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Dissertations / Theses on the topic 'Finite Impulse Filter (FIR)'

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Published: 4 June 2025

Last updated: 15 July 2025

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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.

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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.

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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 literture their 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.

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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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Submitted by JULIO CEZAR UZINSKI null (uzinski.jc@gmail.com) on 2016-12-15T21:43:22Z No. of bitstreams: 1 Uzinski JC.pdf: 2380247 bytes, checksum: 910b14a40501433136262e638e586b5f (MD5) Approved for entry into archive by Felipe Augusto Arakaki (arakaki@reitoria.unesp.br) on 2016-12-20T16:20:21Z (GMT) No. of bitstreams: 1 uzinski_jc_dr_ilha.pdf: 2380247 bytes, checksum: 910b14a40501433136262e638e586b5f (MD5) Made available in DSpace on 2016-12-20T16:20:21Z (GMT). No. of bitstreams: 1 uzinski_jc_dr_ilha.pdf: 2380247 bytes, checksum: 910b14a40501433136262e638e586b5f (MD5) Previous issue date: 2016-11-25 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 Alvarado Resumo: 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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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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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 Thesis Window 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.

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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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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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Húserka, Jozef. "Návrh reverberátoru pro simulaci akustiky prostoru." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2014. http://www.nusl.cz/ntk/nusl-220649.

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This thesis deals with artificial simulation of acoustic spaces by using reverberators. Output of this document consists of four reverberation algorithms and function that evaluates objective parameters of acoustic space from impulse responses. Reverberators and script were implemented using Matlab. Graphical user interface is used to present all of the algorithms for easier usability. First chapter deals with objective parameters of acoustic spaces and the ways they are computed from impulse response. Second chapter describes various structures which are used to build reverberators. Those structures are used in third chapter in implementations of reverberators. Third chapter also compares all implemented reverberators . In last chapter experiment was made. Impulse responses of three spaces were measured and subsequently aproximated by algorithms implemented in this thesis.

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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.

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Srinivas, L. "FIR System Identification Using Higher Order Cumulants -A Generalized Approach." Thesis, Indian Institute of Science, 1994. http://hdl.handle.net/2005/637.

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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.

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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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Sklar, Alexander Gabriel. "Channel Modeling Applied to Robust Automatic Speech Recognition." Scholarly Repository, 2007. http://scholarlyrepository.miami.edu/oa_theses/87.

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In automatic speech recognition systems (ASRs), training is a critical phase to the system?s success. Communication media, either analog (such as analog landline phones) or digital (VoIP) distort the speaker?s speech signal often in very complex ways: linear distortion occurs in all channels, either in the magnitude or phase spectrum. Non-linear but time-invariant distortion will always appear in all real systems. In digital systems we also have network effects which will produce packet losses and delays and repeated packets. Finally, one cannot really assert what path a signal will take, and so having error or distortion in between is almost a certainty. The channel introduces an acoustical mismatch between the speaker's signal and the trained data in the ASR, which results in poor recognition performance. The approach so far, has been to try to undo the havoc produced by the channels, i.e. compensate for the channel's behavior. In this thesis, we try to characterize the effects of different transmission media and use that as an inexpensive and repeatable way to train ASR systems.

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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

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Jian, Jun-Hong Zhang, and 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.

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Anoh, Kelvin O. O., James M. Noras, Raed A. Abd-Alhameed, Steven M. R. Jones, and Konstantinos N. Voudouris. "A New Approach for Designing Orthogonal Wavelets for Multicarrier Applications." 2014. http://hdl.handle.net/10454/8668.

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yes The 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.

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JIANG, KAI-REN, and 姜凱仁. "Research on Improving Room Impulse Response Using FIR Filter." Thesis, 2018. http://ndltd.ncl.edu.tw/handle/b3cnsp.

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碩士 國立臺北大學 通訊工程研究所 106 Now a day in our life, we often use speakers or others equipment to play music. But when we play music, the sound is often affected by disturbance from the environment or other sources, causing us to hear imperfect sounds. The purpose of this experiment is to use FR filter from the FIR system to equalize the sound so our ears can hear the same sound as what the speaker plays. First, we record and save the sounds from different frequencies onto the computer. Secondly, we use MATLAB to convert the recorded and undisturbed sounds from the impulse response to the frequency domain and averaged all the values. Then, we will get the gain value form each frequency. We design FIR filters that can equalize all gain values. Input the designed FIR filter coefficients to the miniDSP 2x4 hd. MiniDSP 2x4 hd will design the FIR filter that we want. Through the miniDSP 2x4 hd the sound that we heard will be close to the undisturbed sound.

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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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Chung, Pwu-Yueh, and 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, and 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, and 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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Yedlapalli, Satya Sudhakar. "The Line Spectral Frequency Model Of A Finite Length Sequence And Its Applications." Thesis, 2010. https://etd.iisc.ac.in/handle/2005/1262.

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Yedlapalli, Satya Sudhakar. "The Line Spectral Frequency Model Of A Finite Length Sequence And Its Applications." Thesis, 2010. http://etd.iisc.ernet.in/handle/2005/1262.

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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. https://etd.iisc.ac.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.

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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.

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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.

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