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England, Janine V. "Digital filter design techniques/." Thesis, Monterey, California. Naval Postgraduate School, 1988. http://hdl.handle.net/10945/23177.
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An overview and investigation of the more popular digital filter design techniques are presented, with the intent of providing the filter design engineer a complete and concise source of information. Advantages and disadvantages of the various techniques are discussed, and extensive design examples used to illustrate their application to specific design problems. Both IIR (Butterworth, Chebyshev and elliptic), and FIR (Fourier coefficient design, windows and frequency sampling) design methods are featured, as well as, the Optimum FIR Filter Design Program of Parks and McClellan, and the Minimum p - Error IIR Filter Design Method of Deczky. Keywords: Digital filter design, IIR, FIR, Butterworth, Chebyshev, Elliptic, Fourier coefficient, Windows, Frequency sampling, Remez exchange algorithm, Minimum p-error, and IRR filter design
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Kennedy, Paul B. (Paul Brodie). "Filter designer : an intuitive digital filter design environment." Thesis, McGill University, 1996. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=23849.
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The ability to accurately manipulate the spectral content of audio is indispensable for the artistic presentation of sound. While there are many devices currently available that are capable of performing this function, most tend to be either highly complex electrical engineering tools, or music-oriented products that are limited in functionality. Filter Designer was created to fill this gap by providing an environment with which musically-trained users can design and implement digital filters, while having access to control parameters and analysis data previously exclusive to the engineering field. This work explores the factors that motivated the creation of Filter Designer, and examines the process of its development, from basic user interface design to the calculation and implementation of digital filters for use with audio signals.
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Anderson, Martin S. "Design of two-dimensional PCAS digital filters and filter banks." Thesis, University of Warwick, 1994. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.307968.
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Law, Ying Man. "Iterative algorithms for the constrained design of filters and filter banks /." View abstract or full-text, 2004. http://library.ust.hk/cgi/db/thesis.pl?ELEC%202004%20LAW.
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Thesis (M. Phil.)--Hong Kong University of Science and Technology, 2004.Includes bibliographical references (leaves 108-111). Also available in electronic version. Access restricted to campus users.
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Penberthy, Harris Stephen. "Natural algorithms in digital filter design." Thesis, University of Plymouth, 2001. http://hdl.handle.net/10026.1/2752.
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Digital filters are an important part of Digital Signal Processing (DSP), which plays vital roles within the modern world, but their design is a complex task requiring a great deal of specialised knowledge. An analysis of this design process is presented, which identifies opportunities for the application of optimisation. The Genetic Algorithm (GA) and Simulated Annealing are problem-independent and increasingly popular optimisation techniques. They do not require detailed prior knowledge of the nature of a problem, and are unaffected by a discontinuous search space, unlike traditional methods such as calculus and hill-climbing. Potential applications of these techniques to the filter design process are discussed, and presented with practical results. Investigations into the design of Frequency Sampling (FS) Finite Impulse Response (FIR) filters using a hybrid GA/hill-climber proved especially successful, improving on published results. An analysis of the search space for FS filters provided useful information on the performance of the optimisation technique. The ability of the GA to trade off a filter's performance with respect to several design criteria simultaneously, without intervention by the designer, is also investigated. Methods of simplifying the design process by using this technique are presented, together with an analysis of the difficulty of the non-linear FIR filter design problem from a GA perspective. This gave an insight into the fundamental nature of the optimisation problem, and also suggested future improvements. The results gained from these investigations allowed the framework for a potential 'intelligent' filter design system to be proposed, in which embedded expert knowledge, Artificial Intelligence techniques and traditional design methods work together. This could deliver a single tool capable of designing a wide range of filters with minimal human intervention, and of proposing solutions to incomplete problems. It could also provide the basis for the development of tools for other areas of DSP system design.
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Fakhry, Nader. "Design of a Digital Compensation Filter." PDXScholar, 1995. https://pdxscholar.library.pdx.edu/open_access_etds/4961.
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The 24-bit Motorola DSP56001 processor will be used in combination with the DSP56ADC16 and the PCM-56 to design a good FIR compensation filter. Our objective is to digitize the input analog signal, and to compensate for the attenuation in the magnitude response of the digital sine wave. Two different experiments will be conducted, a hands on approach, and a simulation program. The first one will be realized directly, using the DSP system. We will determine the magnitude response of the system, and then deduce the coefficients of the FIR sin(x)/x filter. A look up table will store those values which will be fetched by the DSP program. With a minimum set of instructions we will generate a new digital output sequence after a N-point circular convolution is performed. The output signal is a good reconstruction of the input signal at frequencies below 22 Khz. However, a second experiment will be needed to improve this FIR sin(x)/x compensation filter, because we are not able to go beyond a 300-point impulse sequence. After that value (300-point), the time that each value is read and is ready to be processed by the DSP56001 becomes smaller than the time each instruction in the DSP program is executed and written to the PCM-56 via the SSI register. To be able to expand our experiment, we need to write a simulation program. A simulation program of the previous experiment, which take as input the measured magnitude response of the system. The challenge will be to find ways to map the frequency domain, by using the maximum value of each linear convolution sequence, with a finite input sequence. A step by step approach will be drawn until our final objective is reached. Our final step will be, to increase the number of sampling point in the frequency domain and will be to demonstrate that the result of the simulated program value will coincide with our objective, which is to compensate for the attenuation of the magnitude response of the system. By increasing the sampling frequency we will eventually obtain a good compensation filter.
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Lindblom, Ludvig. "Design of a Digital Octave Band Filter." Thesis, Linköpings universitet, Elektroniksystem, 2012. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-79231.
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This report describes the design and implementation of a fixed audio equalizer based on a scheme where parts of the signal spectrum are downsampled and treated differently for the purpose of reducing the computational complexity and memory requirements. The primary focus has been on finding a way of taking an equalizer based on a simple minimum-phase FIR filter and transform it to the new type of equalizer. To achieve this, a number of undesireable effects such as aliasing distortion and upsampling imaging had to be considered and dealt with. In order to achieve a good amplitude response of the system, optimization procedures were used. As part of the thesis, a cost-effective implementation of the filter has been made for an FPGA, in order to verify that the scheme is indeed usable for equalizing an audio signal.
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Dempster, Andrew. "Digital filter design for low-complexity implementation." Thesis, University of Cambridge, 1995. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.362967.
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Asavathiratham, Chalee. "Digital audio filter design using frequency transformations." Thesis, Massachusetts Institute of Technology, 1996. http://hdl.handle.net/1721.1/39065.
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Thesis (M. Eng.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 1996.Includes bibliographical references (leaves 80-81).
by Chalee Asavathiratham.
M.Eng.
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Tatangsurja, Hendra. "Digital filter design using stored product ROMs." Thesis, University of Ottawa (Canada), 1987. http://hdl.handle.net/10393/5332.
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Kumar, Bhunesh. "Design of Harmonic Filters for Renewable Energy Applications." Thesis, Högskolan på Gotland, Institutionen för kultur, energi och miljö, 2011. http://urn.kb.se/resolve?urn=urn:nbn:se:hgo:diva-1862.
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Harmonics are created by non-linear devices connected to the power system. Power system harmonics are multiples of the fundamental power system frequency and these harmonic frequencies can create distorted voltages and currents. Distortion of voltages and currents can affect the power system adversely causing power quality problems. Therefore, estimation of harmonics is of high importance for efficiency of the power system network. The problem of harmonic loss evaluation is of growing importance for renewable power system industry by impacting the operating costs and the useful life of the system components. Non-linear devices such as power electronics converters can inject harmonics alternating currents (AC) in the electrical power system. The number of sensitive loads that require ideal sinusoidal supply voltage for their proper operation has been increasing. To maintain the quality limits proposed by standards to protect the sensitive loads, it is necessary to include some form of filtering device to the power system. Harmonics also increases overall reactive power demanded by equivalent load. Filters have been devised to achieve an optimal control strategy for harmonic alleviation problems. To achieve an acceptable distortion, increase the power quality and to reduce the harmonics hence several three phase filter banks are used and connected in parallel. In this thesis, high order harmonics cases have been suppressed by employing variants of Butterworth, Chebyshev and Cauer filters. MATLAB/SIMULINK wind farm model was used to generate and analyze the different harmonics magnitude and frequency. High voltage direct current (HVDC) lines for an electrical grid that is more than50km far away wind farm generation plant was investigated for harmonics. These HVDC lines are also used in offshore wind farm plant. Investigated three-phase harmonics filters are shunt elements that are used in power systems for decreasing voltage distortion and for correcting the power factor. Renewable energy sources are not the stable source of energy generation like wind, solar and tidal e.t.c. Though they are secondary sources of generation and hard to connect with electrical grid. In near future the technique is to use the wave digital filter (WDF) or circulator-tree wave digital filter (CTWDF) for the renewable energy application can be employed to mitigate the harmonics. These WDF and CTWDF can b eused in HVDC lines and smart grid applications. A preliminary analysis is conducted for such a study.
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McAllister, Christine Joan. "Automated design of high performance digital filter chips." Thesis, Queen's University Belfast, 1996. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.318797.
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Krukowski, Artur. "Flexible IIR digital filter design and multipath realisation." Thesis, University of Westminster, 1999. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.322993.
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Groff, Wayne C. "Predetection filter design for practical digital communication systems." Thesis, University of Ottawa (Canada), 1989. http://hdl.handle.net/10393/5585.
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Rossi, Michel. "Iterative least squares algorithms for digital filter design." Thesis, University of Ottawa (Canada), 1996. http://hdl.handle.net/10393/10099.
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In this thesis, we propose new algorithms to simplify and improve the design of IIR digital filters and M-band cosine modulated filter banks. These algorithms are based on the Iterative Least Squares (ILS) approach. We first review the various Iterative Reweighted Least Squares (IRLS) methods used to design Chebyshev and $L\sb{p}$ linear phase FIR filters. Then we focus on the ILS design of IIR filters and filter banks. For the design of Chebyshev IIR filters in the log magnitude sense, we propose a Remez-type IRLS algorithm. This novel approach accelerates significantly Kobayashi's and Lim's IRLS methods and simplifies the traditional rational Remez algorithm. For the design of M-band cosine modulated filter banks, we propose three new ILS algorithms. These algorithms are specific to the design of Pseudo Quadrature Mirror Filter (QMF) banks, Near Perfect Reconstruction (NPR) Pseudo QMF banks and Perfect Reconstruction (PR) QMF banks. They are fast convergent, simple to implement and flexible compared to traditional nonlinear optimization methods. Short MATLAB programs implementing the proposed algorithms are included.
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Li, Wenmin. "A framework for digital filter design based on rootmovements." Thesis, Imperial College London, 2005. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.417503.
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Widmann, Andreas, Erich Schröger, and Burkhard Maess. "Digital filter design for electrophysiological data: a practical approach." Elsevier, 2015. https://ul.qucosa.de/id/qucosa%3A32715.
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Background: Filtering is a ubiquitous step in the preprocessing of electroencephalographic (EEG) and magnetoencephalographic (MEG) data. Besides the intended effect of the attenuation of signal components considered as noise, filtering can also result in various unintended adverse filter effects (distortions such as smoothing) and filter artifacts. Method: We give some practical guidelines for the evaluation of filter responses (impulse and frequency response) and the selection of filter types (high-pass/low-pass/band-pass/band stop; finite/infinite impulse response, FIR/IIR) and filter parameters (cutoff frequencies, filter order and roll-off, ripple, delay and causality) to optimize signal to-noise ratio and avoid or reduce signal distortions for selected electrophysiological applications. Results: Various filter implementations in common electrophysiology software packages are introduced and discussed. Resulting filter responses are compared and evaluated. Conclusion: We present strategies for recognizing common adverse filter effects and filter artifacts and demonstrate them in practical examples. Best practices and recommendations for the selection and reporting of filter parameters, limitations, and alternatives to filtering are discussed.
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Petrie, Neil. "The design and implementation of digital wave filter adaptors." Thesis, University of Edinburgh, 1985. http://hdl.handle.net/1842/15641.
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Tsui, Kai-man, and 徐啟民. "New design methods for perfect reconstruction filter banks." Thesis, The University of Hong Kong (Pokfulam, Hong Kong), 2004. http://hub.hku.hk/bib/B30144991.
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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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Becker, Kenneth Alan. "The effects of spectral estimation on matched filter design." Thesis, Virginia Polytechnic Institute and State University, 1985. http://hdl.handle.net/10919/90911.
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Moving-average matched filters (MAMF's) are a class of digital filters used to detect the presence of a known signal in noise. Designing matched filters requires knowledge of the structure of the signal and the noise. If the spectral density of the noise is not known or is changing with time its spectral characteristics must be estimated. Since spectral estimators derive their estimates from a random process realization, the estimates themselves are probabilistic in nature. The performance of MAMF's based on these estimates must, in turn, be distributed in a probabilistic sense.
This thesis investigates the performance of MAMF's designed on the basis of several different spectral estimators. Theoretical aspects of MAMF's and spectral estimators are reviewed and developed. A simulation system is used to exercise the spectral estimators and MAMF's and to provide comparative performance data. A graphical representation, using contour plots, is developed and can be used to predict the performance of a given MAMF/signal/spectral estimator combination.
Finally, several methods of generating MAMF's whose output performance is relatively insensitive (or robust) to the probabilistic variations caused by the spectral estimators are developed and evaluated. The latter incorporates knowledge of the empirical distribution of the particular spectral estimator used, as well as the freedom of manipulating the signal.M.S.
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Critchley, J. "Analysis and design of periodically time varying digital filters." Thesis, University of Cambridge, 1986. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.383172.
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Gardos, Thomas R. "Analysis and design of multidimensional FIR filter banks." Diss., Georgia Institute of Technology, 1993. http://hdl.handle.net/1853/15621.
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Cook, Anthony John. "Digital image processing using colour space transformation." Thesis, University of Hertfordshire, 2000. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.323433.
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The purpose of the work is to explore the feasibility of devising a computer system that implements the desirable effects of a photographic filter and provides an environment for colour filter design for image processing. Using conversion from RGB to the CIELUV colour space a new method for the implementation of photographic filter as a digital filter is described. A filter is implemented by converting image pixel rgb values into CIELUV (u', v') and L* values and operates using the visual wavelength values provided by the (u', v') chromaticity diagram. However, the (u', v') diagram cannot provide wavelength values for pixels that correspond to (u', v') points in the `purple line' sector of the diagram. These pixels are allocated wavelengths by means of a new wavelengths scale that makes it possible for the filter to process any pixel in a digital image. Filter transmittance data for visual spectrum wavelengths is obtained from published tables. The transmittance data for purple sector pixels is provided by a colour model of the (u', v') chromaticity diagram. The system is evaluated by means of the Macbeth ColorChecker chart and the use of physical measurements. The extension of the CIELUV diagram with an equivalent wavelength scale provides a new environment for the enhancement and manipulation of digital colour images.
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Nordebo, Sven. "Robust broadband beamforming and digital filter design : methods and applications." Doctoral thesis, Luleå tekniska universitet, 1995. http://urn.kb.se/resolve?urn=urn:nbn:se:ltu:diva-25667.
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This doctoral thesis consists of a summary and six parts corresponding to six different papers. There are two published and three submitted journal papers, and one research report. The summary will highlight the main results and emphasize the interrelationships between different parts. The thesis comprises two main themes: robust broadband beamforming and digital filter design. In most of the papers in this thesis these concepts are closely related. Some of the results on digital filter design actually concern robust filter design, and one of the main contributions of this thesis concerns the possibility of anomalous designs when using conventional methods in the FIR filter design of broadband beamformers. Part I deals with robustness of broadband adaptive beamformers in the sense that there is an uncertainty in the frequency content and spatial location of the desired target signal. The adaptive beamformer should perform well over a region in space and frequency. This is achieved by defining the Spatial Filter designed Generalized Sidelobe Canceller (SFGSC), and by using an appropriate digital filter design. The novelty of the SFGSC method is the implementation of target signal constraints by using a filter design approach. With this strategy the constraints are approximated over a given design domain, rather than exactly implemented as with the conventional Generalized Sidelobe Canceller (GSC). With this new method (in contrast to the GSC) the SFGSC is able to handle a continuum of constraints. Part II introduces a quadratic programming formulation of the weighted Chebyshev FIR filter design problem for a broadband beamformer in the near-field. This technique may be used to design the digital filters of the broadband SFGSC described in Part I. Part III reveals that the digital filters of a broadband beamformer are incompletely specified whenever the beamformer is specified only in space and frequency. Using conventional filter design criteria with such incomplete specifications may lead to excessively large filter coefficients. Robust weighted least squares and weighted Chebyshev design criteria are introduced in order to avoid this anomaly. "Robustness" in this context means insensitivity to model imperfections such as sensor element placement errors, amplifier mismatch, etc. Again, the filters designed are typically components of the SFGSC described in Part I. Part IV addresses the problem of the non-uniqueness of the Chebyshev approximation for two-dimensional linear phase digital FIR filters. It is shown that the unique Chebyshev approximation having minimum Euclidean filter weight norm can be obtained by using a wellconditioned quadratic programming formulation. This is the same quadratic program that was used to define one of the robust beamformer designs in Part III. Part V deals with robust design for one-dimensional non-linear phase FIR filters which are incompletely specified. The conventional weighted Chebyshev solution can be obtained by using a quadratic programming formulation similar to that given in Part II. A robust weighted Chebyshev design criterion is defined by a modified quadratic program, similar to one of the robust design methods given in Part III. Part VI emphasizes the robustness of adaptive beamformers with respect to channel mismatch, sensor positioning, etc. In particular, the paper addresses the difficulty of mathematically modeling a beamformer for a small enclosure such as a car compartment. A calibrating scheme is proposed which is independent of array geometry and channel matching, and which calibrates the adaptive array to the given acoustic environment and to the given electronic equipment. Results from real measurements in a car compartment are included. An international patent is applied for based on this paper.Godkänd; 1995; 20070426 (ysko)
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Jackson, Brian Aliston. "Digital Filter Design and Synthesis Using High-level Modeling Tools." Thesis, Virginia Tech, 1999. http://hdl.handle.net/10919/35939.
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The purpose of this thesis is to formulate a technically sound approach to designing Infinite Impulse Response (IIR) digital filters using high-level modeling tools. High-level modeling tools provide the ability to build and simulate ideal models. Once proper validation is complete on these ideal models, the user can then migrate to lower levels of abstraction until an actual real world model is designed. High-level modeling tools are the epitome of the top-down design concept in which design first takes place with the basic functional knowledge of a system. With each level of abstraction, validation is performed. High-level modeling tools are used throughout industry and their application is continually growing especially in the DSP area where many modes of communications are expanding. High-level modeling tools and validation significantly address this complex expansion by utilizing an ideal representation of a complicated network.Master of Science
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Zhang, Genzao. "Non-uniform-band digital filter banks: Design and performance analysis." Thesis, University of Ottawa (Canada), 1992. http://hdl.handle.net/10393/7841.
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New design techniques of digital filter banks are presented in this thesis. The research has been focused on systematic design methods for computationally efficient filter banks with arbitrary center frequencies, which could be easily implemented and are useful in acoustic, speech signal processing, and communication systems. Frequency interpolation filter banks (FIFB) are first derived based on an adaptive filtering structure. FIFB banks show good frequency responses, interesting sensitivities and negligible parameter quantization effects and roundoff noise when they are implemented using modern DSP processors with fixed-point arithmetic. For maximally flat FIFB filter banks, the adjusted pole selection strategy by an optimization procedure is applied. Then, the resonator-based arbitrarily spaced center frequency filter banks (RFB) are proposed. They show better behaviour than general FIFB banks from the point of view of frequency responses and the implementation. In order to suppress the non-negligible sidelobes and further improve the passband performance of the non-uniform band filter banks, frequency domain windowing techniques are studied and linear programming is used to produce optimal windows for different banks. Thus the design methods of non-uniform band filter banks with windowing processing are presented and significant improvements are obtained on the filter bank performance. Finally, the FIFB filter bank techniques are successfully extended to the field of allpass and perfect reconstruction analysis/synthesis systems. This widens the application area of the FIFB filter banks. In an allpass system, a synthesis FIFB filter bank is derived based on a general analysis FIFB bank and the allpass requirement. In the perfect reconstruction system, multirate techniques are applied and the efficient general FIR synthesis filter bank and IIR FIFB filter bank are designed to construct the system.
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Krishnan, Ashok, and Kurt Kosbar. "AN INVESTIGATION INTO USING EXPERT SYSTEMS FOR DIGITAL FILTER DESIGN." International Foundation for Telemetering, 2003. http://hdl.handle.net/10150/605359.
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International Telemetering Conference Proceedings / October 20-23, 2003 / Riviera Hotel and Convention Center, Las Vegas, NevadaIt can be challenging to select the best architecture for DSP filters for a given application. Design constraints often include both objective and subjective information. This paper discusses the initial results of an investigation into using expert system techniques to address this problem. The goal is a system that allows users to specify traditional constraints such as impulse response, frequency response, stability, SNR, etc., but they may also constrain the filter’s cost, complexity, or any parameter which can be clearly identified for the specific application.
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Xie, Xuemei, and 謝雪梅. "New design and realization methods for perfect reconstruction nonuniform filter banks." Thesis, The University of Hong Kong (Pokfulam, Hong Kong), 2004. http://hub.hku.hk/bib/B31246175.
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Kwan, Hing-kit. "Design algorithms for delta-sigma modulator loop filter topologies." Click to view the E-thesis via HKUTO, 2008. http://sunzi.lib.hku.hk/hkuto/record/B4150883X.
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Sodagar, Iraj. "Analysis and design of time-varying filter banks." Diss., Georgia Institute of Technology, 1994. http://hdl.handle.net/1853/13437.
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Rosler, Lucas Owen. "Design and Analysis of an FPGA Based Low Tap Band-stop FIR Filter." Youngstown State University / OhioLINK, 2021. http://rave.ohiolink.edu/etdc/view?acc_num=ysu1619798270047225.
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Sundaralingam, Sathiaseelan. "Evolving optimal IIR and adaptive filters." Thesis, University of Glasgow, 1999. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.300977.
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Loce, Robert P. "Morphological filter mean-absolute-error representation theorems and their application to optimal morphological filter design /." Online version of thesis, 1993. http://hdl.handle.net/1850/11065.
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Ng, Chee We 1975. "Design of a power-scalable digital least-means-square adaptive filter." Thesis, Massachusetts Institute of Technology, 2001. http://hdl.handle.net/1721.1/87168.
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Li, Min. "Induced norm optimal multirate filter bank design using LMI constraints /." View Abstract or Full-Text, 2002. http://library.ust.hk/cgi/db/thesis.pl?ELEC%202002%20LI.
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Thesis (M. Phil.)--Hong Kong University of Science and Technology, 2002.Includes bibliographical references (leaves 55-58). Also available in electronic version. Access restricted to campus users.
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Kwan, Hing-kit, and 關興杰. "Design algorithms for delta-sigma modulator loop filter topologies." Thesis, The University of Hong Kong (Pokfulam, Hong Kong), 2008. http://hub.hku.hk/bib/B4150883X.
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Yin, Shishu, and 殷仕淑. "New design and factorization methods for perfect reconstruction causalstable IIR filter banks." Thesis, The University of Hong Kong (Pokfulam, Hong Kong), 2006. http://hub.hku.hk/bib/B38674592.
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Chen, Charng-Kann, and 陳常侃. "Optimal Design of Digital Filters and Digital Filter Banks." Thesis, 1994. http://ndltd.ncl.edu.tw/handle/60246289762306954645.
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博士國立臺灣大學
電機工程研究所
82
The work of this dissertation is to devise novel and efficient techniques for optimally designing one-dimensional (1-D) digital filters, two-dimensional (2-D) digital filters, 1-D multirate filter banks, and 2-D multirate filter banks in the appropriate optimal sense. As for the 1-D digital filters, the proposed techniques include the design algorithms for designing quasi-equiripple FIR and IIR digital filters, discrete coefficient FIR digital filters with arbitrary amplitude and phase response, digital all-pass filters, high order digital differentiators using $L_{1}$ error criteria, sharp FIR digital filters with prescribed group delay phase response, and cascade form FIR digital filters with powers-of-two coefficients in the complex domain, and Chebyshev design of IIR digital filters with arbitrary magnitude and phase responses. On the other hand, this dissertation also presents two approaches for equiripple design of 2-D linear-phase FIR digital filters. One approach is based on a novel minimax design method and the other approach is the McClellan transform based design techniques. With regard to the 1-D filter bank systems, a deep study of designing quadrature mirror filters (QMF) with linear phase in the frequency domain using different optimal criteria is given. The powers-of-two coefficient design of QMF bank is considered, too. Other than the uniform-division QMF banks, the design of two-channel nonuniform-division maximally decimated filter banks is thoroughly studied. Considering the design of 2-D multirate filter bank systems, a novel minimax design of 2-D nonseparable QMF banks with non-diagonal decimation/ expansion matrix is presented. It is shown that the quincunx QMF bank and the parallelogram QMF bank can be easily designed using the method.
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WU, YI-LONG, and 吳儀隆. "Sequency digital filter design." Thesis, 1988. http://ndltd.ncl.edu.tw/handle/04446159270564390850.
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Chung, Wei-Han, and 鍾威漢. "Optimal Design of Digital Filters and Digital Filter Banks with Continuous Coefficients." Thesis, 2001. http://ndltd.ncl.edu.tw/handle/29622534109707760244.
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碩士國立臺灣大學
電信工程學研究所
89
In this thesis,we deal with the problem of designing FIR digital filters ,FIR and IIR filter banks with low-delay property. The design techniques based on modified primal-affine scaling algorithm and modified dual-affine scaling algorithm,in conjunc-tion with approximation schemes,are then developed for solving the resulting nonlinear optimization. For FIR digital filters,we using minimax criteria to formul-ate our design problem. With regard to the FIR and IIR digital filter banks,we using criteria to formulate our design problem,and some simulation results are provided for illustration and comparision. To compare with the L2 design in[26][27],we find the L1 design we proposed have the satisfactory design results.
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Shih-Ken, Yang, and 楊世任. "Minimax Design of Digital Filters and Perfect-Reconstructioo Filter Banks." Thesis, 1998. http://ndltd.ncl.edu.tw/handle/40541120838292130400.
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博士國立臺灣科技大學
電子工程技術研究所
86
This dissertation presents several novel and efficient techniques for optimally designing one-dimensional (1-D) perfect-reconstruction (PR) filter banks, two-dimensional (2-D) FIR digital filters, and 2-D perfect- reconstruction and near-perfect-reconstruction parallelogram filter banks in the minimax senses. The proposed approaches are developed based on the affine and dual affine scaling variants of Karmarkar''s algorithm. As for the 1-D perfect-reconstruction digital filter banks, two novel techniques are proposed for designing PR filter banks with FIR analysis and synthesis filters having linear phase responses as well as low delay characteristics. The designed analysis and synthesis filters for both cases are optimal in the minimax sense subject to the perfect-reconstruction constraints. With regard to the design of 2-D digital filters, we propose design techniques for continuous and powers-of-two coefficients 2-D digital filters based on the minimax sense. The optimal continuous coefficient filters are first designed by an affine scaling variant of Karmarkar''s algorithm. Then a suboptimal powers-of-two coefficient filter is obtained by an efficient method from the optimal continuous filter coefficients. The design of 2-D parallelogram filter banks is also studied thoroughly. The linear-phase FIR analysis and synthesis optimal in minimax sense are considered. Two novel techniques for designing perfect-reconstruction and near-perfect-reconstruction 2-D parallelogram filter banks are presented. From the simulation examples demonstrated in each chapter of this dissertation, the effectiveness of the proposed design techniques for each considered design problem can be confirmed.
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Van, Lan-Da, and 范倫達. "Design of Efficient VLSI Architectures:Multiplier, 2-D Digital Filter, and Adaptive Digital Filter." Thesis, 2001. http://ndltd.ncl.edu.tw/handle/21849845445622361753.
Full textAbstract:
博士國立臺灣大學
電機工程學研究所
89
In this dissertation, we propose several efficient very-large-scale-integration (VLSI) architectures and algorithms including fixed-width multipliers, two-dimensional (2-D) digital filter, and delay least-mean-square (DLMS)-based as well as recursive-least-squares (RLS)-based adaptive digital filters. First, a general methodology for designing lower-error area-efficient fixed-width two’s-complement multipliers that receive two -bit numbers and produce an -bit product is proposed. While keeping different columns in the subproduct array, we propose several better and realizable error-compensation biases to reduce truncation error by properly choosing the proposed binary thresholding and generalized indices. Therefore, several lower-error area-efficient fixed-width multipliers suitable for VLSI implementation can be obtained. Furthermore, these new multipliers with better error-compensation circuits are suited to the fractional multiplication through a scaling box. Second, 2-D systolic-array infinite-impulse-response (IIR) and finite-impulse-response (FIR) digital filter architectures without global broadcast by the hybrid of a modified reordering scheme and a new systolic transformation are presented. This architecture possesses local broadcast, lower quantization error and zero latency without sacrificing the number of multipliers as well as delay elements under the satisfactory critical period. In addition, we extend this new architecture to a useful 2-D systolic cascade-form architecture and provide the comprehensive error analysis for the proposed architectures. Third, we propose an efficient systolic architecture for the DLMS adaptive FIR digital filter based on a new tree-systolic processing element ( ) and an optimized tree-level rule. Applying our tree-systolic , a higher convergence rate than that of the conventional DLMS structures can be obtained without sacrificing the properties of the systolic-array architecture. The efficient systolic adaptive FIR digital filter not only operates at the highest throughput in the word-level but also considers finite driving/update of the feedback error signal. Furthermore, based on our proposed optimized tree-level rule that takes account of minimum delay and high regularity, an efficient N-tap systolic adaptive FIR digital filter can be easily determined under the constraint of maximum driving of the feedback error signal. Finally, we focus on developing a new relaxed Givens rotation (RGR)-RLS algorithm to pipeline the RGR-RLS systolic array. The novel systolic adaptive architecture has faster convergence rate than that of the least-mean-square (LMS) and the DLMS-based adaptive filters. On the other hand, an arbitrarily high throughput due to the fine-grain pipelining and square-root free computation can be achieved.
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Zhou, Bo. "High speed digital FIR filter design." Thesis, 1996. http://hdl.handle.net/1957/34326.
Full textAbstract:
The objective of this thesis is to design a high speed digital FIR filter. The inputs of the
system come from a Delta-Sigma modulator. This FIR filter takes 1024 inputs,
multiplies them with their coefficients and adds the results. The main design task is to
take the input data, which are unweighted single-bit binary numbers at 156MHz,
multiply each bit with the corresponding coefficient and add them to get a weighted
multi-bit output at 20MHz.Graduation date: 1997
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Selesnick, Ivan William. "New techniques for digital filter design." Thesis, 1996. http://hdl.handle.net/1911/16955.
Full textAbstract:
Several new techniques for open problems in the frequency domain design of FIR and IIR digital filters, both iterative and analytic, are presented.
This thesis begins by putting forth the notion that explicitly specified transition bands have been introduced in the literature in part as an indirect approach for dealing with discontinuities in the desired frequency response. To overcome this, a rapidly converging algorithm is presented for the design of symmetric FIR filters according to a square error criterion that does not require specified transition bands. It does not exclude from the integral square error a region around the cut-off frequency, and yet, it overcomes Gibbs' phenomenon without resorting to windowing or 'smoothing out' the discontinuity of the ideal lowpass filter.
Also presented are algorithms for symmetric FIR filter design that modify the Parks-McClellan algorithm and a variation due to Vaidyanathan, to give a fairly complete set of design techniques for the design equiripple symmetric FIR filters.
Two types of filters, which are particularly well suited for both the approximation and implementation problems, are frequently overlooked because their design is substantially more difficult than the design of symmetric FIR and classical IIR filters. These two filter types are (1) non-symmetric FIR filters, and (2) IIR filters for which the numerator and denominator degrees need not be equal. For maximally-flat lowpass responses, analytic techniques for these two filter types are presented.
For non-symmetric FIR filter design, in which the magnitude and group delay are regarded separately (a classic problem in the design of both digital and analogue filters), it is shown that the delay can be reduced significantly while maintaining a very constant passband group delay, with no degradation in the magnitude response.
For IIR filters for which the numerator and denominator degrees are unequal, techniques for the design of generalized Butterworth and Chebyshev filters are presented.
This thesis also presents a technique to make more practical the rational Remez exchange algorithm. Lastly, a problem examined by Souto is revisited, and the use of Grobner bases from computational algebraic geometry for this problem is described.
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Chou, Yang-chou, and 周泱州. "Optimized Design of Digital FIR Filter." Thesis, 2010. http://ndltd.ncl.edu.tw/handle/93751870927545455847.
Full textAbstract:
碩士國立高雄第一科技大學
電子與資訊工程研究所
98
In this dissertation, we propose an area-efficient finite impulse response (FIR) digital filter design by using two Method-s for optimizing the coefficient and architecture of filter. The first Method- optimizes the coefficients of FIR by applying the common sub-expression elimination (CSE) algorithm, and the second Method- designs a multiplexer-based architecture in filter. We propose eight CSE algorithms and utilize them to optimize the expressions represented by different data formats including binary, canonic sign digit (CSD), and specified one. The performance comparison of CSE algorithms is based on the optimization ratio of the coefficients in a 61-tap high pass filter. The 61-tap filter is realized by 8- and 16-bit architectures for different requirements of output signals. We also realize these two architectures for comparing the performance of eight CSE algorithms. In 8-bit filter, we observe that our CSE Method--4 achieves better performance compared with others. But in 16-bit filter, our CSE Method--5 has better optimization ratio than others. An expression of the filter consists of constant multiplications and their additions. The proposed two multiplexer-based architectures use four specified common factors (CFs) to construct the expressions of 61-tap filter. The proposed architecture-1 calculates the expressions iteratively by multiplexing these four CFs in multiple constant multiplication (MCM) block, and outputs the results to perform their additions. The proposed architecture-2 uses sixty-one copies of MCM block for accelerating the calculations. We also apply our best CSE algorithm to optimize the coefficients of these two proposed filter designs. The experimental results show that our proposed CSE-based and multiplexer-based filter designs achieve performance improvement compared with previous designs.
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Chen, Jin Qin, and 陳進欽. "Eigen-approach for digital filter design." Thesis, 1995. http://ndltd.ncl.edu.tw/handle/65429471382279870438.
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DAI, YONG-LONG, and 戴永龍. "Computationally efficient FIR digital filter design." Thesis, 1992. http://ndltd.ncl.edu.tw/handle/85293113781971740877.
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Liang, Zhi-Hong, and 梁志鴻. "Design of Timing-Error-Tolerant Digital Filters for Various Filter Transformations." Thesis, 2014. http://ndltd.ncl.edu.tw/handle/4c2ycw.
Full textAbstract:
碩士國立東華大學
電機工程學系
102
In modern VLSI design, especially in system-on-chip, the number of transistors in a single chip keeps increasing thanks to the advance of chip manufacturing technology. However, as the feature size of modern chips shrinks, the circuits become more and more susceptible to noise, wire delay, and soft errors. One of these main problems is timing errors which are caused by process variation, device aging, etc. Such timing error problems can cause system failures. Hence, it is an important issue to solve the timing error problem while maintaining the performance of a chip. This thesis proposes various transformation designs for VLSI digital filters for tolerating multiple timing errors. We have developed a design methodology for VLSI digital filters, which can detect and tolerate multiple timing errors on-line. In order to achieve high performance of the digital filters, different transformations for various digital filter designs are applied. According to the design requirements, we choose the appropriate transformation for the filter in order to improve the performance, while it can still tolerate multiple timing errors. We have applied our techniques to two example digital signal filter designs, including a FIR filter and an IIR filter. Four examples for each circuit are studied and evaluated. We have implemented them using cell-based design flow on TSMC manufacturing technology. The implementation results show that our designs achieve high performance and tolerance of multiple timing errors for digital filters with reasonable cost.
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Hsu, Hong-Jie, and 許弘傑. "Digital Filter Design Using Fractional Bilinear Transform." Thesis, 2007. http://ndltd.ncl.edu.tw/handle/87726641015401787053.
Full textAbstract:
碩士國立臺灣大學
電信工程學研究所
95
The fractional bilinear transform used in analog-to-digital (A/D) conversion is proposed. This transformation is derived by means of fractional delay filter. Two forms are presented. One is approximated by Lagrange FIR fractional delay filter; the other is approximated by Thiran IIR fractional delay filter. Their magnitude responses are both linear in low frequency band, but differ in high frequency band for distinct applications. According to this transformation, first-order digital differentiator and integrator can be easily designed. Additionally, some design examples are illustrated to show the linear frequency mapping property when performing A/D conversion. The designed digital differentiator and integrator are compared with existing differentiators and integrators. The designed low-order differentiator is suitable for real-time applications. To improve highly selective 3-D recursive filters transformed from the bilinear transformation for tracking moving objects, first use 2-D examples to discuss the effects of diverse transformations. The experimental results show that the designed 3-D recursive plane filter can extract the desired moving object. Index terms—Bilinear transformation, digital differentiator, digital integrator, fractional bilinear transformation, fractional delay filter, three-dimensional planar-resonant recursive filter, tracking moving objects.