura polifásica para filtragem IIR adaptativa, que, em troca de um aumento de complexidade computacional, pode apresentar características mais favoráveis do que a estrutura direta comumente usada. O aumento da complexidade computacional, relativamente a um algoritmo do tipo newton, por exemplo, é pequeno. Apresenta-se uma análise dos efeitos da proximidade ao círculo unitário dos pólos do sistema sendo modelado. Um dos efeitos considerados é o comportamento limite do condicionamento da matriz de estados associada ao algoritmo de adaptação. São considerados algoritmos de adaptação de passo constante de uso comum para filtros IIR adaptativos. O método utilizado é particularmente útil para a verificação do efeito da posição dos pólos do sistema sendo modelado e também para a introdução de certas restrições ao mesmo, como, por exemplo, norma L2 fixa e resposta em freqüência passa-tudo. Um resultado interessante é que a única situação, entre as testadas, em que o condicionamento da matriz mencionada não tende a infinito quando um número qualquer de polosndo sistema sendo modelado H(z) se aproxima da circunferência unitária, é quando H(z) é passa-tudo e emprega-se o algoritmo PLR. São analisadas também a superfície de erro e a superfície de erro reduzida para filtros IIR adaptativos. Mostra-se que, quando o sistema sendo modelado possui polos próximos à circunferência unitária, a superfície de erro reduzida apresenta regiões planas com erro quadrático médio elevado. A existência destas regiões resulta em uma baixa velocidade de convergência global de algoritmos de passo constante. A partir da decomposição em valores singulares (SVD) da forma de Hankel do sistema sendo modelado, é apresentada também uma decomposição da superfície de erro reduzida, a partir da qual pode-se obter uma separaçãoparcial dos efeitos do sistema sendo modelado e da forma de realização do filtro adaptativo. Uma estrutura polifásica para filtragem IIR adaptativa é apresentada e seu desempenho é comparado com o de filtros IIR adaptativos na forma direta. Mostra-se o possível ganho da estrutura polifásica quanto à velocidade de convergência local e quanto às características da superfície de erro reduzida e à velocidade de convergência global. Demonstra-se, para a estrutura polifásica, que, com entrada branca e modelamento suficiente, todos os pontos estacionários da superfície de erro são mínimos globais da mesma. Este resultado não decorre diretamente de propriedades análogas relativas à estrutura direta, já conhecidas. Tudo para a estrutura direta quanto para a estrutura polifásica, são apresentados os resultados de várias simulações dos algoritmos de adaptação considerados.A study on IRR adaptive filters and polyphase structure for IIR adaptive filtering are presented. In exchange for an increase in computational complexity, which is small if compared to Newton algorithms, the polyphaser structure may exhibit a better performance than direct structures. An analysis of the effects of the proximity to the unit circle of the modelled system\'s poles is presented. One of the considered points is the limiting behavior of the condition of the state matrix related to the adaptive algorithm. Commonly used constant gain algorithms are considered. The method of analysis is specially usefull for verifying the effects of the position of the system\'s poles and also for introducing certain restrictions to the system, as fixed L2 norm and all-pass frequency response. An interesting result is that, among the situations that were tested, the only one in which the condition of the aforementioned matrix does not tend to infinity as the poles of the modelled system H(z) tend to the unit circle is when H)z) is all-pass and the PLR algorithm is employed. The error surface and the reduced error surface for IIR adaptive filters are also analyzed. It is shown that the modelled system has poles close to the unit circle the reduced error surface presents flat regions with high mean square error. The presence of these flat regions results in low global convergence speed for constant gain adaptive algorithms. Based on the singular value decomposition (SVD) of the modelled system\'s Hankel form, a decomposition of the reduced error surface is also presented. In it there exists a partial separation of the effects of the system and the adaptive filter\'s structure. A polyphaser structure for IIR adaptive filtering is presented and its performance is compared to the performance of the direct structure. The gain in local convergence and global convergencespeed, as well as the better behavior of the reduced error surface which may be attained , are shown. It is demonstrated, for the polyphaser structure, that, with while input and sufficient modelling, all the stationary points of the error surface are global minima. This result does not follow directly from similar well known results for the direct structure. Simulation results for the considered algorithms are also presented.
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Antonakoudi, Anixi. "Interpolation techniques for unsupervised deconvolution of FIR and IIR systems." Thesis, Imperial College London, 2002. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.271616.
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Marcelino, Márcio Abud. "Identificação de sistemas variantes no tempo com filtragem adaptativa IIR." Instituto Tecnológico de Aeronáutica, 1994. http://www.bd.bibl.ita.br/tde_busca/arquivo.php?codArquivo=1723.
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Neste trabalho apresenta-se uma estrutura e o respectivo algoritmo de filtragem adaptativa para identificação de sistemas variantes no tempo, usando modelamento com resposta infinita ao impulso. Esta estrutura tem as características de não polarizar a estimativa na presença de ruído aditivo branco e possuir uma função de erro de estimação linear em relação aos coeficientes estimados, gerando o algoritmo do USARF - "Unbiased Single Adaptive Regressive Filter". Apresenta-se também um estudo das principais técnicas de filtragem adaptativa IIR cujos algoritmos mantem-se ativos e com cargas computacionais reduzidas. Analisa-se as características de estabilidade e convergências e apresenta-se os principais testes de estabilidade. Algumas conclusões são obtidas por simulações e os resultados relevantes são apresentados. Entre eles, destacam-se, o desempenho da estrutura para diferentes ordens do filtro em relação a ordem da planta, e a superioridade em relação aos demais algoritmos de filtragem adaptativa IIR, principalmente em relação as condições iniciais e a velocidade de convergência. Aproximações desta estrutura são apresentadas, objetivando aplicações especificas. E ressaltado que aproximações podem gerar não linearidade, aumentando a dependência das condições iniciais e reduzindo a velocidade de convergência.
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Oliveira, Daniel Rossato de. "Projeto de filtros digitais IIR com técnicas de computação evolucionária." Universidade Tecnológica Federal do Paraná, 2012. http://repositorio.utfpr.edu.br/jspui/handle/1/211.
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O projeto de filtros digitais do tipo IIR é um problema clássico da engenharia. Filtros digitais possuem diversas aplicações, e muitas variantes de comportamento. Existem vários métodos especialistas consagrados na literatura, cada um com suas peculiaridades e adequações a diferentes casos. Porém, a maior parte destes métodos não é flexível, impedindo a especificação de todos os parâmetros importantes de um filtro, pois alguns desses são um sub-produto do processo de cálculo. Por isso, este problema também é atacado na literatura através de técnicas de computação evolucionária. Diversos estudos foram encontrados com esta abordagem, mas em quase todos, o enfoque era dado no algoritmo evolucionário em vez de no filtro, utilizando este apenas como uma função de \emph{benchmark}. Além disso, a estabilidade do filtro - condição imprescindível para a utilização do mesmo - é desprezada em quase todos os trabalhos. Portanto, este trabalho propõe uma função de \emph{fitness} e uma nova codificação para este problema, de forma a possibilitar a obtenção de bons filtros, dentro das especificações, com algoritmos de Computação Evolucionária na forma canônica, isto é, sem modificações estruturais. A função de \emph{fitness} proposta busca corrigir distorções causadas pela função tradicional, que não leva em conta a obediência às especificações do filtro. A codificação mapeia o espaço de busca apenas para as soluções estáveis, sem excluir nenhuma solução válida nesta transformação. Além disso, um pós-processamento permite equalizar a resposta em fase do filtro, isto é, tornar o atraso de fase na banda de passagem linear, condição necessária para a utilização em diversos sistemas, especialmente os de telecomunicações. O desempenho das modificações é comparado com as abordagens clássicas utilizadas na literatura, e o conjunto escolhido como o mais eficiente é utilizado para comparar os dois algoritmos mais utilizados em Computação Evolucionária, o PSO e o AG. Após esse passo, experimentos extensivos de ajuste de parâmetros foram realizados, para que a versão final fosse comparada com o método especialista mais poderoso, que é o cálculo de filtros elípticos. Os resultados mostraram que o conjunto de modificações proposto fez com que excelentes filtros fossem obtidos, com uma taxa de obediência às especificações muito superior à obtida sem o mesmo. Comparando com o método especialista, o desempenho foi semelhante, com pontos a favor e contra cada um, mostrando que o projeto de filtros IIR através de Computação Evolucionária pode ser utilizado em sistemas reais. Em trabalhos futuros poderão ser estudadas novas modificações na funçao de \emph{fitness}, além do desempenho obtido com outros algoritmos evolucionários. A utilização em sistemas \emph{online} é uma aplicação promissora, e o comportamento deste método com especificações não-estacionárias, oriundas de informações de estimação de canal também deve ser investigado.IIR digital filter design is classic problem in Electrical Engineering. Digital filter have many applications, and dozens of different behaviors. There are many specialist methods in literature, each with specific differences and characteristics. But most of these methods are not flexible, preventing one from fixing all parameters, because some of them are a sub-product of the calculation process. Therefore, this problem is also solved in literature with Evolutionary Computing techniques. Several works were found with this approach, but in almost all, the focus was in the evolutionary algorithm rather than on the filter problem, using this only as a benchmark function. Moreover, the filter stability - prerequisite for the filter application - is neglected in almost all of them. So, this work proposes a new fitness function, as well as a new codification, in order to obtain good filters, obedient to the specifications, with Evolutionary Computing algorithms in the canonic form, this is, without structural modifications. The proposed fitness function tries to correct distortions caused by the traditional one, which does not takes into account the filter gain specifications. The new coding maps the search space only to the stable solutions, not excluding any valid solution in the transformation. Moreover, a post-processing allows the filter phase equalization, this is, to make the pass band phase delay linear. This property is necessary in many types of applications, specially the telecommunication ones. The modifications performance is compared to the classic approaches found in the literature, and the most efficient set is use to compare the two most used Evolutionary Computing algorithms, GA and PSO. After this, extensive parameter tuning experiments were made, producing a final version of the method to compare with the specialist one. The chosen specialist method was the Elliptic filter, because the filter response has similarities with the one found by the proposed algorithm, and also because it is the most powerful one. Results showed excellent filters found with the proposed modifications, with a specification obedience rate well above the ones obtained with the classical approaches. Comparing with the specialist method, a similar performance was achieved, showing that IIR filter design with Evolutionary Computing can be used in real systems. Future work will address new modifications in the fitness function, and the performance of different EC algorithms in this problem. The application in online systems is promising, so the behavior of this approach with non-stationary specifications coming from channel estimation techniques should also be investigated.
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Williams, Sierra S. "Investigation of roundoff noise in IIR digital filters using MATLAB." Connect to this title online, 2009. http://etd.lib.clemson.edu/documents/1249065585/.
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Ameri, Ali. "Time-mode reconstruction IIR filters for sigma-delta phase modulation applications." Thesis, McGill University, 2011. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=104809.
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The design of several low-pass IIR time-mode filters for use as reconstruction filters in digital-to-time conversion (DTC) applications is proposed. Previously, such reconstruction filters were implemented using phase-locked loops. The proposed filters are constructed from a simple digital-like structure involving voltage-controlled delay units. The resulting circuits require very small silicon area and consume very little power. A first-order filter design for wideband reconstruction applications was fabricated in a 0.13 um CMOS process occupying a silicon area of 170 um x 100 um and consumes 670 uW. The results prove for the first time that the concept of time-mode filtering is feasible in a CMOS monolithic process. Another design, intended for narrowband sigma-delta phase signal generation applications, is proposed that utilizes similar building blocks but uses a filter topology that is better suited for implementations with transfer functions having low-frequency poles. High-order realizations can be constructed as a cascade of several first-order sections. Such an approach will be demonstrated in the design of a sigma-delta phase-encoding signal-generation scheme.Dans cette dissertation nous proposons plusieurs filtres IIF passe-bas qui opèrent en mode temps. Ces dispositifs sont conçus pour être utilisé comme filtres de reconstruction dans les convertisseurs numérique-temps (CNT). Dans le passé, de tels filtres ont été implémenté à partir de boucles à verrouillage de phase. Les filtres proposés dans cette thèse sont construits à partir d'une simple structure numérique impliquant des unités de retards commandés en tension. Les circuits résultant de cette approche requièrent de petites surfaces sur silicium et consomment très peu d'énergie. Un filtre du premier ordre pour les applications larges bandes a été fabriqué dans un processus CMOS 0.13 um. Le filtre occupe une surface de silicium de 170 um x 100 um et consomme 670 uW. Les résultats montrent pour la première fois que la notion de filtrage en mode temps est possible dans un processus CMOS monolithique. Un autre filtre destiné à des applications de génération de signal de phase sigma-delta à bande étroite est aussi proposé. Ce filtre utilise des blocs de construction similaire au premier mais utilise une topologie qui est mieux adapté pour les implémentations de fonctions de transfert ayant des pôles à forte valeur de Q. Les filtre d'ordre supérieur peuvent être construits en cascadant plusieurs filtres du premier ordre. Une telle approche sera démontrée par la conception d'un système de génération de signaux de phase codes en sigma-delta.
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Sezen, Umut. "Anticausal inverses for IIR filter banks, quincunx wavelets and image coding." Thesis, University of Warwick, 2003. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.397219.
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Nicolson, Laurence John. "The design of IIR digital filters to magnitude and delay specifications." Thesis, University of Liverpool, 1993. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.320608.
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Craney, Edward Peter. "Adaptive IIR filtering for acoustic echo cancellation on a mobile handset." Thesis, University of Southampton, 2003. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.402220.
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Ma, Qiang. "The application of genetic algorithms to the adaptation of IIR filters." Thesis, Loughborough University, 1995. https://dspace.lboro.ac.uk/2134/32269.
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The adaptation of an IIR filter is a very difficult problem due to its non-quadratic performance surface and potential instability. Conventional adaptive IIR algorithms suffer from potential instability problems and a high cost for stability monitoring. Therefore, there is much interest in adaptive IIR filters based on alternative algorithms. Genetic algorithms are a family of search algorithms based on natural selection and genetics. They have been successfully used in many different areas. Genetic algorithms applied to the adaptation of IIR filtering problems are studied in this thesis, and show that the genetic algorithm approach has a number of advantages over conventional gradient algorithms, particularly, for the adaptation of high order adaptive IIR filters, IIR filters with poles close to the unit circle and IIR filters with multi-modal error surfaces. The conventional gradient algorithms have difficulty solving these problems. Coefficient results are presented for various orders of IIR filters in this thesis. In the computer simulations presented in this thesis, the direct, cascade, parallel and lattice form IIR filter structures have been used and compared. The lattice form IIR filter structure shows its superiority over the cascade and parallel form IIR filter structures in terms of its mean square error convergence performance.
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Sun, Hanwu. "Adaptive techniques for multiband parameter estimation and extraction." Thesis, University of Westminster, 1996. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.296126.
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Kaake, Fadi M. "A VLSI-nMOS hardware implementation of an IIR bandpass orthogonal digital filter." Ohio : Ohio University, 1986. http://www.ohiolink.edu/etd/view.cgi?ohiou1183133214.
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Yin, Shishu, and 殷仕淑. "New design and factorization methods for perfect reconstruction causalstable IIR filter banks." Thesis, The University of Hong Kong (Pokfulam, Hong Kong), 2006. http://hub.hku.hk/bib/B38674592.
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Yin, Shishu. "New design and factorization methods for perfect reconstruction causal stable IIR filter banks." Click to view the E-thesis via HKUTO, 2006. http://sunzi.lib.hku.hk/hkuto/record/B38674592.
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Ruiz, Fontes Natanael. "An analysis of the IIR an FIR Wiener filters with applications to underwater acoustics." Monterey, Calif. : Springfield, Va. : Naval Postgraduate School ; Available from National Technical Information Service, 1997. http://handle.dtic.mil/100.2/ADA333441.
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Thesis (M.S. in Engineering Acoustics and M.S. in Electrical Engineering) Naval Postgraduate School, June 1997.Thesis advisor, Charles W. Therrien. Includes bibliographical references (p. 75). Also available online.
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Arumemi-Ikhide, Martin Ikpehai Idemudia. "Empirical design of an evolutionary algorithm for optimisation of 1-D digital IIR filters." Thesis, University of Warwick, 2007. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.444848.
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Chen, Junyi. "Oligo(ß-Alanine)-Grafted Butyl Rubber (IIR) with Mixed O(ligo(ß-Alanine) Lengths." University of Akron / OhioLINK, 2015. http://rave.ohiolink.edu/etdc/view?acc_num=akron1427827046.
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Sweeney, Paul. "THE NEXT GENERATION AIRBORNE DATA ACQUISITION SYSTEMS. PART 1 - ANTI-ALIASING FILTERS: CHOICES AND SOME LESSONS LEARNED." International Foundation for Telemetering, 2003. http://hdl.handle.net/10150/605378.
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International Telemetering Conference Proceedings / October 20-23, 2003 / Riviera Hotel and Convention Center, Las Vegas, NevadaThe drive towards higher accuracy and sampling rates has raised the bar for modern FTI signal conditioning. This paper focuses on the issue of anti-alias filtering. Today's 16-bit (and greater resolution) ADC’s, coupled with the drive for optimum sampling rates, means that filters have to be more accurate and yet more flexible than ever before. However, in order to take full advantage of these advances, it is important to understand the trade-offs involved and to correctly specify the system filtering requirements. Trade-offs focus on: • Analog vs. Digital signal conditioning • FIR vs. IIR Digital Filters • Signal bandwidth vs. Sampling rate • Coherency issues such as filter phase distortion vs. delay This paper will discuss each of these aspects. In particular, it will focus on some of the advantages of digital filtering various analog filter techniques. This paper will also look at some ideas for specifying filter cut-off and characteristics.
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Tesařík, Jan. "Prostředí pro verifikaci digitálních filtrů." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2016. http://www.nusl.cz/ntk/nusl-242188.
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Diploma thesis deals with design of verification environment for analyzing systems with digital filters. Verification environment is written in SystemVerilog language and it is generated by program, which is also providing generation of input data for system of filters. Matlab environment is used for gaining the reference data. The simulation of the designed involvement with digital filters is performed by program ModelSim. The most watched parameter is functional coverage which indicates how big part of the HDL description has been tested.
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Cheng, Hai-Ling Margaret. "3D spatio-temporal interpolation of of digital image sequences using low-order 3D IIR filters." Thesis, National Library of Canada = Bibliothèque nationale du Canada, 1997. http://www.collectionscanada.ca/obj/s4/f2/dsk2/ftp04/mq20866.pdf.
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Joshi, Rimesh M. "Analog and Digital Array Processor Realization of a 2D IIR Beam Filter for Wireless Applications." University of Akron / OhioLINK, 2012. http://rave.ohiolink.edu/etdc/view?acc_num=akron1326776749.
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Doğanata, Zinnur Vaidyanathan P. P. "General structural representations for multi-input multi-output discrete-time FIR and IIR lossless systems /." Diss., Pasadena, Calif. : California Institute of Technology, 1990. http://resolver.caltech.edu/CaltechETD:etd-02222007-083438.
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Galabada, Kankanamge Nilan Udayanga. "Digital 2-D/3-D Beam Filters For Adaptive Applebaum ReceiveAnd Transmit Arrays." University of Akron / OhioLINK, 2015. http://rave.ohiolink.edu/etdc/view?acc_num=akron1445799928.
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Grossman, Hy, and Steve Pellarin. "A Time Correlated Approach to Adaptable Digital Filtering." International Foundation for Telemetering, 2006. http://hdl.handle.net/10150/604156.
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ITC/USA 2006 Conference Proceedings / The Forty-Second Annual International Telemetering Conference and Technical Exhibition / October 23-26, 2006 / Town and Country Resort & Convention Center, San Diego, CaliforniaSignal conditioning is a critical element in all data telemetry systems. Data from all sensors must be band limited prior to digitization and transmission to prevent the potentially disastrous effects of aliasing. While the 6th order analog low-pass Butterworth filter has long been the de facto standard for data channel filtering, advances in digital signal processing techniques now provide a potentially better alternative. This paper describes the challenges in developing a flexible approach to adaptable data channel filtering using DSP techniques. Factors such as anti-alias filter requirements, time correlated sampling, decimation and filter delays will be discussed. Also discussed will be the implementation and relative merits and drawbacks of various symmetrical FIR and IIR filters. The discussion will be presented from an intuitive and practical perspective as much as possible.
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Kondapalli, Soumya. "Digital Systolic Architectures for UWB Beamforming using 2-D IIR Spatio-temporal Frequency-planar Beam Filters." University of Akron / OhioLINK, 2013. http://rave.ohiolink.edu/etdc/view?acc_num=akron1376925582.
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蕭智中. "Adaptive IIR Filter." Thesis, 1997. http://ndltd.ncl.edu.tw/handle/18044663472704974063.
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碩士中原大學
電子工程學系
85
In the areas of radar, communication and biomedical engineering, it is desired to extract or eliminate a narrowband sinusoidal signal in broadband noise. This can be done by an adaptice line enhance (ALE) or an adaptive notch filter (ANF), respectively [1][2][5][9][10]. Generally speaking, the adaptive IIR filter is more computationally. efficient and better statistival performance than the adaptive FIR filter for the noisy sinusoidal signal tracking .But most higher order adaptive IIR filter have some drawbacks, such as : stability problem, convergence to local minimum and difficult analysis of the performance, ... In order to avoid these problems, the excellent constrained adaptive IIR filter have been given [2]-[4]. The proposed first order adaptive IIR filter can overcome these problems. The enhancement and tracking of real or complex sinusoidal signals in broadband noise has received considerable interest in recent years. Here, a complex sinusoidal signal will be used as the input of the adaptive filter, it can simplify the algebraic derivations [13] . The angular frequency of the system pole location, not it's magnitude, is used as the adaptaiton parameter. In order to keep the filter as a stable system, the pole location will be controlled inside the unit circle. It mean that the system is always stable and doesn't require a stability monitor. The convergence, and misadjustment analysis of the adaptive IIR filter are obtained. The computer simulation verifies an excellent performance of sinusoidal tracking and shows the superior performance for the detection of closely-spaced sinusoids.
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CHEN, KE-REN, and 陳克任. "IIR adaptive filter." Thesis, 1992. http://ndltd.ncl.edu.tw/handle/99928635555067664814.
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Chen, Hsu Chia, and 許家禎. "Design of IIR eigenfilters." Thesis, 2000. http://ndltd.ncl.edu.tw/handle/36236421836601528167.
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碩士國立臺灣大學
電信工程學研究所
88
The eigenfilter approach is widely used to design kinds of digital filters recently. This method is based on the eigenvector computation of an appropriate symmetric, positive-definite matrix, and the solution can be found efficiently. There are two major kinds of implementation of digital filters: IIR (Infinite Impulse Response) and FIR (Finite Impulse Response). The eigenfilter approach has been used to effectively design linear phase FIR filters, FIR Hilbert transformers, and digital differentiators, and complex FIR filters. With the existence of denominator, design of IIR digital filters is somewhat different from that of FIR filters. This dissertation is focus on digital IIR filter design based on eigenfilter approach. The effectiveness of this approach of adding time and frequency domain constraints is presented. In chapter 2 to 4, we introduce this approach and extend it to design filters with time and frequency domain constraints such as notch filters, Nyquist filters, and partial response filters, or even the filters with linear constraints. We formulate the transfer function according to the constraints and implement the formulation with eigenfilter approach. In chapter 5 and 6, we formulate IIR complex filters and IIR log filters and apply eigenfilter approach to implement the filters. We will also present a new method to construct FIR filter with sub-filters optimally in chapter 7 and conclude this dissertation and suggest our future works in chapter 8.
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Niu, I.-Ching, and 牛挹青. "Optimal Design of 1-D IIR Filter Banks and 2-D IIR Filters." Thesis, 1999. http://ndltd.ncl.edu.tw/handle/83124583717594043411.
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碩士國立臺灣大學
電信工程學研究所
87
Owing to the increasing use of digital signal processing, the design of digital filters and digital filter banks has become more and more important.In this thesis, it should be noted that a systematic algorithm is proposed as the basis to design the IIR filters and filter banks. However, this basic algorithm, in the various designing structure, is sometimes aided by WLS and Karmarkar''s algorithms to achieve a good design in certain optimal senses. To control the stability of the IIR filter, we replace the denominator of filter with lattice structure. That is, the reflection coefficients have been considered as the coefficients in the design. We therefore can directly control the stability of the filters by controlling the magnitude of these coefficients. The structure has been used in this thesis to design 1-D filter banks and 2-D filters. The satisfactory results as shown in the design examples given in this thesis have therefore successfully proven that the proposed design methods to be sound and viable.
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Lee, Hsiao-Fu, and 李孝甫. "IIR Lattice-based Blind Equalization Algorithms." Thesis, 2012. http://ndltd.ncl.edu.tw/handle/87448900891724583497.
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碩士輔仁大學
電機工程學系
100
This thesis develops infinite impulse response (IIR) lattice structures based on two widely known blind equalization algorithms, which are the multi-modulus algorithm (MMA) and the constant modulus algorithm (CMA). The motivation for doing so is that generally for algorithms to achieve the same performance, an IIR filter requires fewer coefficients than its corresponding finite impulse response (FIR) counterpart. Moreover, the monitoring of the stability of an IIR lattice filter is simpler and less computationally complicated than that of the direct-form IIR filter. In computer simulations, the IIR lattice-based blind equalization algorithm was compared with the corresponding FIR transversal algorithms and FIR-based lattice algorithms. Simulation results demonstrate that the IIR lattice-based blind equalization algorithm is especially suitable for equalizing channels yielding inter-symbol interference (ISI) that mostly results from the post-cursor term.
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Ocloo, Senanu K. "Global optimization methods for adaptive IIR filters." 2007. http://www.lib.ncsu.edu/theses/available/etd-07112007-150000/unrestricted/etd.pdf.
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Chen, Ying-Shen, and 陳映先. "Adaptive IIR Filter for Active Noise Control." Thesis, 2006. http://ndltd.ncl.edu.tw/handle/94603722740050144759.
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碩士國立中興大學
機械工程學系所
94
The aim of this thesis is to examine applications of adaptive IIR filters with a FuLMS algorithm for active noise control (ANC). Due to existence of a secondary path in ANC systems, a small value of step size is applied to FuLMS algorithm and generally leads to slow convergence. To improve this difficulty, two approaches have been taken into account. One approach applies a modified residual error for the FuLMS algorithm. Another approach applies a robustly controlled secondary path for the adaptive IIR filter. Results of computer simulation show that adaptive IIR filters in our proposed approaches allow larger value of step size and lead to better convergence property as compared with that of the existing approaches. Results of experiment support the simulation results and demonstrate effective noise reduction performance.
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Sha, Chun-Shen, and 謝忠賢. "Design of IIR Filter via Genetic Algorithm." Thesis, 1994. http://ndltd.ncl.edu.tw/handle/95820337348167246766.
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碩士國立清華大學
電機工程研究所
82
A new method is proposed for IIR digital filter design with desired magnitude and phase response simultaneously. L-1 norm, L-2 norm, L-infinity norm criterion are all considered for the optimal IIR filter approximation. The techniques of genetic algorithm is employed to directly solve these optimization design problem at the same time. Genetic algorithms are optimization and machine learning algorithms, initially inspired from the process of natural selection and evolutionary genetics. Several design examples are given to illustrate the design procedure and the performance of the proposed design method.
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SHIUE, JIAN CHIOU, and 薛建秋. "Adaptive IIR filter for multiple sinusoids detection." Thesis, 1993. http://ndltd.ncl.edu.tw/handle/65052810521066559337.
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碩士中原大學
電子工程學系
81
This paper proposes a new constrainted adaptive infinite im- pulse response (IIR) filter in order to detect multiple sinusoids signals. Conventional constrained adaptive IIR filter depends upon the converge rate and the initial value of the filter weights, and it may fail to detect the signals when the frequency of the input signal is very low or very high. The improved struc- ture suggested in this paper can avoid these problems and the converge rate of the system is much improved. Apply this improved structure in detection for multiple sinu- soids, we obtain better detecting effects. This paper discuss in details the cases of detecting multiple sinusoids in time domain and its subband, and analyze the conditions the weight co- efficients converge.
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"Simulación de Filtos FIR e IIR en VHDL." Tesis, Universidad de las Américas Puebla, 2005. http://catarina.udlap.mx/u_dl_a/tales/documentos/lep/vargas_f_cd/.
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"Efficient implementations of 2-D noncausal IIR filters." Massachusetts Institute of Technology, Laboratory for Information and Decision Systems], 1995. http://hdl.handle.net/1721.1/3405.
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Michael M. Daniel, Alan S. Willsky.Includes bibliographical references (p. 34-35).
Supported by the Office of Naval Research. N00014-91-J-1004 Supported by the Air Force Office of Scientific Research. F49620-95-1-0083 Supported by the Army Research Office. DAAL03-92-G-0115
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Badyal, Rajeev. "VLSI implementation of adaptive BIT/serial IIR filters." Thesis, 1992. http://hdl.handle.net/1957/36521.
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A new structure for the implementation of bit/serial adaptive IIR filter is
presented. The bit level system consists of gated full adders for the arithmetic
unit and data latches for the data path. This approach allows recursive
operation of the IIR filter to be implemented without any global
interconnections, minimal delay time, chip area and I/O pins. The
coefficients of the filter can be updated serially in real time for time invariant
and adaptive filtering. A fourth order bit/serial IIR filter is implemented on a
2 micron CMOS technology clocked at 55 MHz.Graduation date: 1992
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Jin, J. C., and 鄭張權. "Applications of IIR Fiters to Acoustic Echo Cancellation." Thesis, 1995. http://ndltd.ncl.edu.tw/handle/93589497488326215388.
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碩士國立交通大學
電信研究所
83
AEC (Accoustic Echo Cancellation) is usually implemented by FIR filters.However, such filters require up to thousands of orders to identify the corresponding room impulse responses. In this thesis, we attempts to use IIR filters to reduce the filter order.Firstly, we measure true room impulse response data based on the MLS method.Secondly, the IIR filter coefficients are estimated using the SM method which provides an accurate and stable model for the room impulse response. our major focus is to develop a new iterative SM (NSM) method to reduce computational complexity.Finally, computer simulations are performed to compare the performances of AEC among IIR and FIR filters.
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Zheng, Zhang Quan, and 鄭張權. "Applications of IIR fiters to acoustic echo cancellation." Thesis, 1995. http://ndltd.ncl.edu.tw/handle/09846885509006838061.
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