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1
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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Josephson, Chad. "ON THE DESIGN OF A SQUARE-ROOT NYQUIST PULSE SHAPING FILTER FOR AERONAUTICAL TELEMETRY." International Foundation for Telemetering, 2017. http://hdl.handle.net/10150/626963.
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Bandwidth efficient linear modulations require a pulse shape with finite support in the time domain while simultaneously achieving good spectral containment in the frequency domain. The square-root Nyquist pulse achieves zero intersymbol interference (ISI) at its matched-filter output but does so with infinite support in the time domain. This paper investigates three different methods for generating an FIR approximation of a square-root Nyquist pulse.
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Vergés, Fortià Vila. "Finite dimensional optimal linear mean square filter for continuos time Markovian jump linear systems." Laboratório Nacional de Computação Científica, 2017. https://tede.lncc.br/handle/tede/277.
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Fundação Carlos Chagas Filho de Amparo à Pesquisa do Estado do Rio de Janeiro (FAPERJ)
Stochastic differential equations with Markovian jump parameters constitute one of the most important class of hybrid dynamical systems, which has been extensively used for the modeling of dynamical systems which are subject to abrupt changes in their structure. The abrupt changes can be due, for instance, to abrupt environmental disturbances, component failure, volatility in economic systems, changes in subsystems interconnections, abrupt changes in the operation of a nonlinear plant, etc. This can be found, for instance, in aircraft control systems, robot systems, large flexible structure for space station, etc. We shall be particularly interested in the linear class which is dubbed in the literature as the class of Markov jump linear systems (MJLS). The jump mechanism is modeled by a Markov process, which is also known in the literature as the operation mode. The dissertation address the filtering problem of the operation mode for the class of MJLS. Previous result in the literature on this problem has been obtained by Wonham, which has shown the existence of an optimal nonlinear filter for this problem. The main hindrance with Wonham’s result, in the context of the control problem with partial observation of operation mode, is that it introduces a great deal of nonlinearity in the Hamilton-Jacobi- Belman equation, which makes it difficult to get an explicit closed solution for the control problem. Motivated by this, the main contribution of this dissertation is to devise an optimal linear filter for the mode operation, which we believe could be more favorable in the solution of the control problem with partial observations. In addition, relying on Murayama’s stochastic numerical method and the results of Yuan and Mao, we carry out simulation of Wonham’s filter, and the one devised in the dissertation, in order to compare their performances.
As equações diferenciais estocáticas com salto Markoviano constituem uma das clases de sistemas dinâmicos híbridos mais importantes, e tem sido muito usados para modelar sistemas sujeitos a mudanças abruptas na sua estructura. Essas mudanças podem ser devido a, por exemplo, perturbações ambientais, falhas em componentes, volatilidade em sistemas econômicos, mudanças em interconexões de subsistemas, mudanças abruptas em operações de plantas não lineares, etc. Estas falhas podem ser encontradas em sistemas de controle para aeronaves, sistemas robóticos, estructuras grandes e flexíveis em estações espaciais, etc. Nós estamos especialmente interessados na clase de sistemas lineares que é referenciada na literatura como sistemas lineares com salto Markoviano (SLSM). O mecanismo de salto é modelado por um processo de Markov, que é conhecido na literatura como modo de operação do sistema. Essa dissertação visa o problema de filtragem para o modo de operação do sistema linear com salto. Na literatura pode-se encontrar resultados já obtidos para esse problema como é o caso do filtro ótimo não linear deduzido por Wonham. Mas no contexto de controle ótimo com observações parciais do modo de operação, o filtro de Wonham introduz não linearidades na equação de Hamilton-Jacobi-Belman, fazendo com que seja muito complexo obter uma solução fechada para o problema de controle. A principal motivação desta dissertação é deduzir o filtro ótimo linear para o modo de operação, já que esta pode ser uma solução mais favorável para o problema de controle ótimo. Finalmente, usando o método numérico para equações diferenciais estocásticas de Euler-Murayama e o resultado de Yuan e Mao, realizamos a simulação do filtro de Wonham tal como o filtro deduzido neste trabalho, com o objetivo de comparar as respectivas performances.
4
Paulsen, Trevor H. "Low cost/high precision flight dynamics estimation using the square-root unscented Kalman filter /." Diss., CLICK HERE for online access, 2010. http://contentdm.lib.byu.edu/ETD/image/etd3181.pdf.
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Paulsen, Trevor H. "Low Cost/ High Precision Flight Dynamics Estimation Using the Square-Root Unscented Kalman Filter." BYU ScholarsArchive, 2009. https://scholarsarchive.byu.edu/etd/1922.
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For over a decade, Brigham Young University's Microwave Earth Remote Sensing (MERS) team has been developing SAR systems and SAR processing algorithms. In order to create the most accurate image reconstruction algorithms, detailed aircraft motion data is essential. In 2008, the MERS team purchased a costly inertial measurement unit (IMU) coupled with a high precision global positioning system (GPS) from NovAtel, Inc. In order to lower the cost of obtaining detailed motion measurements, the MERS group decided to build a system that mimics the capability the NovAtel system as closely as possible for a much lower cost. As a first step, the same sensors and a simplified set of flight dynamics are used. This thesis presents a standalone motion sensor recording system (MOTRON), and outlines a method of utilizing the square-root Unscented Kalman filter (SR-UKF) to estimate aircraft flight dynamics, based on recorded flight data, as an alternative to the extended Kalman filter. While the results of the SR-UKF are not as precise as the NovAtel results, they approach the accuracy of the NovAtel system despite the simplified dynamics model.
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Yamamoto, Kazuhiro, and Koji Tsuneyoshi. "Experimental study of hexagonal and square diesel particulate filters under controlled and uncontrolled catalyzed regeneration." Elsevier, 2013. http://hdl.handle.net/2237/20050.
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Lu, Ting Shan. "A microstrip square-loop dual-mode balun-bandpass filter with simultaneous size reduction and spurious response suppression." Thesis, University of Macau, 2009. http://umaclib3.umac.mo/record=b2129882.
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Vavruška, Marek. "Realised stochastic volatility in practice." Master's thesis, Vysoká škola ekonomická v Praze, 2012. http://www.nusl.cz/ntk/nusl-165381.
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Realised Stochastic Volatility model of Koopman and Scharth (2011) is applied to the five stocks listed on NYSE in this thesis. Aim of this thesis is to investigate the effect of speeding up the trade data processing by skipping the cleaning rule requiring the quote data. The framework of the Realised Stochastic Volatility model allows the realised measures to be biased estimates of the integrated volatility, which further supports this approach. The number of errors in recorded trades has decreased significantly during the past years. Different sample lengths were used to construct one day-ahead forecasts of realised measures to examine the forecast precision sensitivity to the rolling window length. Use of the longest window length does not lead to the lowest mean square error. The dominance of the Realised Stochastic Volatility model in terms of the lowest mean square errors of one day-ahead out-of-sample forecasts has been confirmed.
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Schmidt, Jason Knudsen. "Analysis of Square-Root Kalman Filters for Angles-Only Orbital Navigation and the Effects of Sensor Accuracy on State Observability." DigitalCommons@USU, 2010. https://digitalcommons.usu.edu/etd/627.
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Angles-only navigation is simple, robust, and well proven in many applications. However, it is sometimes ill-conditioned for orbital rendezvous and proximity operations because, without a direct range measurement, the distance to approaching satellites must be estimated by firing thrusters and observing the change in the target's bearing. Nevertheless, the simplicity of angles-only navigation gives it great appeal. The viability of this technique for relative navigation is examined by building a high-fidelity simulation and evaluating the sensitivity of the system to sensor errors. The relative performances of square-root filtering methods, including Potter, Carlson, and UD factorization filters, are compared to the conventional and Joseph formulations. Filter performance is evaluated during closed-loop "station keeping" operations in simulation.
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Kim, Taeho, and Monika Ivantysynova. "Active Vibration Control of Axial Piston Machine using Higher Harmonic Least Mean Square Control of Swash Plate." Saechsische Landesbibliothek- Staats- und Universitaetsbibliothek Dresden, 2016. http://nbn-resolving.de/urn:nbn:de:bsz:14-qucosa-199412.
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Noise emission is a major drawback of the positive displacement machine. The noise source can be divided into structure borne noise source (SBNS) and fluid borne noise source (FBNS). Passive techniques such as valve plate optimization have been used for noise reduction of axial piston machines. However, passive techniques are only effective for limited operating conditions or at least need compromises in design. In this paper, active vibration control of swash plate is investigated for vibration and noise reduction over a wide range of operating conditions as an additional method to passive noise reduction techniques. A 75cc pump has been modified for implementation of active vibration control using the swash plate. One tri-axial acceleration sensor and one angle sensor are installed on the swash plate and a high speed servovalve is used for the swash plate actuation. The multi-frequency two-weight least mean square (LMS) filter synthesizes the servovalve input signal to generate a destructive interference force which minimizes the swash plate vibration. An experimental test setup has been realized using Labview field-programmable gate array (FPGA) via cRIO. Simulation and experimental studies are conducted to investigate the possibility of active vibration control.
11
Gribaudo, Michael Louis. "Development of a system model and least mean square (LMS) filter for the Naval Postgraduate School (NPS) Infrared Search and Target Designation (IRSTD) system." Thesis, Monterey, California. Naval Postgraduate School, 1989. http://hdl.handle.net/10945/26990.
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Nino, Ruiz Elias David. "Efficient formulation and implementation of ensemble based methods in data assimilation." Diss., Virginia Tech, 2016. http://hdl.handle.net/10919/64438.
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Ensemble-based methods have gained widespread popularity in the field of data assimilation. An ensemble of model realizations encapsulates information about the error correlations driven by the physics and the dynamics of the numerical model. This information can be used to obtain improved estimates of the state of non-linear dynamical systems such as the atmosphere and/or the ocean. This work develops efficient ensemble-based methods for data assimilation.
A major bottleneck in ensemble Kalman filter (EnKF) implementations is the solution of a linear system at each analysis step. To alleviate it an EnKF implementation based on an iterative Sherman Morrison formula is proposed. The rank deficiency of the ensemble covariance matrix is exploited in order to efficiently compute the analysis increments during the assimilation process. The computational effort of the proposed method is comparable to those of the best EnKF implementations found in the current literature. The stability analysis of the new algorithm is theoretically proven based on the positiveness of the data error covariance matrix.
In order to improve the background error covariance matrices in ensemble-based data assimilation we explore the use of shrinkage covariance matrix estimators from ensembles. The resulting filter has attractive features in terms of both memory usage and computational complexity. Numerical results show that it performs better that traditional EnKF formulations.
In geophysical applications the correlations between errors corresponding to distant model components decreases rapidly with the distance. We propose a new and efficient implementation of the EnKF based on a modified Cholesky decomposition for inverse covariance matrix estimation. This approach exploits the conditional independence of background errors between distant model components with regard to a predefined radius of influence. Consequently, sparse estimators of the inverse background error covariance matrix can be obtained. This implies huge memory savings during the assimilation process under realistic weather forecast scenarios. Rigorous error bounds for the resulting estimator in the context of data assimilation are theoretically proved. The conclusion is that the resulting estimator converges to the true inverse background error covariance matrix when the ensemble size is of the order of the logarithm of the number of model components.
We explore high-performance implementations of the proposed EnKF algorithms. When the observational operator can be locally approximated for different regions of the domain, efficient parallel implementations of the EnKF formulations presented in this dissertation can be obtained. The parallel computation of the analysis increments is performed making use of domain decomposition. Local analysis increments are computed on (possibly) different processors. Once all local analysis increments have been computed they are mapped back onto the global domain to recover the global analysis. Tests performed with an atmospheric general circulation model at a T-63 resolution, and varying the number of processors from 96 to 2,048, reveal that the assimilation time can be decreased multiple fold for all the proposed EnKF formulations.Ensemble-based methods can be used to reformulate strong constraint four dimensional variational data assimilation such as to avoid the construction of adjoint models, which can be complicated for operational models. We propose a trust region approach based on ensembles in which the analysis increments are computed onto the space of an ensemble of snapshots. The quality of the resulting increments in the ensemble space is compared against the gains in the full space. Decisions on whether accept or reject solutions rely on trust region updating formulas. Results based on a atmospheric general circulation model with a T-42 resolution reveal that this methodology can improve the analysis accuracy.Ph. D.
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Yapici, Yavuz. "A Bidirectional Lms Algorithm For Estimation Of Fast Time-varying Channels." Phd thesis, METU, 2011. http://etd.lib.metu.edu.tr/upload/12613220/index.pdf.
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Effort to estimate unknown time-varying channels as a part of high-speed mobile communication systems is of interest especially for next-generation wireless systems. The high computational complexity of the optimal Wiener estimator usually makes its use impractical in fast time-varying channels. As a powerful candidate, the adaptive least mean squares (LMS) algorithm offers a computationally efficient solution with its simple first-order weight-vector update equation. However, the performance of the LMS algorithm deteriorates in time-varying channels as a result of the eigenvalue disparity, i.e., spread, of the input correlation matrix in such chan nels. In this work, we incorporate the L MS algorithm into the well-known bidirectional processing idea to produce an extension called the bidirectional LMS. This algorithm is shown to be robust to the adverse effects of time-varying channels such as large eigenvalue spread. The associated tracking performance is observed to be very close to that of the optimal Wiener filter in many cases and the bidirectional LMS algorithm is therefore referred to as near-optimal. The computational complexity is observed to increase by the bidirectional employment of the LMS algorithm, but nevertheless is significantly lower than that of the optimal Wiener filter. The tracking behavior of the bidirectional LMS algorithm is also analyzed and eventually a steady-state step-size dependent mean square error (MSE) expression is derived for single antenna flat-fading channels with various correlation properties. The aforementioned analysis is then generalized to include single-antenna frequency-selective channels where the so-called ind ependence assumption is no more applicable due to the channel memory at hand, and then to multi-antenna flat-fading channels. The optimal selection of the step-size values is also presented using the results of the MSE analysis. The numerical evaluations show a very good match between the theoretical and the experimental results under various scenarios. The tracking analysis of the bidirectional LMS algorithm is believed to be novel in the sense that although there are several works in the literature on the bidirectional estimation, none of them provides a theoretical analysis on the underlying estimators. An iterative channel estimation scheme is also presented as a more realistic application for each of the estimation algorithms and the channel models under consideration. As a result, the bidirectional LMS algorithm is observed to be very successful for this real-life application with its increased but still practical level of complexity, the near-optimal tracking performa nce and robustness to the imperfect initialization.
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Saghaian, Nejad Esfahani Sayed Mahdi. "STATISTICAL MODELS FOR CONSTANT FALSE-ALARM RATE THRESHOLD ESTIMATION IN SOUND SOURCE DETECTION SYSTEMS." UKnowledge, 2010. http://uknowledge.uky.edu/gradschool_theses/46.
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Constant False Alarm Rate (CFAR) Processors are important for applications where thousands of detection tests are made per second, such as in radar. This thesis introduces a new method for CFAR threshold estimation that is particularly applicable to sound source detection with distributed microphone systems. The novel CFAR Processor exploits the near symmetry about 0 for the acoustic pixel values created by steered-response coherent power in conjunction with a partial whitening preprocessor to estimate thresholds for positive values, which represent potential targets.
To remove the low frequency components responsible for degrading CFAR performance, fixed and adaptive high-pass filters are applied. A relation is proposed and it tested the minimum high-pass cut-off frequency and the microphone geometry.
Experimental results for linear, perimeter and planar arrays illustrate that for desired false alarm (FA) probabilities ranging from 10-1 and 10-6, a good CFAR performance can be achieved by modeling the coherent power with Chi-square and Weibull distributions and the ratio of desired over experimental FA probabilities can be limited within an order of magnitude.
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Arnal, Nicholas Christian. "A Study on 2.45 GHz Bandpass Filters Fabricated With Additive Manufacturing." Scholar Commons, 2015. http://scholarcommons.usf.edu/etd/5635.
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Square open loop resonator (SOLR) bandpass filters fabricated with additive manufacturing techniques are presented and studied. One filter contains novel 3D capacitive plates used to enhance resonator coupling. The filters are centered at 2.45 GHz and loaded with capacitors for miniaturization as low as 21% that of a conventional SOLR bandpass filter. The pass-band insertion loss of the filters ranges from 3.8 dB to 5.5 dB and the 3 dB bandwidth ranges from 180 MHz to 250 MHz. Also, degradation in the effective conductivity of printed ink as a function of substrate roughness is analyzed. Finally, a study of dielectric and metallic 3D printing processes that are candidates for digital manufacturing of integrated mobile phone client antennas is presented.
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Casselgren, Carl Johan, and Roland Häggkvist. "Completing partial Latin squares with one filled row, column and symbol." Linköpings universitet, Matematik och tillämpad matematik, 2013. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-92689.
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Let P be an n×n partial Latin square every non-empty cell of which lies in a fixed row r, a fixed column c or contains a fixed symbol s. Assume further that s is the symbol of cell (r,c) in P. We prove that P is completable to a Latin square if n≥8 and n is divisible by 4, or n≤7 and n∉{3,4,5}. Moreover, we present a polynomial algorithm for the completion of such a partial Latin square.
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Eskiyerli, Mirat Hayri. "Square root domain filters." Thesis, Imperial College London, 1998. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.299973.
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Jones, Haley M., and Haley Jones@anu edu au. "On multipath spatial diversity in wireless multiuser communications." The Australian National University. Research School of Information Sciences and Engineering, 2001. http://thesis.anu.edu.au./public/adt-ANU20050202.152811.
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The study of the spatial aspects of multipath in wireless communications environments
is an increasingly important addition to the study of the temporal aspects
in the search for ways to increase the utilization of the available wireless channel
capacity. Traditionally, multipath has been viewed as an encumbrance in wireless
communications, two of the major impairments being signal fading and intersymbol
interference. However, recently the potential advantages of the diversity offered by
multipath rich environments in multiuser communications have been recognised.
Space time coding, for example, is a recent technique which relies on a rich scattering
environment to create many practically uncorrelated signal transmission
channels. Most often, statistical models have been used to describe the multipath
environments in such applications. This approach has met with reasonable success
but is limited when the statistical nature of a field is not easily determined or is
not readily described by a known distribution.¶
Our primary aim in this thesis is to probe further into the nature of multipath
environments in order to gain a greater understanding of their characteristics and
diversity potential. We highlight the shortcomings of beamforming in a multipath
multiuser access environment. We show that the ability of a beamformer to resolve
two or more signals in angle directly limits its achievable capacity.¶
We test the probity of multipath as a source of spatial diversity, the limiting
case of which is co-located users. We introduce the concept of separability to define
the fundamental limits of a receiver to extract the signal of a desired user from
interfering users signals and noise. We consider the separability performances of
the minimum mean square error (MMSE), decorrelating (DEC) and matched filter
(MF) detectors as we bring the positions of a desired and an interfering user closer
together. We show that both the MMSE and DEC detectors are able to achieve
acceptable levels of separability with the users as close as λ/10.¶
In seeking a better understanding of the nature of multipath fields themselves,
we take two approaches. In the first we take a path oriented approach. The
effects on the variation of the field power of the relative values of parameters such
as amplitude and propagation direction are considered for a two path field. The
results are applied to a theoretical analysis of the behaviour of linear detectors
in multipath fields. This approach is insightful for fields with small numbers of
multipaths, but quickly becomes mathematically complex.¶
In a more general approach, we take a field oriented view, seeking to quantify
the complexity of arbitrary fields. We find that a multipath field has an intrinsic
dimensionality of (πe)R/λ≈8.54R/λ, for a field in a two dimensional circular region, increasing only linearly with the radius R of the region. This result implies that there is no such thing as an arbitrarily complicated multipath field. That is, a field generated by any number of nearfield and farfield, specular and diffuse
multipath reflections is no more complicated than a field generated by a limited
number of plane waves. As such, there are limits on how rich multipath can
be. This result has significant implications including means: i) to determine a
parsimonious parameterization for arbitrary multipath fields and ii) of synthesizing
arbitrary multipath fields with arbitrarily located nearfield or farfield, spatially
discrete or continuous sources. The theoretical results are corroborated by examples
of multipath field analysis and synthesis.
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Novanda, Happy. "Monitoring of power quality indices and assessment of signal distortions in wind farms." Thesis, University of Manchester, 2012. https://www.research.manchester.ac.uk/portal/en/theses/monitoring-of-power-quality-indices-and-assessment-of-signal-distortions-in-wind-farms(403a470c-279a-4b00-94dc-eaa2507dc579).html.
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Power quality has become one of major concerns in the power industry. It can be described as the reliability of the electric power to maintain continuity operation of end-use equipment. Power quality problems are defined as deviation of voltage or current waveforms from the ideal value. The expansion plan of wind power generation has raised concern regarding how it influences the voltage and current signals. The variability nature of wind energy and the requirements of wind power generation increase the potential problems such as frequency and harmonic distortions. In order to analyze and mitigate problems in wind power generation, it is important to monitor power quality in wind farm. Therefore, the more accurate and reliable parameter estimation methods suitable for wind power generation are needed. Three parameter estimation methods are proposed in this thesis to estimate the unknown parameters, i.e. amplitude and phase angle of fundamental and harmonic components, DC component and system frequency, during the dynamic change in wind farm. In the first method, a self-tuning procedure is introduced to least square method to increase the immunity of the algorithm to noise. In the second method, nonrecursive Newton Type Algorithm is utilised to estimate the unknown parameters by obtaining the left pseudoinverse of Jacobian matrix. In the last technique, unscented transformation is used to replace the linearization procedure to obtain mean and covariance which will be used in Kalman filter method. All of the proposed methods have been tested rigorously using computer simulated data and have shown their capability to track the unknown parameters under extreme distortions. The performances of proposed methods have also been compared using real recorded data from several wind farms in Europe and have demonstrated high correlation. This comparison has verified that UKF requires the shortest processing time and STLS requires the longest.
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Meng, Rui Daniel. "Design and implementation of sensor fusion for the towed synthetic aperture sonar." Thesis, University of Canterbury. Electrical and Computer Engineering, 2007. http://hdl.handle.net/10092/1199.
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For synthetic aperture imaging, position and orientation deviation is of great concern. Unknown motions of a Synthetic Aperture Sonar (SAS) can blur the reconstructed images and degrade image quality considerably. Considering the high sensitivity of synthetic aperture imaging technique to sonar deviation, this research aims at providing a thorough navigation solution for a free-towed synthetic aperture sonar (SAS) comprising aspects from the design and construction of the navigation card through to data postprocessing to produce position, velocity, and attitude information of the sonar. The sensor configuration of the designed navigation card is low-cost Micro-Electro-Mechanical-Systems (MEMS) Magnetic, Angular Rate, and Gravity (MARG) sensors including three angular rate gyroscopes, three dual-axial accelerometers, and a triaxial magnetic hybrid. These MARG sensors are mounted orthogonally on a standard 180mm Eurocard PCB to monitor the motions of the sonar in six degrees of freedom. Sensor calibration algorithms are presented for each individual sensor according to its characteristics to precisely determine sensor parameters. The nonlinear least square method and two-step estimator are particularly used for the calibration of accelerometers and magnetometers. A quaternion-based extended Kalman filter is developed based on a total state space model to fuse the calibrated navigation data. In the model, the frame transformations are described using quaternions instead of other attitude representations. The simulations and experimental results are demonstrated in this thesis to verify the capability of the sensor fusion strategy.
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Kahaei, Mohammad Hossein. "Performance analysis of adaptive lattice filters for FM signals and alpha-stable processes." Thesis, Queensland University of Technology, 1998. https://eprints.qut.edu.au/36044/7/36044_Digitised_Thesis.pdf.
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The performance of an adaptive filter may be studied through the behaviour
of the optimal and adaptive coefficients in a given environment. This thesis
investigates the performance of finite impulse response adaptive lattice filters for
two classes of input signals: (a) frequency modulated signals with polynomial
phases of order p in complex Gaussian white noise (as nonstationary signals),
and (b) the impulsive autoregressive processes with alpha-stable distributions (as
non-Gaussian signals).
Initially, an overview is given for linear prediction and adaptive filtering. The
convergence and tracking properties of the stochastic gradient algorithms are discussed
for stationary and nonstationary input signals. It is explained that the
stochastic gradient lattice algorithm has many advantages over the least-mean
square algorithm. Some of these advantages are having a modular structure,
easy-guaranteed stability, less sensitivity to the eigenvalue spread of the input autocorrelation
matrix, and easy quantization of filter coefficients (normally called
reflection coefficients).
We then characterize the performance of the stochastic gradient lattice algorithm
for the frequency modulated signals through the optimal and adaptive
lattice reflection coefficients. This is a difficult task due to the nonlinear dependence
of the adaptive reflection coefficients on the preceding stages and the input
signal. To ease the derivations, we assume that reflection coefficients of each
stage are independent of the inputs to that stage. Then the optimal lattice filter
is derived for the frequency modulated signals. This is performed by computing
the optimal values of residual errors, reflection coefficients, and recovery errors.
Next, we show the tracking behaviour of adaptive reflection coefficients for
frequency modulated signals. This is carried out by computing the tracking model
of these coefficients for the stochastic gradient lattice algorithm in average. The
second-order convergence of the adaptive coefficients is investigated by modeling
the theoretical asymptotic variance of the gradient noise at each stage. The
accuracy of the analytical results is verified by computer simulations.
Using the previous analytical results, we show a new property, the polynomial
order reducing property of adaptive lattice filters. This property may be used to
reduce the order of the polynomial phase of input frequency modulated signals.
Considering two examples, we show how this property may be used in processing
frequency modulated signals. In the first example, a detection procedure in carried
out on a frequency modulated signal with a second-order polynomial phase
in complex Gaussian white noise. We showed that using this technique a better
probability of detection is obtained for the reduced-order phase signals compared
to that of the traditional energy detector. Also, it is empirically shown that
the distribution of the gradient noise in the first adaptive reflection coefficients
approximates the Gaussian law. In the second example, the instantaneous frequency
of the same observed signal is estimated. We show that by using this
technique a lower mean square error is achieved for the estimated frequencies at
high signal-to-noise ratios in comparison to that of the adaptive line enhancer.
The performance of adaptive lattice filters is then investigated for the second
type of input signals, i.e., impulsive autoregressive processes with alpha-stable
distributions . The concept of alpha-stable distributions is first introduced. We
discuss that the stochastic gradient algorithm which performs desirable results
for finite variance input signals (like frequency modulated signals in noise) does
not perform a fast convergence for infinite variance stable processes (due to using
the minimum mean-square error criterion). To deal with such problems, the
concept of minimum dispersion criterion, fractional lower order moments, and
recently-developed algorithms for stable processes are introduced.
We then study the possibility of using the lattice structure for impulsive stable
processes. Accordingly, two new algorithms including the least-mean P-norm
lattice algorithm and its normalized version are proposed for lattice filters based
on the fractional lower order moments. Simulation results show that using the
proposed algorithms, faster convergence speeds are achieved for parameters estimation
of autoregressive stable processes with low to moderate degrees of impulsiveness
in comparison to many other algorithms. Also, we discuss the effect of
impulsiveness of stable processes on generating some misalignment between the
estimated parameters and the true values. Due to the infinite variance of stable
processes, the performance of the proposed algorithms is only investigated using
extensive computer simulations.
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Tambara, Rodrigo Varella. "Um controlador adaptativo robusto aplicado a conversores estáticos conectados à rede elétrica através de filtro LCL." Universidade Federal de Santa Maria, 2014. http://repositorio.ufsm.br/handle/1/3686.
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Conselho Nacional de Desenvolvimento Científico e TecnológicoThis Thesis deals with the development of a novel robust model reference adaptive controller (RMRAC), in discrete-time applied to grid-connected systems using LCL filter. This controller uses a modified robust parameters identifier based on a recursive least-squares algorithm. Two control structures are analyzed: state feedback approach and input-output approach. The robust stability analysis of the controller is presented including unmodeled dynamics. Thus, through these analyses, constraints design, in discrete-time, are obtained. For the validation of the proposed control algorithm, simulation and experimental results of a grid-connected power converter with LCL-filter, with current control, are presented.
Esta Tese de Doutorado apresenta o desenvolvimento de um novo controlador adaptativo por modelo de referência, totalmente desenvolvido em tempo discreto, aplicado a sistemas conectados à rede de energia elétrica empregando filtro LCL. Este controlador utiliza um identificador de parâmetros modificado robusto baseado no método dos mínimos quadrados recursivos. Em relação à estrutura do controlador, a abordagem por realimentação de estados e a abordagem entrada-saída são utilizadas. A análise de estabilidade robusta do controlador é apresentada incluindo dinâmicas não-modeladas. Por meio destas análises, restrições de projeto (em tempo discreto) são obtidas. Para a validação do algoritmo proposto, resultados de simulação e experimentais do sistema de controle de corrente em um conversor conectado à rede de energia elétrica com filtro LCL são apresentados.
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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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Štukovská, Petra. "Algoritmy detekce radarových cílů." Doctoral thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2021. http://www.nusl.cz/ntk/nusl-451229.
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This thesis focuses on detection algorithms of radar targets, namely on group of techniques for removing of disturbing reflections from static objects - clutter and for suppression of distortion products caused by the phase noise of the transmitter and receiver. Methods for distortion suppression in received signal are designed for implementation in the developed active multistatic radar, which operates in the code division multiplex of several transmitters on single frequency. The aim of the doctoral thesis is to design, implement in tool for technical computing MATLAB and analyze the effectiveness and computational complexity of these techniques on simulated and real data.
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Munoz, Maldonado Yolanda. "Mixed models, posterior means and penalized least squares." Texas A&M University, 2005. http://hdl.handle.net/1969.1/2637.
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In recent years there has been increased research activity in the area of Func-
tional Data Analysis. Methodology from finite dimensional multivariate analysis has
been extended to the functional data setting giving birth to Functional ANOVA,
Functional Principal Components Analysis, etc. In particular, some studies have pro-
posed inferential techniques for various functional models that have connections to
well known areas such as mixed-effects models or spline smoothing. The methodol-
ogy used in these cases is computationally intensive since it involves the estimation of
coefficients in linear models, adaptive selection of smoothing parameters, estimation
of variances components, etc.
This dissertation proposes a wide-ranging modeling framework that includes
many functional linear models as special cases. Three widely used tools are con-
sidered: mixed-effects models, penalized least squares, and Bayesian prediction. We
show that, in certain important cases, the same numerical answer is obtained for these
seemingly different techniques. In addition, under certain assumptions, an applica-
tion of a Kalman filter algorithm is shown to improve the order of computations, by
two orders of magnitude, for point and interval estimates (with n being the sample
size). A functional data analysis setting is used to exemplify our results.
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Lee, Ho Tong. "Prism-coupled square optical micropillar resonator-based filters for optical communications /." View abstract or full-text, 2004. http://library.ust.hk/cgi/db/thesis.pl?ELEC%202004%20LEE.
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Thesis (M. Phil.)--Hong Kong University of Science and Technology, 2004.Includes bibliographical references (leaves 134-138). Also available in electronic version. Access restricted to campus users.
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Fong, Chung Yan. "Silicon-based laterally waveguide-coupled square microcavity channel add-drop filters /." View abstract or full-text, 2004. http://library.ust.hk/cgi/db/thesis.pl?ELEC%202004%20FONG.
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Thesis (M. Phil.)--Hong Kong University of Science and Technology, 2004.Includes bibliographical references (leaves 98-103). Also available in electronic version. Access restricted to campus users.
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Barbosa, Daniel. "Estimação da freqüência em sistemas elétricos de potência através de filtragem adaptativa." Universidade de São Paulo, 2007. http://www.teses.usp.br/teses/disponiveis/18/18154/tde-11092007-155259/.
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Este trabalho apresenta um método para a estimação da freqüência em sistemas elétricos de potência utilizando filtros adaptativos baseados no algoritmo dos mínimos quadrados (LMS - least mean square). A análise do sistema de potência é realizada através da conversão das tensões trifásicas em um sinal complexo pela aplicação da transformada \'alfa\'\'beta\', cuja forma complexa foi direcionada ao algoritmo de filtragem adaptativa. O método é baseado na aplicação da filtragem adaptativa para a realização de rastreio do sinal de entrada, o que permite verificar o seu comportamento variante no tempo. O algoritmo proposto foi testado através de formas de ondas geradas com o software Matlab e de simulações realizadas através do software Alternative Transients Program (ATP). É importante salientar que nas simulações do ATP foram modelados diversos equipamentos que constituem o sistema elétrico de potência, incluindo um gerador síncrono com regulação de velocidade, linhas de transmissão com variação em freqüência e transformadores de potência com suas respectivas curvas de saturação. Estas modelagens tiveram por objetivo gerar dados das mais diversas e distintas situações para a verificação e análise da metodologia proposta. Os resultados da pesquisa mostram a excelência na aplicabilidade do algoritmo proposto na estimação da freqüência de um sistema elétrico, mesmo com sinais ruidosos, além do rastreio fiel da freqüência em situações de manobra e operação. Alguns dos resultados apresentados comparam as estimações obtidas pela técnica proposta em relação às estimações de um determinado relé comercial, habilitado à supervisão da freqüência.This work presents a method for frequency estimation in power systems using adaptive filters based in the algorithm of least mean square (LMS). The analysis of the power system is made through the conversion of the three-phase voltages in a complex signal with the application of \'alfa\'\'beta\' transform, whose complex form was directed to the algorithm of adaptive filtering. The method is based on the application of the adaptive filtering for tracking the input signal, and it allows verifying its variant behavior in time. The algorithm was tested through waveforms generated by Matlab software and simulations carried out through Alternative Transients Program (ATP) software. It is important to point out that in the simulations using ATP many diferent power system equipments had been modeled, including a synchronous generator with speed regulation, transmission lines with variation in frequency and power transformers with their saturation curves. The objective of these tests was to generate data for diverse and distinct situations for the verification and the analysis of the proposed methodology. The results of the research show the excellence in the applicability of the algorithm considered in frequency estimation of an electrical system, even with noisy signals, as well as the tracking of the frequency during operation. Some of the results are compared to the ones presented by a commercial relay set to track frequency.
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Tian, Yongsheng. "Low turbulence natural convection in an air filled square cavity." Thesis, London South Bank University, 1997. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.245135.
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Daniel, Timothy Seth. "The effects of precision on the fast, recursive least-squares transversal filters for adaptive filtering." Thesis, This resource online, 1990. http://scholar.lib.vt.edu/theses/available/etd-03242009-040454/.
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Karlsson, Erlendur. "Least squares arma modeling of linear time-varying systems : lattice filter structures and fast RLS algorithms." Diss., Georgia Institute of Technology, 1987. http://hdl.handle.net/1853/15936.
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Hardika, Made S. "Seismic resistance of concrete-filled square steel hollow structuralsection beam-columns." Thesis, National Library of Canada = Bibliothèque nationale du Canada, 2002. http://www.collectionscanada.ca/obj/s4/f2/dsk3/ftp05/NQ66151.pdf.
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Göransson, Herman. "Completing partial latin squares with 2 filled rows and 3 filled columns." Thesis, Linköpings universitet, Matematiska institutionen, 2020. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-163092.
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The set PLS(a, b; n) is the set of all partial latin squares of order n with a completed rows, b completed columns and all other cells empty. We identify reductions of partial latin squares in PLS(2, 3; n) by using permutations described by filled rows and intersections of filled rows and columns. We find that all partial latin squares in PLS(2, 3;n), where n is sufficiently large, can be completed if such a reduction can be completed. We also show that all partial latin squares in PLS(2, 3; n) where the intersection of filled rows and columns form a latin rectangle have completions for n ≥ 8.
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Šimoník, Petr. "Měřič odstupu signálu od šumu obrazových signálů." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2008. http://www.nusl.cz/ntk/nusl-217681.
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The diplomma thesis is dealing with possibilities of Signal to noise ratio measurement by method, which is based on direct measurement. It is chosen the most suitable method – signal and noise separation to two different parallel signal branches, where is measured signal strength in one branch and root mean square value in the other. The thesis is consisted of a concept of detail block scheme of Signal to noise ratio meter, which was designed in terms of theoretical knowledge. Particular functional blocks were circuit-designed, the active and passive parts were chosen and their function were described. There were made simulation and displayed input and output time flows. There is designed the whole connection of engineered Signal to noise ratio meter in the last part of my thesis. The double-sided board of printed circuit is contained too. It was created simple programme for supervisor micro-processor. Thereby were constructed complete bases for realization.
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Zadeh, Ramin Agha. "Performance control of distributed generation using digital estimation of signal parameters." Thesis, Queensland University of Technology, 2010. https://eprints.qut.edu.au/47011/1/Ramin_Agha_Zadeh_Thesis.pdf.
Full textAbstract:
The Queensland University of Technology (QUT) allows the presentation of a thesis for the Degree of Doctor of Philosophy in the format of published or submitted papers, where such papers have been published, accepted or submitted during the period of candidature. This thesis is composed of seven published/submitted papers, of which one has been published, three accepted for publication and the other three are under review. This project is financially supported by an Australian Research Council (ARC) Discovery Grant with the aim of proposing strategies for the performance control of Distributed Generation (DG) system with digital estimation of power system signal parameters.
Distributed Generation (DG) has been recently introduced as a new concept for the generation of power and the enhancement of conventionally produced electricity. Global warming issue calls for renewable energy resources in electricity production. Distributed generation based on solar energy (photovoltaic and solar thermal), wind, biomass, mini-hydro along with use of fuel cell and micro turbine will gain substantial momentum in the near future. Technically, DG can be a viable solution for the issue of the integration of renewable or non-conventional energy resources. Basically, DG sources can be connected to local power system through power electronic devices, i.e. inverters or ac-ac converters. The interconnection of DG systems to power system as a compensator or a power source with high quality performance is the main aim of this study. Source and load unbalance, load non-linearity, interharmonic distortion, supply voltage distortion, distortion at the point of common coupling in weak source cases, source current power factor, and synchronism of generated currents or voltages are the issues of concern.
The interconnection of DG sources shall be carried out by using power electronics switching devices that inject high frequency components rather than the desired current. Also, noise and harmonic distortions can impact the performance of the control strategies. To be able to mitigate the negative effect of high frequency and harmonic as well as noise distortion to achieve satisfactory performance of DG systems, new methods of signal parameter estimation have been proposed in this thesis. These methods are based on processing the digital samples of power system signals. Thus, proposing advanced techniques for the digital estimation of signal parameters and methods for the generation of DG reference currents using the estimates provided is the targeted scope of this thesis. An introduction to this research – including a description of the research problem, the literature review and an account of the research progress linking the research papers – is presented in Chapter 1.
One of the main parameters of a power system signal is its frequency. Phasor Measurement (PM) technique is one of the renowned and advanced techniques used for the estimation of power system frequency. Chapter 2 focuses on an in-depth analysis conducted on the PM technique to reveal its strengths and drawbacks. The analysis will be followed by a new technique proposed to enhance the speed of the PM technique while the input signal is free of even-order harmonics. The other techniques proposed in this thesis as the novel ones will be compared with the PM technique comprehensively studied in Chapter 2.
An algorithm based on the concept of Kalman filtering is proposed in Chapter 3. The algorithm is intended to estimate signal parameters like amplitude, frequency and phase angle in the online mode. The Kalman filter is modified to operate on the output signal of a Finite Impulse Response (FIR) filter designed by a plain summation. The frequency estimation unit is independent from the Kalman filter and uses the samples refined by the FIR filter. The frequency estimated is given to the Kalman filter to be used in building the transition matrices. The initial settings for the modified Kalman filter are obtained through a trial and error exercise.
Another algorithm again based on the concept of Kalman filtering is proposed in Chapter 4 for the estimation of signal parameters. The Kalman filter is also modified to operate on the output signal of the same FIR filter explained above. Nevertheless, the frequency estimation unit, unlike the one proposed in Chapter 3, is not segregated and it interacts with the Kalman filter. The frequency estimated is given to the Kalman filter and other parameters such as the amplitudes and phase angles estimated by the Kalman filter is taken to the frequency estimation unit.
Chapter 5 proposes another algorithm based on the concept of Kalman filtering. This time, the state parameters are obtained through matrix arrangements where the noise level is reduced on the sample vector. The purified state vector is used to obtain a new measurement vector for a basic Kalman filter applied. The Kalman filter used has similar structure to a basic Kalman filter except the initial settings are computed through an extensive math-work with regards to the matrix arrangement utilized.
Chapter 6 proposes another algorithm based on the concept of Kalman filtering similar to that of Chapter 3. However, this time the initial settings required for the better performance of the modified Kalman filter are calculated instead of being guessed by trial and error exercises. The simulations results for the parameters of signal estimated are enhanced due to the correct settings applied. Moreover, an enhanced Least Error Square (LES) technique is proposed to take on the estimation when a critical transient is detected in the input signal. In fact, some large, sudden changes in the parameters of the signal at these critical transients are not very well tracked by Kalman filtering. However, the proposed LES technique is found to be much faster in tracking these changes. Therefore, an appropriate combination of the LES and modified Kalman filtering is proposed in Chapter 6. Also, this time the ability of the proposed algorithm is verified on the real data obtained from a prototype test object.
Chapter 7 proposes the other algorithm based on the concept of Kalman filtering similar to those of Chapter 3 and 6. However, this time an optimal digital filter is designed instead of the simple summation FIR filter. New initial settings for the modified Kalman filter are calculated based on the coefficients of the digital filter applied. Also, the ability of the proposed algorithm is verified on the real data obtained from a prototype test object.
Chapter 8 uses the estimation algorithm proposed in Chapter 7 for the interconnection scheme of a DG to power network. Robust estimates of the signal amplitudes and phase angles obtained by the estimation approach are used in the reference generation of the compensation scheme. Several simulation tests provided in this chapter show that the proposed scheme can very well handle the source and load unbalance, load non-linearity, interharmonic distortion, supply voltage distortion, and synchronism of generated currents or voltages. The purposed compensation scheme also prevents distortion in voltage at the point of common coupling in weak source cases, balances the source currents, and makes the supply side power factor a desired value.
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Ampofo, Felix. "Turbulent natural convection in an air filled standard or partitioned square cavity." Thesis, London South Bank University, 2001. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.618656.
Full textAbstract:
Natural convection is a common phenomenon occurring both in nature and in industrial applications. Among the various confined enclosures, the rectangular cavity (empty or partitioned) is the most extensively studied enclosure because many engineering geometries can be simplified to this configuration. Natural convection in a rectangular cavity is also a very good vehicle for both experimental and theoretical studies. In experimental terms, the geometry of the rectangular cavity is simple and its boundary conditions are relatively easy to realise so that researchers can focus on the measurements of important quantities such as velocity and temperature profiles. In numerical terms, the flow phenomena in the cavity are so complicated and so plentiful that they intrigue both physicists and engineers. The first part of the thesis introduces the background of this project and an extensive review of the available literature. The literature review is divided into two parts. The first part presents an overview of natural convection in standard cavities with particular emphasis on previous studies pertaining to the simultaneous measurement of local velocity and temperature. The second part reviews and discusses previous studies, especially in the turbulent flow region, in natural convection in partitioned rectangular cavities. The experimental and numerical results are compared. Problems encountered in earlier studies are discussed. A proposal for the present study is given at the end of this part, which is then followed by the mathematical description and definition of the problem. The second part of the thesis introduces the existing experimental system at South Bank University together with the modifications made to suit the present study. The third part of the thesis presents the experimental results, which are then compared with earlier work. The thermal and fluid flow fields were systematically surveyed. The experimental contour plots of T, T', u~, v~ and u'v' and the velocity vector plots are reported for the first time for turbulent natural convection in both standard and partitioned square cavities. In addition to these quantities, the experimental turbulent quantities, u'T' and v'T', are also reported for the first time for natural convection in a standard square cavity. Also, in this part, the results of numerical simulations performed with the commercial software CFX-5.4.1 will be presented and compared with the experimental results. Despite the numerical simulations, this thesis is mainly concerned with the experimental investigation of low level turbulence natural convection in an air filled standard or partitioned square cavity. The experiments were performed with high precision and avoided some of the problems and shortcomings of earlier work. The results form benchmark data for low level turbulence natural convection and are useful for comparison with numerical predictions and CFD code validation and development.
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Saket, Hassan Khalil. "The fatigue behaviour of fillet welded joints of plates and square hollow sections." Thesis, University of Manchester, 1988. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.281213.
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Callender, Christopher Peter. "Numerically robust implementations of fast recursive least squares adaptive filters using interval arithmetic." Thesis, University of Edinburgh, 1991. http://hdl.handle.net/1842/10853.
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Algorithms have been developed which perform least squares adaptive filtering with great computational efficiency. Unfortunately, the fast recursive least squares (RLS) algorithms all exhibit numerical instability due to finite precision computational errors, resulting in their failure to produce a useful solution after a short number of iterations. In this thesis, a new solution to this instability problem is considered, making use of interval arithmetic. By modifying the algorithm so that upper and lower bounds are placed on all quantities calculated, it is possible to obtain a measure of confidence in the solution calculated by a fast RLS algorithm and if it is subject to a high degree of inaccuracy due to finite precision computational errors, then the algorithm may be rescued, using a reinitialisation procedure. Simulation results show that the stabilised algorithms offer an accuracy of solution comparable with the standard recursive least squares algorithm. Both floating and fixed point implementations of the interval arithmetic method are simulated and long-term stability is demonstrated in both cases. A hardware verification of the simulation results is also performed, using a digital signal processor(DSP). The results from this indicate that the stabilised fast RLS algorithms are suitable for a number of applications requiring high speed, real time adaptive filtering. A design study for a very large scale integration (VLSI) technology coprocessor, which provides hardware support for interval multiplication, is also considered. This device would enable the hardware realisation of a fast RLS algorithm to operate at far greater speed than that obtained by performing interval multiplication using a DSP. Finally, the results presented in this thesis are summarised and the achievements and limitations of the work are identified. Areas for further research are suggested.
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Ledezma, Luis Manuel. "A Study on the Miniaturization of Microstrip Square Open Loop Resonators." Scholar Commons, 2011. http://scholarcommons.usf.edu/etd/3202.
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A miniaturization technique that allows the size of microstrip square open loop resonators to be reduced by more than 80% is presented and studied. The technique is based on the loading of the resonator with a series surface mount capacitor. It is shown that this technique allows the design of microwave bandpass filters with a wider stopband when compared with conventional designs. It is also proved that the insertion loss of the miniaturized filter is not degraded, but in fact can be maintained or even enhanced by the miniaturization process; this is true whenever the quality factor of the lumped capacitor is higher than the quality factor of the microstrip resonator. Finally, the feasibility of using the effect of the capacitor loss in the miniaturized resonator quality factor as a method to measure the effective series resistance of surface mount capacitors is studied, and recommendations towards its implementation are presented.
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Felixson, Henrik. "Vehicle Ahead Property Estimation in Heavy Duty Vehicles." Thesis, Linköpings universitet, Reglerteknik, 2014. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-108341.
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Doheny, David A. "Real Time Digital Signal Processing Adaptive Filters for Correlated Noise Reduction in Ring Laser Gyro Inertial Systems." [Tampa, Fla.] : University of South Florida, 2004. http://purl.fcla.edu/fcla/etd/SFE0000306.
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Chang, Long Wee. "Effects of video bandwidth on the performance of a square law detector with Gaussian IF and video filters." Thesis, Monterey, California. Naval Postgraduate School, 1991. http://hdl.handle.net/10945/28151.
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Zheng, Xiang. "Optimization of Sampling Structure Conversion Methods for Color Mosaic Displays." Thesis, Université d'Ottawa / University of Ottawa, 2014. http://hdl.handle.net/10393/31189.
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Although many devices can be used to capture images of high resolution, there is still a need to show these images on displays with low resolution. Existing methods of subpixel-based down-sampling are reviewed in this thesis and their limitations are described. A new approach to optimizing sampling structure conversion for color mosaic displays is developed. Full color images are filtered by a set of optimal filters before down-sampling, resulting in better image quality according to the SCIELAB measure, a spatial extension of the CIELAB metric measuring perceptual color difference. The typical RGB stripe display pattern is tested to get the optimal filters using least-squares filter design. The new approach is also implemented on a widely used two-dimensional display pattern, the Pentile RGBG. Clear images are produced and color fringing artifacts are reduced. Quality of down-sampled images are compared using SCIELAB and by visual inspection.
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Sladká, Pavla. "Využití rekurzivní metody nejmenších čtverců pro analýzu dynamiky vozidel." Master's thesis, Vysoké učení technické v Brně. Fakulta strojního inženýrství, 2010. http://www.nusl.cz/ntk/nusl-229022.
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Tato diplomová práce nastiňuje teoretické základy potřebné pro návrh algoritmu rekurzivní metody nejmenších čtverců a následně jeho aplikaci na experimentální data naměřená při testovacím manévru uskutečněném v roce 2001. Analyzována byla příčná dynamika jednostopého rovinného modelu vozidla. Práce také obsahuje srovnání výsledků získaných jednak rekurzivním algoritmem a dále i algoritmem Kalmanova filtru.
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Lai, Jui-chi, and 賴瑞麒. "1.5V Square-Root Domain Filter." Thesis, 2009. http://ndltd.ncl.edu.tw/handle/736nqq.
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碩士國立中山大學
電機工程學系研究所
97
Conventional gm-c filters have limited voltage swings in low voltage operation. CMOS companding filters replace gm-c filters in low voltage environment for high dynamic range. The square-root domain filter and log-domain filter belongs to this companding filter category. In this thesis, a second order low pass square root domain filter (SRD filter) based on the up-down TL (translinear loop) circuit structure is presented. The SRD filter consists of four geometric-mean cells and three squarer/divider cells. The advantages of the proposed circuits are low supply voltage, low power consumption, high bandwidth, and low total harmonic distortion (THD). The circuit has been fabricated with 0.35μm CMOS technology. It operates with a supply voltage of 1.5V, and the bias current varies from 0.5μA to 30μA. Measurement results show that the cutoff frequency can be tuned from 3.12MHz to 8.11MHz when the Capacitance (C) is 5pF.The total harmonic distortion is 0.28%, and the power consumption is 1.09mW.
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LIN, JI-FENG, and 林繼豐. "The arithmetic design on square-root normalized least-square lattice filter." Thesis, 1990. http://ndltd.ncl.edu.tw/handle/60399553446033199656.
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Chang, Shih-Hao, and 張世豪. "A Square Root Domain Filter with Translinear Principle." Thesis, 2008. http://ndltd.ncl.edu.tw/handle/6uu3d2.
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碩士國立中山大學
電機工程學系研究所
96
In this thesis, a first order low pass square root domain filter (SRD filter) based on the novel operational transconductor amplifiers (OTAs) is presented. The SRD filter consists of a translinear filter and two OTAs. Because the conventional OTA has small input voltage swings, which violates the large signal operation of a SRD filter. We propose the novel OTA which is based on the large signal behaviors of MOSFETs, and the OTA also has large signal operation. We improve Cruz’s SRD filter [22], reduce the number of the transconductors from 3 to 2, and replace Class-AB linear transconductors with the proposed OTAs. The MOSFET count of whole circuit can be reduced. Therefore, the OTAs have many advantages: wider input voltage swing, low supply voltage, low power consumption, and small chip area. The circuit has been fabricated with 0.35μm CMOS technology. It operates with a supply voltage 1.5V and the bias current varies from 0.3μA to 15μA. Measurement results show that the cutoff frequency can be tuned from 1.1kHz to 35.2kHz when the external capacitance C is 1nF and the cutoff frequency can be tuned from 8.7kHz to 310.4kHz when the external capacitance C is 100pF. The total harmonic distortions are 0.93% and 0.91% when the external capacitances C are 1nF and 100pF, and the power consumption is 152.29μW.
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Lo, Wan-Chen, and 羅宛珍. "Low Voltage Low Power Square-Root-Domain Filter." Thesis, 2006. http://ndltd.ncl.edu.tw/handle/24526224277511824972.
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碩士國立中山大學
電機工程學系研究所
94
In this thesis, a brand new first-order low pass square root domain filter (SRD filter) based on operational transconductors amplifiers (OTAs) is presented. The SRD filter consists of a translinear filter and two OTAs. We improve Cruz’s SRD filter [15], reduce the number of transconductors from 3 to 2, and replace Class-AB linear transconductors with OTAs. The circuit has the least number of transistors up to date, therefore, the least power consumption and least chip area. The circuit has been fabricated with 0.35μm CMOS technology. It operates with a supply voltage 1.5V and the biasing current varies from 0.05uA to 15uA. Measurement results show that the cutoff frequency of the filter can be tuned from 250 Hz to 29 kHz when the external capacitance C is 1nF and the cutoff frequency can be tuned from 1.8 kHz to 237kHz when the external capacitance C is 100pF. The total harmonic distortion is 1.03% and 1.01% when the external capacitance C is 1nF and 100pF and the power consumption is 116μW.
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LIU, PEI-QUAN, and 劉培權. "The ranked order least mean square adaptive filter." Thesis, 1992. http://ndltd.ncl.edu.tw/handle/57527452536338497384.
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Soewito, Atmadji Wiseso. "Least square digital filter design in the frequency domain." Thesis, 1991. http://hdl.handle.net/1911/16483.
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This thesis develops new methods for obtaining optimal frequency domain approximation in the design of digital filters. The approach uses a squared error approximation criterion and allows a transition band in the desired frequency response specification. Four particular cases are considered.
The first case considers FIR filters whose frequency responses include transition bands and whose errors are uniformly weighted. The new technique defines the exact transition edges, maintains optimality, reduces the Gibbs' phenomenon, and has low computational requirement. Two error measures are investigated, the discrete squared error and the integral squared error.
The second case involves the near-singularity problems which occur in the design of FIR filter with zero transition band error. A new approach is developed and a considerable improvement over the existing techniques is obtained. This new design method could accommodate almost any weighting function, and has computational complexity comparable to those of the existing ones.
The third case includes a complex frequency domain approximation in the IIR filter design. A technique based on the quasilinearization method is developed, and comparison shows that in most design examples the new approach appears to converge more rapidly than other competitive algorithms. Unlike other frequency domain linearization methods, the new algorithm does not modify the error criterion.
The fourth case includes an IIR filter design method with magnitude specification in the frequency domain. The problem is formulated as a successive complex frequency domain approximation. A new algorithm to solve this problem is developed, and experiments indicate that the algorithm converges faster than the other competitors do for most of filter design examples.