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1
Macpherson, Kenneth Noble. "Low hardware cost, high speed, full-parallel finite impulse response digital filters on field programmable gate arrays". Thesis, University of Strathclyde, 2004. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.405323.
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2
Eshra, Islam. "Un FIRDAC programmable pour émetteurs RF re-configurable". Electronic Thesis or Diss., Sorbonne université, 2020. http://www.theses.fr/2020SORUS461.
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Le convertisseur numérique-analogique à réponse impulsionnelle finie (FIRDAC) proposé est programmable avec un ordre entièrement reconfigurable et des coefficients capables de fournir un ordre jusqu'à 62 et un rapport entre le coefficient maximum et minimum de 159. Cela a permis une large gamme de facteurs d'atténuation pouvant atteindre 100dB et une large gamme de bandes de transition normalisées (>0.0156). Le filtre FIRDAC a été conçu et implémenté en technologie CMOS 65 nm avec une surface active totale de 0,867 mm2. Au niveau du circuit, le FIRDAC peut atteindre une fréquence d'échantillonnage de 2,56 GHz pour une consommation en puissance moyenne de 9mW. Pour une entrée sinusoïdale, le filtre FIRDAC atteint un rapport signal sur bruit (SNR) jusqu'à 67,3 dB et une dynamique (SFDR) de 72 dBc. Les performances du filtre FIRDAC ont été testées dans des émetteurs QPSK, 16-QAM et 64-QAM avec OFDM et avec différentes largeurs de bande. Les simulations montrent un EVM (Error Vector Magnitude) de 2,66%, 1,9% et 2,29% respectivement. Une partie de ce travail concerne la conception du Front-End d’un émetteur RF programmable. Le Front-End RF est composé d'un mélangeur RF, d'un amplificateur de pré-puissance et d'un filtre LC réglable. Le Front-End RF complet a un gain programmable total de 23 dB avec un pas de 1,53 dB et capable de fonctionner sur une plage de 1,5 GHz à 5 GHz. La puissance RF de sortie maximale est de -11 dBm avec une consommation électrique de 23 mW. Les résultats montrent une dynamique (SFDR) maximum de -61,95 dBc pour deux tonalités à une fréquence porteuse de 4 GHz, tandis que pour un signal OFDM 16-QAM, l'EVM obtenu était de 4,76 %The first part of this work relates to the design and implementation of a programmable Finite Impulse Response Digital to Analog Converter (FIRDAC). The programmability is in the filter's order (N-1) and its coefficients. The proposed FIRDAC is capable of providing an order up to 62 and a ratio between maximum to minimum coefficient up to 159. This allowed the filter to provide up to 100dB of attenuation and a wide range of normalized transition-band (>0.0156). The FIRDAC filter has been designed and implemented in 65nm CMOS with total active area 0.867mm2. The FIRDAC can operate up to 2.56 GHz of sampling frequency at an average power consumption of 9mW. For a single tone input, the FIRDAC filter managed to provide an SNR up to 67.3dB and a SFDR of 72dBc. The FIRDAC filter was tested with different modulation techniques: OFDM, 16-QAM OFDM and 64-QAM OFDM having different channel Bandwidth. The circuit achieved an Error Vector Magnitude (EVM) of 2.66%, 1.9% and 2.29% respectively, complying with the LTE and the 802.11ac standards. The second part of this work relates to the design of a programmable RF front-end circuit. The RF front-end is composed of an analog RF mixer, a programmable Pre-Power Amplifier (PPA) and a tunable LC tank. The whole RF front-end introduced a total programmable gain of 23dB with a gain step of 1.53dB operating in the 1.5GHz - 5GHz frequency range. The maximum output RF power is -11dBm with a power consumption of 23mW. Simulation result showed a maximum SFDR of -61.95dBc for two tones at a carrier frequency of 4GHz. While for a 16-QAM OFDM signal, the obtained EVM was 4.76%
3
Broddfelt, Michel. "Design of a Finite-Impulse Response filter generator". Thesis, Linköping University, Department of Electrical Engineering, 2003. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-2027.
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In this thesis a FIR filter generator has been designed. The program generates FIR filters in the form of VHDL-files. Four different filter structures have been implemented in the generator, Direct Form (DF), Differential Coefficients Method (DCM), polyphase filters and (2-by-2) filters.
The focus of the thesis was to implement filter structures that create FIR filters with as low power consumption and area as possible.
The generaterator has been implemented i C++. The C++ program creates text-files with VHDL-code. The user must then compile and synthesize the VHDL-files. The program uses an text-file with the filter coefficients as input.
4
Carter, Scott Edward. "Finite impulse response utilizing the principles of superposition". Master's thesis, University of Central Florida, 1995. http://digital.library.ucf.edu/cdm/ref/collection/RTD/id/15187.
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University of Central Florida College of Engineering ThesisWindow functions have been greatly utilized in the synthesis of finite impulse response (FIR) filters implemented using surface acoustic wave (SAW) devices. The critical parameter in any FIR design in the impulse response length, which must be optimized for the given design specifications in order to reduce the size of each device. To this end, many design algorithms have been intorduced such as Remez excange, linear programming, and least mean squares. A new algorithm has been derived which is efficient and accurate for the design of arbitrary filter specifications requiring less computationsthan the current algorithms. The FIR design is applicaable to general SAW filter design and allows two weighted transducers to be designed in a near optimal method without the need to perform zero aplitting of de-convolution. The thesis first provides the definition of the window functions used for the design process. Then the overview of the design process is discussed using a flowchart of the modeling program for designing and FIR without tranducer separation and sample simulation is presented. Next, the effects of monotonically increasing sidelobes on the transition bandwidth are discussed. This is followed by a discussion of the addition of arbitary phase to the filter design requirements. Next, the separation of the response into a two transducer design utilizing the two window function series is explained. Finally, the results are discussed and compared with other design techniques.
M.S.;
Electrical and Computer Engineering
Engineering;
Electrical Engineering
69 p.
ix, 69 leaves, bound : ill. ; 28 cm.
5
Sokol, Thomas M. "Finite impulse response (FIR) filters to simulate response of an antenna". Connect to resource, 2006. http://hdl.handle.net/1811/6442.
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Thesis (Honors)--Ohio State University, 2006.Title from first page of PDF file. Document formatted into pages: contains 42 p.; also includes graphics. Includes bibliographical references (p. 42). Available online via Ohio State University's Knowledge Bank.
6
Bishop, Carlton Delos. "Finite impulse response filter design using cosine series functions". Doctoral diss., University of Central Florida, 1988. http://digital.library.ucf.edu/cdm/ref/collection/RTD/id/43377.
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University of Central Florida College of Engineering ThesisWindow functions have been extensively used for the design of SAW filters. The classical truncated cosine series functions, such as the Hamming and Blackmann functions, are only a few of an infinite set of such functions. The derivation of this set of functions from orthonormal basis sets and the criteria for obtaining the constant coefficients of the functions are presented. These functions are very useful because of the closed-form expressions and their easily recognizable Fourier transform. Another approach to the design of Gaussian shaped filters having a desired sidelobe level using a 40 term cosine series will be presented as well. This approach is again non-iterative and a near equi-ripple sidelobe level filter could be achieved. A deconvolution technique will also be presented. this has the advantage of being non-iterative, simple and fast. This design method produces results comparable to the Dolph-Chebyshev technique.
Ph.D.
Doctorate
Electrical Engineering and Communication
Engineering
Electrical Engineering
41 p.
vii, 41 leaves, bound : ill. ; 28 cm.
7
BRUEGGE, THOMAS JOSEPH. "THE USE OF FINITE IMPULSE RESPONSE KERNELS FOR IMAGE RESTORATION". Diss., The University of Arizona, 1985. http://hdl.handle.net/10150/187974.
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This dissertation examines the suitability of Display-Processor (DP) image computers for image enhancement and restoration tasks. Because the major architectural feature of the DP devices is their ability to rapidly evaluate finite impulse response (FIR) convolutions, much of the study focusses on the use of spatial-domain FIR convolutions to approximate Fourier-domain filtering. When the enhancement task requires the evaluation of only a single convolution, it is important that the FIR kernel used to implement the convolution is designed so that the resulting output is a good approximation of the desired output. A Minimum-Mean-Squared-Error design criterion is introduced for the purpose of FIR kernel design and its usefulness is demonstrated by showing some results of its use. If the restoration or enhancement task requires multiple convolutions in an iterative algorithm, it is important to understand how the truncation of the kernel to a finite region of support will affect the convergence properties of an algorithm and the output of the iterative sequence. These questions are examined for a limited class of nonlinear restoration algorithms. Because FIR convolutions are most efficiently performed on computing machines that have limited precision and are usually limited to performing fixed-point arithmetic, the dissertation also examines the effects of roundoff error on output images that have been computed using fixed point math. The number of bits that are needed to represent the data during a computation is algorithm dependent, but for a limited class of algorithms, it is shown that 12 bits are sufficient. Finally, those architectural features in a DP that are necessary for useful enhancement and restoration operations are identified.
8
Campbell, Roy Lee. "Performance assessment of the finite impulse response Adaptive Line Enhancer". Diss., Mississippi State : Mississippi State University, 2002. http://library.msstate.edu/etd/show.asp?etd=etd-05222002-085151.
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9
Li, Liwei. "Microwave Photonic Signal Processing Techniques based on Finite Impulse Response Configurations". Thesis, The University of Sydney, 2013. http://hdl.handle.net/2123/9477.
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Microwave photonic signal processing based on optical delay line structure is a powerful technique for processing high speed signals. Finite impulse response microwave photonic filters are of particular interest because of their inherent flexibility in realizing arbitrary transfer functions and linear phase, together with coherent interference free and phase-induced intensity noise free. This thesis proposes a number of effective solutions to eliminate the undesirable degradation of RF response so that microwave photonic filters can be used in practice. Experimental results demonstrate successfully multi-tap microwave photonic filters with both continuous tunability over a wide range and the elimination of high frequency limitations. One major problem of current true-time-delay is the restricted number of delay lines, and another is the difficulty in controlling the signal power. To overcome these restrictions, this thesis presents an array of multiple true-time-delay elements that can be independently and continuously tuned. Moreover, a new technique is reported to overcome the tap-delay-variation, which is caused by the interaction of the equally-spaced frequency comb lines with the non-uniform group delay of the primary fiber delay line that arises from higher-order dispersion factors. In order both to eliminate non-uniform tap-delay variations and to realize independently tunable time delays, for the first time, a hybrid phase-time shifter structure is reported. It can simultaneously form true-time-delays and phase shifts to the input RF signals within one device. To overcome the problem of periodic resonances that are inherent in microwave photonic signal processors, three different approaches for realizing single bandpass response are presented. Experimental results present single bandpass RF responses with wideband tunability at high frequencies.
10
Alm, Erik. "Area and Power Efficiency of Multiplier-Free Finite Impulse Response Filters". Thesis, KTH, Skolan för elektroteknik och datavetenskap (EECS), 2018. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-237417.
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In digital radio systems, a large number of finite impulse response filters are typically used. Due to their nature of operation, such filters require many multiplication operations, leading to great costs in terms of both chip area and power consumption. For cost reduction reasons, there is a strong business case for implementing these filters without general multipliers so as to reduce the area and power consumption of the overall system.This thesis explores a method of implementing finite impulse response halfband filters without general multipliers, by using a special filter structure and replacing multipliers with sequences of binary shifts and additions. The savings in terms of area and power consumption are estimated and compared to a conventional filter (with a common structure) implementation containing general multipliers, as well as the same conventional filter implemented without general multipliers by means of manipulating its coefficients such that they can be implemented with shifts and additions.The results show that while using the special filter structure with shifts and additions consumes less area and power than a conventional filter with general multipliers, employing simpler methods to obtain coefficients implementable with shifts and additions in a conventional filter structure produces smaller filters consuming less power. Moreover, the results of this thesis show that using methods allowing for multiplier-free filter implementations with conventional filter structures seems favorable, hence further investigation of such methods is recommended. Future studies could also focus on methods applicable to filters with support for dynamic coefficients.Digitala radiosystem innehåller ofta ett stort antal filter med ändliga impulssvar. På grund av hur sådana filter opererar krävs ett stort antal multiplikationer, vilka implementerade i hårdvara tenderar ockupera stor kiselyta och konsumera hög effekt. För att reducera kostnader finns det därför ett starkt incitament att implementera dessa filter utan generella multiplikatorer. Detta examensarbete utforskar en metod för att implementera digitala halvbandsfilter utan generella multiplicerare, genom att använda en speciell filterstruktur och ersätta multiplikationerna med sekvenser av binära skiftoperationer och additioner. Besparingarna i termer av effektförbrukning och kiselyta uppskattas och jämförs med ett konventionellt implementerat filter (med en vanlig struktur) som uppfyller samma specifikationer samt samma filter med koefficienter manipulerade så att de kan uttryckas som sekvenser av binära skiftoperationer och additioner. Resultaten visar att såväl kiselyta som effektförbrukning ter sig lägre för filtret implementerat med den speciella strukturen och utan generella multiplicerare än för det konventionella filtret innehållande generella multiplicerare. Dock visas också att ännu större besparingar uppnås genom att använda den konventionella filterstrukturen men med koefficienter ma-nipulerade så att dessa kan implementeras utan multiplicerare. Överlag ärslutsatsen att konventionella filterstrukturer i kombination med metoder för att göra dess koefficienter implementerbara utan multiplicerare verkar mer lovande och att ytterligare studier av sådana metoders förtjänster bör stud-eras. Framtida studier skulle även kunna ta i beaktande metoder som ärapplicerbara på filter med icke-konstanta koefficienter.
11
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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Talej, Elie N. "A VLSI design of a finite impulse response low-pass digital filter". Ohio : Ohio University, 1988. http://www.ohiolink.edu/etd/view.cgi?ohiou1182871591.
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13
Rajagopal, Ravikiran. "An algebraic view of multidimensional multiple-input multiple-output finite impulse response equalizers". The Ohio State University, 2003. http://rave.ohiolink.edu/etdc/view?acc_num=osu1070398845.
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14
Zheng, Man. "Modeling problems using Bayes' rule for finite impulse response models and Markov models". Doctoral thesis, Kyoto University, 2021. http://hdl.handle.net/2433/263782.
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15
Agathangelou, Marios Christaki. "Contour smoothing in segmented images for object-based compression". Thesis, Imperial College London, 1998. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.286288.
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Akyurek, Sefa. "The Implementation Complexity Of Finite Impulse Response Digital Filters Under Different Coefficient Quantization Schemes And Realization Structures". Master's thesis, METU, 2004. http://etd.lib.metu.edu.tr/upload/2/12605586/index.pdf.
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It has been aimed to investigate the complexity of discrete-coefficient FIR filters when they are implemented in transposed form and the coefficient redundancy is removed by the n-Dimensional Reduced Adder Graph (RAG-n) approach. Filters with coefficients represented by different quantization schemes have been designed or selected from the literturetheir transposed form implemetations after RAG-n process have been compared in terms of complexity. A Genetic Algorithm (GA) based design algorithm has been implemented and used for the design of integer coefficient filters. Algorithms for the realization of filter coefficients in Canonic Signed Digit (CSD) form and realization of n-Dimensional Reduced Adder Graph (RAG-n) have also been implemented. Filter performance is measured as Normalized Peak Ripple Magnitude and implementation complexity as the number of adders used to implement filter coefficients. Number of adders used to implement filter coefficients is calculated by using two different methods: CSD and RAG-n. RAG-n method has been applied to FIR digital filter design methods that don&rsquo
t consider reduction of implementation complexity via RAG-n with transposed direct form filter structure. For implementation complexity, it is concluded that &ldquo
RAG-n algorithm with transposed direct form filter structure&rdquo
provides better results over the &ldquo
CSD, SPT coefficient design followed by transposed direct form filter structure&rdquo
in terms of number of adders used in the implementation.
17
Oguibe, Chukwuma Nnamdi. "A finite element study of the elastic-plastic indentation and the response of composite structural components to impact and impulse loading". Thesis, Leeds Beckett University, 1996. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.320299.
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Garcia, Lorca Federico. "Filtres récursifs temps réel pour la détection de contours : optimisations algorithmiques et architecturales". Paris 11, 1996. http://www.theses.fr/1996PA112439.
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Dans cette thèse on s'intéresse à deux aspects différents : conceptuel et réalisationel, sur lesquels portent les quatre innovations présentées. Si celles-ci sont illustrées par une application au détecteur de contours de Deriche, elles sont facilement généralisables à d'autres détecteurs qu'ils soient basés sur le calcul de maxima locaux de la dérivée première, ou le calcul des passages par zéro du laplacien. Les filtres à réponse impulsionnelle infinie symétriques ou anti-symétriques peuvent être réalisés sous forme cascade. Le filtre de lissage peut être défini par intégration numérique du filtre dérivateur optimal. Tout filtre détecteur de contours à noyau large peut être considéré comme un filtre de lissage bidimensionnel à noyau large suivi d'un simple filtre Sobel. L’utilisation d'opérateurs blocs série offre le meilleur compromis surface rapidité pour l'intégration en ASICS ou FPGAS. Nous proposons une architecture câblée temps réel optimale en compacité et simplicité de la version récursive du filtre détecteur de contours de Deriche. Nous exposons la méthode qui conduit à notre solution. A travers cette expérience, nous souhaitons transmettre aux concepteurs d'outils de CAO un certain nombre d'idées qui doivent à notre avis être exploitées afin que des outils tels que les graphes flots de données ou les langages synchrones assistent efficacement l'architecte dans les problèmes d'ordonnancement, d'allocation et de repliement temporel du graphe vers l'architecture.
19
Bicen, Ahmet Ozan. "Fundamentals of molecular communication over microfluidic channels". Diss., Georgia Institute of Technology, 2016. http://hdl.handle.net/1853/55009.
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The interconnection of molecular machines with different functionalities to form molecular communication systems can increase the number of design possibilities and overcome the limited reliability of the individual molecular machines. Artificial information exchange using molecular signals would also expand the capabilities of single engineered cell populations by providing them a way to cooperate across heterogeneous cell populations for the applications of synthetic biology and lab-on-a-chip systems. The realization of molecular communication systems necessitates analysis and design of the communication channel, where the information carrying molecular signal is transported from the transmitter to the receiver. In this thesis, significant progress towards the use of microfluidic channels to interconnect molecular transmitter and receiver pairs is presented. System-theoretic analysis of the microfluidic channels are performed, and a finite-impulse response filter is designed using microfluidic channels. The spectral density of the propagation noise is studied and the additive white Gaussian noise channel model is developed. Memory due to inter-diffusion of the transmitted molecular signals is also modeled. Furthermore, the interference modeling is performed for multiple transmitters and its impact on the communication capacity is shown. Finally, the efficient sampling of the signal transduction by engineered bacterial receivers connected to a microfluidic channel is investigated for the detection of the pulse-amplitude modulated molecular signals. This work lays the foundation for molecular communication over microfluidic channels that will enable interconnection of engineered molecular machines.
20
Srinivas, L. "FIR System Identification Using Higher Order Cumulants -A Generalized Approach". Thesis, Indian Institute of Science, 1994. https://etd.iisc.ac.in/handle/2005/637.
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The thesis presents algorithms based on a linear algebraic solution for the identification of the parameters of the FIR system using only higher order statistics when only the output of the system corrupted by additive Gaussian noise is observed.
All the traditional parametric methods of estimating the parameters of the system have been based on the 2nd order statistics of the output of the system. These methods suffer from the deficiency that they do not preserve the phase response of the system and hence cannot identify non-minimum phase systems. To circumvent this problem, higher order statistics which preserve the phase characteristics of a process and hence are able to identify a non-minimum phase system and also are insensitive to additive Gaussian noise have been used in recent years.
Existing algorithms for the identification of the FIR parameters based on the higher order cumulants use the autocorrelation sequence as well and give erroneous results in the presence of additive colored Gaussian noise. This problem can be overcome by obtaining algorithms which do not utilize the 2nd order statistics.
An existing relationship between the 2nd order and any Ith order cumulants is generalized to a relationship between any two arbitrary k, Ith order cumulants. This new relationship is used to obtain new algorithms for FIR system identification which use only cumulants of order > 2 and with no other restriction than the Gaussian nature of the additive noise sequence. Simulation studies are presented to demonstrate the failure of the existing algorithms when the imposed constraints on the 2nd order statistics of the additive noise are violated while the proposed algorithms perform very well and give consistent results.
Recently, a new algebraic approach for parameter estimation method denoted the Linear Combination of Slices (LCS) method was proposed and was based on expressing the FIR parameters as a linear combination of the cumulant slices. The rank deficient cumulant matrix S formed in the LCS method can be expressed as a product of matrices which have a certain structure. The orthogonality property of the subspace orthogonal to S and the range space of S has been exploited to obtain a new class of algorithms for the estimation of the parameters of a FIR system. Numerical simulation studies have been carried out to demonstrate the good behaviour of the proposed algorithms.
Analytical expressions for the covariance of the estimates of the FIR parameters of the different algorithms presented in the thesis have been obtained and numerical comparison has been done for specific cases.
Numerical examples to demonstrate the application of the proposed algorithms for channel equalization in data communication and as an initial solution to the cumulant matching nonlinear optimization methods have been presented.
21
Srinivas, L. "FIR System Identification Using Higher Order Cumulants -A Generalized Approach". Thesis, Indian Institute of Science, 1994. http://hdl.handle.net/2005/637.
Texto completoResumen
The thesis presents algorithms based on a linear algebraic solution for the identification of the parameters of the FIR system using only higher order statistics when only the output of the system corrupted by additive Gaussian noise is observed.
All the traditional parametric methods of estimating the parameters of the system have been based on the 2nd order statistics of the output of the system. These methods suffer from the deficiency that they do not preserve the phase response of the system and hence cannot identify non-minimum phase systems. To circumvent this problem, higher order statistics which preserve the phase characteristics of a process and hence are able to identify a non-minimum phase system and also are insensitive to additive Gaussian noise have been used in recent years.
Existing algorithms for the identification of the FIR parameters based on the higher order cumulants use the autocorrelation sequence as well and give erroneous results in the presence of additive colored Gaussian noise. This problem can be overcome by obtaining algorithms which do not utilize the 2nd order statistics.
An existing relationship between the 2nd order and any Ith order cumulants is generalized to a relationship between any two arbitrary k, Ith order cumulants. This new relationship is used to obtain new algorithms for FIR system identification which use only cumulants of order > 2 and with no other restriction than the Gaussian nature of the additive noise sequence. Simulation studies are presented to demonstrate the failure of the existing algorithms when the imposed constraints on the 2nd order statistics of the additive noise are violated while the proposed algorithms perform very well and give consistent results.
Recently, a new algebraic approach for parameter estimation method denoted the Linear Combination of Slices (LCS) method was proposed and was based on expressing the FIR parameters as a linear combination of the cumulant slices. The rank deficient cumulant matrix S formed in the LCS method can be expressed as a product of matrices which have a certain structure. The orthogonality property of the subspace orthogonal to S and the range space of S has been exploited to obtain a new class of algorithms for the estimation of the parameters of a FIR system. Numerical simulation studies have been carried out to demonstrate the good behaviour of the proposed algorithms.
Analytical expressions for the covariance of the estimates of the FIR parameters of the different algorithms presented in the thesis have been obtained and numerical comparison has been done for specific cases.
Numerical examples to demonstrate the application of the proposed algorithms for channel equalization in data communication and as an initial solution to the cumulant matching nonlinear optimization methods have been presented.
22
Uzinski, Julio Cezar [UNESP]. "A state-space parameterization for perfect-reconstruction wavelet FIR filter banks with special orthonormal basis functions". Universidade Estadual Paulista (UNESP), 2016. http://hdl.handle.net/11449/146716.
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Conselho Nacional de Desenvolvimento Científico e Tecnológico (CNPq)
Esta tese apresenta uma parametrização no espaço de estados para a transformada wavelet rápida. Esta parametrização é baseada em funções de base ortonormal e filtros de resposta finita ao impulso simultaneamente, uma vez que, a transformada rápida wavelet é um algoritmo que consiste em decompor sinais no domínio do tempo em sequências de coeficientes baseados numa base ortogonal de funções wavelet. Deste modo, vantagens apresentadas por ambas as propostas são incorporadas. Modelos de resposta finita ao impulso têm propriedades atrativas como vantagens computacionais e analíticas, garantia de estabilidade BIBO e robustez para a mudança de alguns parâmetros, dentre outras. Por outro lado, séries de funções de base ortonormal têm características que as fazem atrativas para a modelagem de sistemas dinâmicos, como ausência de recursão da saída, a não necessidade de se conhecer previamente a estrutura exata do vetor de regressão, possibilidade de aumentar a capacidade de representação do modelo aumentando-se o número de funções ortonormais utilizadas, desacoplamento natural das saídas em modelos multivariáveis; tolerância a dinâmicas não modeladas. Além disso, a realização no espaço de estados é mínima. A contribuição deste trabalho consiste no desenvolvimento de uma realização no espaço de estados para bancos de filtros wavelet, em que há a presença explícita de parâmetros que podem ser livremente ajustados mantendo as propriedades de reconstrução perfeita e ortonormalidade. Para ilustrar o funcionamento e as vantagens da técnica proposta, alguns exemplos de decomposição de sinais no contexto de processamento de sinais mostrando que ela proporciona os mesmos coeficientes wavelet que a transformada wavelet rápida, e uma aplicação em controle através de realimentação dinâmica de estados também são apresentados nesta tese.
This thesis presents a state-space parameterization for the fast wavelet transform. This parameterization is based on orthonormal basis functions and finite impulse response filters at the same time, since the fast wavelet transform is an algorithm, which converts a signal in the time domain into a sequence of coefficients based on an orthogonal basis of small finite wavelet functions. Advantages presented by both proposals are incorporated. Finite impulse response systems have attractive properties, for instance, computational and analytical advantages, BIBO stability and robustness guarantee to some parameter changes, and others. On the other hand, orthonormal basis functions have some characteristics that make them attractive for dynamic systems modeling, examples are, output recursion absence, not requiring prior regression vector exact structure knowledge; possibility of increasing the model representation capacity by increasing the number of orthonormal functions employed; natural outputs uncoupling in multivariable models; tolerance to unmodeled dynamics, and others. Furthermore, the state-space realization is minimal. The contribution of this work consists in the development of a state-space realization for a wavelet filter bank, with the explicit presence of the parameters that can be freely adjusted, keeping perfect-reconstruction and orthonormality guarantees. In order to illustrate advantages and how the proposed technique works, some decomposition examples in signal processing context are presented showing that it provides the same wavelet coefficients as the fast wavelet transform, and an application on dynamic state feedback control is also presented in this thesis.
CNPq: 160545/2013-7
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Uzinski, Julio Cezar. "A state-space parameterization for perfect-reconstruction wavelet FIR filter banks with special orthonormal basis functions /". Ilha Solteira, 2016. http://hdl.handle.net/11449/146716.
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Orientador: Francisco Villarreal AlvaradoResumo: Esta tese apresenta uma parametrização no espaço de estados para a transformada wavelet rápida. Esta parametrização é baseada em funções de base ortonormal e filtros de resposta finita ao impulso simultaneamente, uma vez que, a transformada rápida wavelet é um algoritmo que consiste em decompor sinais no domínio do tempo em sequências de coeficientes baseados numa base ortogonal de funções wavelet. Deste modo, vantagens apresentadas por ambas as propostas são incorporadas. Modelos de resposta finita ao impulso têm propriedades atrativas como vantagens computacionais e analíticas, garantia de estabilidade BIBO e robustez para a mudança de alguns parâmetros, dentre outras. Por outro lado, séries de funções de base ortonormal têm características que as fazem atrativas para a modelagem de sistemas dinâmicos, como ausência de recursão da saída, a não necessidade de se conhecer previamente a estrutura exata do vetor de regressão, possibilidade de aumentar a capacidade de representação do modelo aumentando-se o número de funções ortonormais utilizadas, desacoplamento natural das saídas em modelos multivariáveis; tolerância a dinâmicas não modeladas. Além disso, a realização no espaço de estados é mínima. A contribuição deste trabalho consiste no desenvolvimento de uma realização no espaço de estados para bancos de filtros wavelet, em que há a presença explícita de parâmetros que podem ser livremente ajustados mantendo as propriedades de reconstrução perfeita e ortonormalidade. ... (Resumo completo, clicar acesso eletrônico abaixo)
Doutor
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Fayolle, Gérard. "Conceptions et applications de circuits à mémoire de courant basse tension". Grenoble 1, 1998. http://www.theses.fr/1998GRE10151.
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Ce travail presente deux circuits bases sur les memoires de courant. Les applications developpees sont capables de travailler avec des alimentations en tension de 3v et moins. Pour ces realisations, il a ete necessaire de developper a la fois de nouvelles memoires de courants ainsi que des architectures originales pour repondre a tous les criteres des cahiers des charges. La premiere realisation est un fir pour une application video dont la fonction est, en sortie d'un filtre continu, d'y transferer une partie des contraintes de rejection. La phase lineaire que presente ce type de filtre se prete parfaitement bien a cette application tres sensible au delai de groupe. L'architecture developpee utilise un adressage cyclique qui permet de fortement reduire les degradations engendrees par les memoires de courant tout en conservant la dynamique. Le deuxieme circuit est un modulateur sigma-delta d'ordre 2 dont les performances requises sont une resolution superieure a 12 bits dans la bande 300-3400 hz, une surface d'integration reduite et une faible consommation. Pour arriver a obtenir la precision necessaire avec la faible tension d'alimentation, une nouvelle memoire a ete developpee. Sa particularite est d'utiliser un transistor de memorisation en zone triode qui lui confere une grande linearite. Elle laisse esperer des resultats en terme de linearite et de resolution proche de 14 bits.
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Rangel, Luciano dos Santos. "Identificação de danos estruturais utilizando dados no domínio do tempo provenientes de ensaios de vibração". Universidade do Estado do Rio de Janeiro, 2014. http://www.bdtd.uerj.br/tde_busca/arquivo.php?codArquivo=7146.
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Fundação Carlos Chagas Filho de Amparo a Pesquisa do Estado do Rio de JaneiroO presente trabalho aborda o problema de identificação de danos em uma estrutura a partir de sua resposta impulsiva. No modelo adotado, a integridade estrutural é continuamente descrita por um parâmetro de coesão. Sendo assim, o Modelo de Elementos Finitos (MEF) é utilizado para discretizar tanto o campo de deslocamentos, quanto o campo de coesão. O problema de identificação de danos é, então, definido como um problema de otimização, cujo objetivo é minimizar, em relação a um vetor de parâmetros nodais de coesão, um funcional definido a partir da diferença entre a resposta impulsiva experimental e a correspondente resposta prevista por um MEF da estrutura. A identificação de danos estruturais baseadas no domínio do tempo apresenta como vantagens a aplicabilidade em sistemas lineares e/ou com elevados níveis de amortecimento, além de apresentar uma elevada sensibilidade à presença de pequenos danos. Estudos numéricos foram realizados considerando-se um modelo de viga de Euler-Bernoulli simplesmente apoiada. Para a determinação do posicionamento ótimo do sensor de deslocamento e do número de pontos da resposta impulsiva, a serem utilizados no processo de identificação de danos, foi considerado o Projeto Ótimo de Experimentos. A posição do sensor e o número de pontos foram determinados segundo o critério D-ótimo. Outros critérios complementares foram também analisados. Uma análise da sensibilidade foi realizada com o intuito de identificar as regiões da estrutura onde a resposta é mais sensível à presença de um dano em um estágio inicial. Para a resolução do problema inverso de identificação de danos foram considerados os métodos de otimização Evolução Diferencial e Levenberg-Marquardt. Simulações numéricas, considerando-se dados corrompidos com ruído aditivo, foram realizadas com o intuito de avaliar a potencialidade da metodologia de identificação de danos, assim como a influência da posição do sensor e do número de dados considerados no processo de identificação. Com os resultados obtidos, percebe-se que o Projeto Ótimo de Experimentos é de fundamental importância para a identificação de danos.
The present work deals with the damage identification problem in mechanical structures from their impulse response. In the adopted model, the structural integrity is continually described by a cohesion parameter and the finite element model (FEM) is used to spatially discretize both the displacement and cohesion fields. The damage identification problem is then posed as an optimization one, whose objective is to minimize, with respect to the vector of nodal cohesion parameters, a functional based on the difference between the experimentally obtained impulse response and the corresponding one predicted by an FEM of the structure. The damage identification problem built on the time domain presents some advantages, as the applicability in linear systems with high levels of damping an/or closed spaced modes, and in nonlinear systems. Besides, the time domain approaches present high sensitivities to the presence of small damages. Numerical studies were carried out considering a simply supported Euler-Bernoulli beam. Optimal experiment design techniques were considered with the aim at determining the optimal position of the displacement sensor and also the number of points of the impulse response to be considered in the identification process. The Differential Evolution optimization method and the Levenberg-Marquardt method were considered to solve the inverse problem of damage identification. Numerical analysis were carried out in order to assess the influence, on the identification results, of noise in the synthetic experimental data, of the sensor position, and of the number of points retained in the impulse response. The presented results shown the potentiality of the proposed damage identification approach and also the importance of the optimal experiment design for the quality of the identification. al importance for the identification of damage.
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"Micromachined Acoustic Programmable Tunable Finite Impulse Response (FIR) Filters for Microwave Applications". Doctoral diss., 2013. http://hdl.handle.net/2286/R.I.18723.
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abstract: This dissertation proposes a miniature FIR filter that works at microwave frequencies, whose filter response can ideally be digitally programmed. Such a frequency agile device can find applications in cellular communications and wireless networking. The basic concept of the FIR filter utilizes a low loss acoustic waveguide of appropriate geometry that acts as a traveling wave tapped-delay line. The input RF signal is applied by an array of capacitive transducers at various locations on the acoustic waveguide at one end that excites waves of a propagating acoustic mode with varying spatial delays and amplitudes which interfere as they propagate. The output RF signal is picked up at the other end of the waveguide by another array of capacitive transducers. Tuning of the FIR filter coefficients is realized by controlling the DC voltage profile applied to the individual transducers which essentially shapes the overall filter response. Equivalent circuit modeling of the capacitive transducer, acoustic waveguide and transducer-line coupling is presented in this dissertation. A theoretical model for the filter is developed from a general theory of an array of transducers exciting a waveguide and is used to obtain a set of filter design equations. A MATLAB based circuit simulator is developed to simulate the filter responses. Design parameters and simulation results obtained for an example waveguide structure are presented and compared to the values estimated by the theoretical model. A waveguide structure utilizing the Rayleigh-like mode of a ridge is then introduced. A semi-analytical method to obtain propagating elastic modes of such a ridge waveguide etched in an anisotropic crystal is presented. Microfabrication of a filter based on ridges etched in single crystal Silicon is discussed along with details of the challenges faced. Finally, future work and a few alternative designs are presented that can have a better chance of success. Analysis and modeling work to this point has given a good understanding of the working principles, performance tradeoffs and fabrication pitfalls of the proposed device. With the appropriate acoustic waveguide structure, the proposed device could make it possible to realize miniature programmable FIR filters in the GHz range.Dissertation/Thesis
Ph.D. Electrical Engineering 2013
27
Samprit. "Finite impulse response filter implementation using LMS algorithm". Thesis, 2007. http://ethesis.nitrkl.ac.in/4269/1/%E2%80%9CFINITE_IMPULSE_RESPONSE_FILTER.pdf.
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The principle objective this project is to determine the coefficients of the FIR filters that met desired specifications. The determinations of coefficients involve channel equalization, system identification and SNR vs. BER plot using LMS algorithm. This project is to investigate the performance of an FIR filter equalizer for data transmission over a channel that causes intersymbol interference. FIR filter removes unwanted parts of the signal, such as random noise, or extracts the useful parts of the signal, such as the components lying within a certain frequency range. In signal processing, there are many instances in which an input signal to a system contains extra unnecessary content or additional noise which can degrade the quality of the desired portion. In such cases we may remove or filter out the useless samples using FIR filters. Here by using LMS algorithm in channel equalization we determined coefficients in Matlab programming. In project by inducing white Gaussian signal or random signal (noise) with data signal we equalize for data transmission over a channel. They can easily be designed to be "linear phase" (and usually are). Put simply, linearphase filters delay the input signal, but don’t distort its phase. They are simple to implement. On most DSP microprocessors, the FIR calculation can be done by looping a single instruction. They are suited to multi-rate applications. By multi-rate, we mean either "decimation" (reducing the sampling rate), "interpolation" (increasing the sampling rate), or both. Whether decimating or interpolating, the use of FIR filters allows some of the calculations to be omitted, thus providing an important computational efficiency. In contrast, if IIR filters are used, each output must be individually calculated, even if it that output will discard (so the feedback will be incorporated into the filter). They have desirable numeric properties. In practice, all DSP filters must be implemented using "finite-precision" arithmetic, that is, a limited number of bits. The use of finiteprecision arithmetic in IIR filters can cause significant problems due to the use of feedback, but FIR filters have no feedback, so they can usually be implemented using fewer bits, and the designer has fewer practical problems to solve related to non-ideal arithmetic. In the implementation of FIR in system identification estimated channel parameters are almost same as channel parameter. So FIR filter provides effective way to remove unwanted signals, channel equalization and system identification.
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Oren, Joel A. "Design of an asynchronous third-order finite impulse response filter". Thesis, 1994. http://hdl.handle.net/1957/35774.
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With the increased demand for complex digital signal processing systems,
real-time signal processing requires higher throughput systems. In the past, the
throughput has been increased by increasing the clock rates, but
synchronization can become increasingly more difficult. Recently there has
been renewed interest in designing asynchronous digital systems. In an
asynchronous system, there is no global clock, and all modules communicate
through handshaking. In this thesis we demonstrate an implementation of an
FIR filter using asynchronous digital circuit techniques. These asynchronous
design techniques are used to test whether a practical signal processing filter
can be implemented with asynchronous logic. A third-order four-bit filter is
developed and simulated with SPICE, comparing favorably with other available
technologies in speed and power consumption. Although in practice 8-16 bits
are needed, this work is sufficient to demonstrate the feasibility of asynchronous
circuits for filtering applications. A chip is laid out in 2 micron CMOS, and
testing shows that it has a speed-power product comparable with asynchronous
designs fabricated by others.Graduation date: 1994
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Ralhan, Sameer. "Robust model predictive control of linear finite impulse response plants". Thesis, 1998. http://hdl.handle.net/1911/17206.
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Model Predictive Control (MPC) has become one of the dominant methods of chemical process control in terms of successful industrial applications. A rich theory has been developed to study the closed loop stability of MPC algorithms when the plant model is perfect, referred to as the nominal stability problem. In practical applications, however, the process model is never perfect and nominal stability results are not strictly applicable. The primary disadvantage of the current design techniques for MPC is their inability to deal explicitly with the plant model uncertainty.
In this thesis we develop a new framework for robust MPC synthesis that allows explicit incorporation of the plant uncertainty description in the problem formulation. Model uncertainty is parameterized by ellipsoid bounds on the Finite Impulse Response (FIR) parameters of the plant model. Robust stability is achieved through the addition of constraints that prevent the sequence of the optimal controller costs from increasing for the true plant. The framework developed here can also be used for constant output disturbance rejection. Ward input and soft output constraints can be easily added to the algorithms without affecting the closed loop stability properties.
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Chou, Yung-Chen y 周昀辰. "Incorporation of Finite Impulse Response Neural Network into the FDTD Method". Thesis, 2005. http://ndltd.ncl.edu.tw/handle/34703736828723907065.
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碩士國立中山大學
電機工程學系研究所
93
The Finite-Difference Time-Domain Method (FDTD) is a very powerful numerical method for the full wave analysis electromagnetic phenomena. Due to its flexibility, it can be used to solve numerous electromagnetic scattering problems on microwave circuits, dielectrics, and electromagnetic absorption in biological tissue at microwave frequencies. However, it needs so much computation time to simulate microwave integral circuits by applying the FDTD method. If the structure we simulated is complicated and we want to obtain accurate frequency domain scattering parameters, the simulation time will be so much longer that the efficiency of simulation will be bad as well. Therefore, in the thesis, we introduce an artificial neural networks (ANN) method called “Finite Impulse Response Neural Networks (FIRNN)” can speed up the FDTD simulation time. In order to boost the efficiency of the FDTD simulation time by stopping the simulation after a sufficient number of time steps and using FIRNN as a predictor to predict time series signal.
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Chung, Pwu-Yueh y 鍾步嶽. "Design of a Finite-Impulse Response Filter Using Redundant Number Representations". Thesis, 1997. http://ndltd.ncl.edu.tw/handle/14495007529332220541.
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碩士大同大學
電機工程研究所
85
A design of a 10-tap linear phase finite-impulse response filter using redundant number representations is presented in the thesis. In this design, the inputs and outputs are represented in two's complement binary fomi and the internal numbers are represented in radix-2 redundant digit form. This redundant result is then converted to two's complement binary form by using the proposed fast parallel conversion scheme. The use of a redundant number system leads to fast carry-free addition, where the cany propagates only through two stages, independent of the word length. Thus, the FIR filter can operate at very high frequency. However, this design trades off area with time and needs more area. To realize this circuit in a single chip, we use the CADENCE tools and the cell library, COMPASS06 (0.6μm single-poly, double-metal CMOS technology), to implement it. The gate count of tills filter is about 17596 gates. The operating clock can be speeded up to lOOMHz.
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Yang, Yu-Jen y 楊友仁. "Implementation of Finite Impulse Response Digital Filter Using Partial Reconfigurable FPGA Platform". Thesis, 2009. http://ndltd.ncl.edu.tw/handle/75036492372523386636.
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碩士元智大學
資訊工程學系
97
This paper presents a parallel FIR(Finite Impulse Response) filter system design ,using PR(Partial Reconfiguration) to change tap of FIR ,which can achieve high flexibility , high performance ,and shorten the time of configuration. In the present thesis, we use the Verilog HDL within Xilinx ISE 9.1i design tool to complete this architecture. After that, we use FPGA for function simulation and verify computation data. Until simulation result is correct , using Xilinx Planahead 9.2 to merge all of the architecture ,it generates full configuration circuit file and partial reconfiguration circuit file. Then, to verify by using iMPACT to download full configuration circuit file to FPGA, and download partial reconfiguration circuit file to observe the advantage of shorten time of configuration by using partial reconfiguration.
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Yuan, Tarng Ing y 唐英原. "The design of finite impulse response digital filter based on residue number system". Thesis, 1996. http://ndltd.ncl.edu.tw/handle/06457508171679041743.
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碩士國立臺灣科技大學
電子工程學系
84
A residue code finite impulse response (FIR) digital filter consists of three subsystems: residue analog-to-digital converter (ADC), convolution operation unit, and residue digital-to analog converter (DAC). However, additional circuits are required to do the conversions between binary code and residue code when they are used in the system mentioned above. Since conventional DAC and ADC circuits are binary code oriented, the circuits increase not only system processing time but also power consumption and chip die size. Consequently, in this thesis a new architecture that combines both operations of conventional ADC and DAC, as well as the required conversions between binary and residue codes is proposed to simplify the input/output conversion problems. In the input/output conversion subsystem: residue ADC is realized by the chopper- type comparator with subranging approach along with a pipelined scheme to reduce comparator count and increase the conversion rate. To simplify the system architecture and circuit design, voltage scaling approach is used in the residue DAC circuit. As for the convolution operation unit,instead of using traditional arithmetic circuit, a permutation network based scheme is used to increase the operation speed and simplify circuit designs. Finally, the whole system is realized with full-custom VLSI technology. The chip sets are as follows. The residue ADC has been integrated into an area of 2804um ×3035 um, using UMC CMOS 0.8um DPDM technology. The chip consumes 95mW at a 20MS/s sampling rate and the INL is less than ±1LSB. The convolution operation unit uses TSMC 0.8um SPDM technology and occupies an area of 2658um×5284um. The power dissipation is 50mW at the operating frequency of 66Mhz. The residue DAC is processed in a TSMC 0.8um SPDM technology and the die size is 2284um×2499um. Both INL and DNL are less than ±0.1LSB and the chip dissipates 50mW at the conversion speed of 50MHz.
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陳世權. "Use genetic algorithm based on CSD code to design finite impulse response digital filter". Thesis, 2004. http://ndltd.ncl.edu.tw/handle/02654749395121931986.
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Jian, Jun-Hong Zhang y 張簡峻泓. "Design of Low Cost Finite-Impulse Response (FIR) Filters Using Multiple Constant Truncated Multipliers". Thesis, 2012. http://ndltd.ncl.edu.tw/handle/17475964081023321341.
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碩士國立中山大學
資訊工程學系研究所
100
Finite impulse response (FIR) digital filters are frequently used in many digital signal processing and communication applications, such as IS-95 CDMA, Digital Mobile Phone Systems (D-AMPS), etc. FIR filter achieves the frequency response of system requirement using a series of multiplications and additions. Previous papers on FIR hardware implementations usually focus on reducing area and delay of the multiple constant multiplications (MCM) through common sub-expression elimination (CSE) in the transpose FIR filter structure. In this thesis, we first perform optimization for the quantization of FIR filter coefficients that satisfy the target frequency response. Then suitable encoding methods are adopted to reduce the height of the partial products of the MCM in the direct FIR filter structure. Finally, by jointly considering the errors in the truncated multiplications and additions, we can design the hardware-efficient FIR filter that meets the bit accuracy requirement. Experimental results show that although CSE in the transpose FIR structure can reduce more area in MCM, the direct form takes smaller area in registers. Compared with previous approaches, the proposed FIR implementations with direct form has the minimum area cost.
36
B, Mallikarjun. "Response of extended eulerbernoulli beam under impulse load using wavelet spectral finite element method". Thesis, 2012. http://ethesis.nitrkl.ac.in/3909/1/RESPONSE_OF_EXTENDED_EULER-BERNOULLI_BEAM_UNDER_IMPULSE_LOAD_USING_WAVELET_SPECTRAL_FINITE_ELEMENT_METHOD.pdf.
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Transform methods are some of those methods which are able to solve certain difficult ordinary and partial differential equation. The most commonly used transform for these solutions are Laplace and Fourier transforms. Wavelet transforms are new entrants in to this area, although
they are quite popular with electrical and communication engineers in characterizing and synthesizing the time signals. The utility of wavelet transforms is shown in structural engineering by addressing problems involving solutions of ordinary and partial differential equations
encountered in dynamical related problems.
Dynamical problems in structural engineering fall under two categories, one involving low frequencies, which is called structural dynamics problems, and the other involving very high frequencies, which is called the wave propagation problems. The most problems in structural engineering fall under the former category, wherein the response of the entire structural system is characterized using only the first few vibrational modes. The wave propagation is a multi-modal phenomenon involving vibrational modes of very high frequencies. Conventional analysis tools
such as finite element cannot handle these problems due to modeling limitations and extensive computational cost. The only alternative to such problems is the method based on transforms.Spectral finite element (SFE) method is one such transform method, which can be a viable alternative to solving problems involving high frequency excitations. SFE based on Fourier transform is quite well known and established. However, it has severe limitations in handling
finite structures and specifying non-zero boundary/initial conditions, and thus its utility in solving real world problems involving high frequency excitation is limited.
The aim of the present work is to show that the wavelet transform is very useful in solving ordinary differential equations by modeling the structure as a discrete system involving structural dynamic problems and it is to use wavelet transform to solve those problems involving partial
differential equations. In this work, the response of an cantilever Extended Euler-Bernoulli aluminum beam under impulse load applied axial and transverse at the free end is shown. The response is being obtained by coding programs in MATLAB.
37
Lin, Yi-Cheng y 林益誠. "Impulse Response of Porous Beams Coupled with Porous Structures by Finite Element Frequency-Domain Analysis". Thesis, 2012. http://ndltd.ncl.edu.tw/handle/82026455752639152884.
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碩士淡江大學
機械與機電工程學系碩士班
100
Under the assumptions of Euler beam and plane stress, this study formulates the governing equations of flexural vibrations for the porous beam and plate using Biot’s poroelastic theory. Then, the stiffness matrices of the porous beam, plate and medium elements are derived in Laplace domain. Thereafter, using the impulsive loading and the elastic boundary conditions, the finite element frequency domain analyses are performed to study the dynamic behaviors of porous beams, plates, and mediums. In order to match the application condition, the dynamic behaviors of stiffened porous plates (porous beam coupled with porous plate) and porous beam coupled with porous medium are also evaluated. The porous beam, plate, and medium present a typical dissipation effect due to the interaction between the saturated fluid and the solid skeleton. Upon examining the reduction of modal amplitudes of the saturated porous beam and plate, the dissipation effect is found growing with the increase of the fluid’s viscosity, and the bulk modulus of the fluid has major effect on their modal frequencies. Therefore, by changing of the saturated fluid, the modal frequency and amplitude of the porous beam and plate can be adjusted. From the modal frequency fluctuations of the stiffened porous plate, the increase of both dissipation coefficient and modal frequency are clearly observed with the raise of porosity. Hence, the dynamic behavior of the stiffened porous plate can be precisely adjusted by the changes of the porosity and the saturated fluid. In addition, the analysis results of the coupling of a porous beam with an acoustic field show that the coupled modal frequencies of the porous beam and the acoustic field can be simultaneously observed, as well as the remarkable changes on the modal frequencies and amplitudes.
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"THE IMPLEMENTATION COMPLEXITY OF FINITE IMPULSE RESPONSE DIGITAL FILTERS UNDER DIFFERENT COEFFICIENT QUANTIZATION SCHEMES AND REALIZATION STRUCTURES". Master's thesis, METU, 2004. http://etd.lib.metu.edu.tr/upload/2/12605586/index.pdf.
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Wang, Neng-Li y 王能立. "Performance Analysis of Dithering Techniques for Binary Frequency Shift Keying and the Design of New Finite Impulse Response Filters". Thesis, 2017. http://ndltd.ncl.edu.tw/handle/g7ykjs.
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碩士國立彰化師範大學
電信工程學研究所
106
In the first topic, we discuss whether dithering technology is effective in binary frequency shift keying (BFSK) modulation. The dithering technology in this topic is adding the appropriate amount of independent random signal on binary frequency shift keying signal in front of the analog-to-digital converter (ADC). We analyze the signal spectrum and performance of the quantized binary frequency-shift keying signal with and without using the dithering technology. We found that out-of-band signal can be effectively reduced after dithering. The price paid is an increase of bit error rate due to more quantization noise. In the second topic, we propose a finite impulse response (FIR) filter design method using cascade techniques. We cascade a number of complementary comb filter as a shaping filter and use the modified window design method to design a prototype filter. Then we cascade the two parts to form the composite filter. The composite filter has a very sharp transition band and a very deep cut-off band. It also has a linear phase response. We also design a flexible software interface for designing the composite filter. Users can follow the instructions to generate a number of different designs. This method can also be used in the design of high-pass and band-pass filters of different filter specifications.
40
Anoh, Kelvin O. O., James M. Noras, Raed A. Abd-Alhameed, Steven M. R. Jones y Konstantinos N. Voudouris. "A New Approach for Designing Orthogonal Wavelets for Multicarrier Applications". 2014. http://hdl.handle.net/10454/8668.
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yesThe Daubechies, coiflet and symlet wavelets, with properties of orthogonal wavelets are suitable for multicarrier transmission over band-limited channels. It has been shown that similar wavelets can be constructed by Lagrange approximation interpolation. In this work and using established wavelet design algorithms, it is shown that ideal filters can be approximated to construct new orthogonal wavelets. These new wavelets, in terms of BER behave slightly better than the wavelets mentioned above, and much better than biorthogonal wavelets, in multipath channels with additive white Gaussian noise (AWGN). It is shown that the construction, which uses a simple simultaneous solution to obtain the wavelet filters from the ideal filters based on established wavelet design algorithms, is simple and can easily be reproduced.
41
Lamb, Matthew. "Monitoring the structural integrity of packaging materials subjected to sustained random loads". Thesis, 2011. https://vuir.vu.edu.au/16054/.
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During the distribution phase, packaged consignments are exposed to a variety of environmental hazards, such as vibrations, which, if excessively severe, may cause damage to or even destroy the product. The ability of packaging systems to withstand these dynamic loads is an important factor, since their function is not only to protect the product but also to ensure that damage to the package itself is not evidently apparent as this is likely to affect
consumer confidence. Such concerns often result in the use of excessive packaging material and this is no longer acceptable. Therefore, engineered protective packaging systems need to be optimised. Such optimisation requires a suitable technique for continuously monitoring the
structural integrity of the systems during laboratory based vibration testing.
42
Venkataraman, S. "On Adaptive Filtering Using Delayless IFIR Structure : Analysis, Experiments And Application To Active Noise Control And Acoustic Echo Cancellation". Thesis, 2005. https://etd.iisc.ac.in/handle/2005/1436.
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Venkataraman, S. "On Adaptive Filtering Using Delayless IFIR Structure : Analysis, Experiments And Application To Active Noise Control And Acoustic Echo Cancellation". Thesis, 2005. http://etd.iisc.ernet.in/handle/2005/1436.
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Κουρέτας, Ιωάννης. "Κυκλώματα αριθμητικής υπολοίπων με χαμηλή κατανάλωση και ανοχή σε διακυμάνσεις παραμέτρων". Thesis, 2012. http://hdl.handle.net/10889/5551.
Texto completoResumen
Το αριθμητικό σύστημα υπολοίπων (RNS) έχει προταθεί ως ένας τρόπος για επιτάχυνση των αριθμητικών πράξεων του πολλαπλασιασμού και της πρόσθεσης. Ένα από τα σημαντικά πλεονεκτήματα της χρήσης του RNS είναι ότι οδηγεί σε κυκλώματα που έχουν το χαρακτηριστικό της χαμηλής κατανάλωσης.
Πιο συγκεκριμένα στην παρούσα διατριβή γίνεται μια αναλυτική μελέτη πάνω στην ταχύτητα διεξαγωγής της πράξης του πολλαπλασιασμού και της άθροισης. Ο λόγος που γίνεται αυτό είναι διότι οι εφαρμογές επεξεργασίας σήματος χρησιμοποιούν ιδιαιτέρως τις προαναφερθείσες πράξεις. Επίσης γίνεται μελέτη της ισχύος που καταναλώνεται κατά την επεξεργασία ενός σήματος με τη χρήση των προτεινόμενων αριθμητικών κυκλωμάτων. Ιδιαίτερη έμφαση δίνεται στη χρήση απλών αρχιτεκτονικών τις οποίες μπορούν τα εργαλεία σύνθεσης να διαχειριστούν καλύτερα παράγοντας βέλτιστα κυκλώματα.
Τέλος η διατριβή μελετά τα προβλήματα διακύμανσης των παραμέτρων του υλικού που αντιμετωπίζει η σύγχρονη τεχνολογία κατασκευής ολοκληρωμένων κυκλωμάτων. Συγκεκριμένα σε τεχνολογία μικρότερη των 90nm παρατηρείται το φαινόμενο ίδια στοιχεία κυκλώματος να συμπεριφέρονται με διαφορετικό τρόπο. Το φαινόμενο αυτό γίνεται σημαντικά πιο έντονο σε τεχνολογίες κάτω των 45nm. Η παρούσα διατριβή προτείνει λύσεις βασισμένες στην παραλληλία και την ανεξαρτησία των επεξεργαστικών πυρήνων που παρέχει το RNS, για να αντιμετωπίσει το συγκεκριμένο φαινόμενο.The Residue Number System (RNS) has been proposed as a means to speed up the implementation of multiplication-addition intensive applications, commonly found in DSP. The main benefit of RNS is the inherent parallelism, which has been exploited to build efficient multiply-add structures, and more recently, to design low-power systems. In particular, this dissertation deals with the delay complexity of the multiply-add operation (MAC). The reason for this is that DSP applications are MAC intensive and hence this dissertation proposes solutions to increase the speed of processing. Furthermore, the study of the multiply-add operations is extended to power consumption matters. The dissertation focus on simple architectures such that EDA tools produce efficient in both power and delay, synthesized circuits. Finally the dissertation deals with variability matters that came up as the vlsi technology shrinks below 90nm. Variability becomes unaffordable especially for the 45nm technology node. This dissertation proposes solutions based on parallelism and the independence of the RNS cores to derive variation-tolerant architectures.