Dissertations / Theses: 'Signal processing Electric filters'

1

Orcutt, Edward Kerry 1964. "Correlation filters for time domain signal processing." Thesis, The University of Arizona, 1989. http://hdl.handle.net/10150/277215.

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This study proposes employing new filters in various configurations for use in digital communication systems. We believe that significant improvements in such performance areas as transmission rate and synchronization may be achieved by incorporating these filters into digital communications receivers. Recently reported in the literature, these filters may offer advantages over the matched filter which allow enhancements in data rates, ISI tolerance, and synchronization. To make full use of the benefits of these filters, we introduce the concept of parallel signal transmission over a single channel. We also examine the effects of signal set selection and noise on performance.

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Tsim, Man-tat Jimmy. "High speed realisation of digital filters /." [Hong Kong] : University of Hong Kong, 1989. http://sunzi.lib.hku.hk/hkuto/record.jsp?B12374088.

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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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Wepman, Jeffery Alan. "THE MODELING AND ANALYSIS OF AN AUTOMATICALLY TUNED FILTER." Thesis, The University of Arizona, 1985. http://hdl.handle.net/10150/275276.

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詹文達 and Man-tat Jimmy Tsim. "High speed realisation of digital filters." Thesis, The University of Hong Kong (Pokfulam, Hong Kong), 1989. http://hub.hku.hk/bib/B31208939.

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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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Van, Duyne Scott A. "Digital filter applications to modeling wave propagation in springs, strings, membranes and acoustical space /." May be available electronically:, 2007. http://proquest.umi.com/login?COPT=REJTPTU1MTUmSU5UPTAmVkVSPTI=&clientId=12498.

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8

Kwan, Wai Ming Hercule. "Parallel implementation of a fast third-order volterra digital filter /." Digital version accessible at:, 1998. http://wwwlib.umi.com/cr/utexas/main.

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9

Karam, Lina J. "Design of complex digital FIR filters in the chebyshev sense." Diss., Georgia Institute of Technology, 1995. http://hdl.handle.net/1853/22219.

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Luo, Yi. "Theory and design of M-channel cosine modulated filter banks and wavelets /." Hong Kong : University of Hong Kong, 1998. http://sunzi.lib.hku.hk/hkuto/record.jsp?B19471130.

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Chan, Chi-wing. "Design of 1-D and 2-D perfect reconstruction filter banks /." Hong Kong : University of Hong Kong, 1996. http://sunzi.lib.hku.hk/hkuto/record.jsp?B20717908.

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12

Belayneh, Sirak. "The identity of zeros of higher and lower dimensional filter banks and the construction of multidimensional nonseparable wavelets." Fairfax, VA : George Mason University, 2008. http://hdl.handle.net/1920/3417.

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Thesis (Ph.D.)--George Mason University, 2008.
Vita: p. 160. Thesis director: Edward J. Wegman. Submitted in partial fulfillment of the requirements for the degree of Doctor of Philosophy in Information Technology. Title from PDF t.p. (viewed Mar. 9, 2009). Includes bibliographical references (p. 151-159). Also issued in print.

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13

Hezar, Rahmi. "Oversampled digital filters : a design methodology and implementation." Diss., Georgia Institute of Technology, 2000. http://hdl.handle.net/1853/14936.

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Nayebi, Kambiz. "A time domain framework for the analysis and design of FIR multirate filter bank systems." Diss., Georgia Institute of Technology, 1990. http://hdl.handle.net/1853/13867.

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Luo, Yi, and 羅毅. "Theory and design of M-channel cosine modulated filter banks and wavelets." Thesis, The University of Hong Kong (Pokfulam, Hong Kong), 1998. http://hub.hku.hk/bib/B31215634.

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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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17

Fraser, David Raye. "Implementation of a modal filtering procedure." Thesis, University of British Columbia, 1988. http://hdl.handle.net/2429/28382.

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A FORTRAN program has been developed in order to investigate the process of modal parameter estimation and non-parametric system identification. The theory underlying the process of modal parameter estimation is reviewed and the decoupling of a MIMO system into several SISO systems is demonstrated. Modal filtering is shown to be useful in the field of non-parametric system identification and it is shown that it may also be of some use in the field of signal processing. The program is documented. It simulates the output of a n-th order system from which a smaller order subsystem can be decoupled. The modal parameters of a subsystem output signal and its first two derivatives and the modal parameters of a second subsystem output and its first derivative are calculated. The unit step response of the theoretical system and the subsystem are then calculated. The signals are then modal filtered to produce the periodic unit step response and the periodic unit square wave response. Finally, the discrete Fourier coefficients of the periodic unit step response are calculated.
Applied Science, Faculty of
Electrical and Computer Engineering, Department of
Graduate

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Lertniphonphun, Worayot. "Unified design procedure for digital filters in the complex domain." Diss., Georgia Institute of Technology, 2001. http://hdl.handle.net/1853/14765.

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黃毅 and Ngai Wong. "Signal processing: linearized noise analysis of delta-operator based filters and nonlinear stability study ofsigma-delta modulators." Thesis, The University of Hong Kong (Pokfulam, Hong Kong), 2002. http://hub.hku.hk/bib/B31244920.

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Kwan, Man-Wai. "Minimal transmit redundancy FIR precoder-equalizer systems design /." View abstract or full-text, 2004. http://library.ust.hk/cgi/db/thesis.pl?ELEC%202004%20KWAN.

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Kucic, Matthew R. "Analog programmable filters using floating-gate arrays." Thesis, Georgia Institute of Technology, 2000. http://hdl.handle.net/1853/13755.

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Alexandrou, Alexandros. "Design of filter banks for subband coding systems." Thesis, McGill University, 1985. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=63318.

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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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陳志榮 and Chi-wing Chan. "Design of 1-D and 2-D perfect reconstruction filter banks." Thesis, The University of Hong Kong (Pokfulam, Hong Kong), 1996. http://hub.hku.hk/bib/B31214915.

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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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Kose, Neslihan. "Light Flicker Evaluation Of Electric Arc Furnaces Based On Novel Signal Processing Algorithms." Master's thesis, METU, 2009. http://etd.lib.metu.edu.tr/upload/12611074/index.pdf.

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In this research work, two new flickermeters are proposed to estimate the light flicker caused by electric arc furnaces (EAFs) where the system frequency deviates significantly. In these methods, analytical expressions of the instantaneous light flicker sensation are obtained beginning from a voltage waveform and these expressions are used to obtain a flicker estimation method based on the IEC (International Electrotechnical Commission) flickermeter. First method is a spectral decomposition based approach using DFT to estimate the light flicker. The leakage effect of the DFT algorithm due to fundamental frequency variation is reduced by employing spectral amplitude correction procedure around the fundamental frequency. Second method is a Kalman filter based approach, in which the frequency domain components of the voltage waveform are obtained by Kalman filtering. Then these components are used to obtain the light flicker. Since the frequency decomposition is obtained by Kalman filtering, no leakage effect of the DFT is involved in case of frequency deviations which is an important advantage. Both methods are tested on both simulated data and field data obtained from three different EAF plants where the flicker level and frequency variation is considerably high. The comparison with the digital realization of the IEC flickermeter shows that the methods are successful in estimating light flicker with low computational complexity. The methods are especially useful for conditions such as disturbances and subsequent system transients where the system frequency deviates significantly, since the methods avoid the need for online sampling rate adjustment to prevent the DFT leakage effect.

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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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28

Vorhies, John T. "Low-complexity Algorithms for Light Field Image Processing." University of Akron / OhioLINK, 2020. http://rave.ohiolink.edu/etdc/view?acc_num=akron1590771210097321.

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29

Weaver, Michael B. "Performance comparison between three different bit allocation algorithms inside a critically decimated cascading filter bank." Diss., Online access via UMI:, 2009.

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Thesis (M.S.)--State University of New York at Binghamton, Thomas J. Watson School of Engineering and Applied Science, Department of Electrical and Computer Engineering, 2009.
Includes bibliographical references.

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Lee, Bong-Woon. "Applications of signal processing techniques in direct-sequence spread spectrum communication systems." Ohio : Ohio University, 1990. http://www.ohiolink.edu/etd/view.cgi?ohiou1173208101.

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31

Hursig, Robert E. "Robust Unconstrained Face Detection and Lip Localization using Gabor Filters." DigitalCommons@CalPoly, 2009. https://digitalcommons.calpoly.edu/theses/145.

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Automatic speech recognition (ASR) is a well-researched field of study aimed at augmenting the man-machine interface through interpretation of the spoken word. From in-car voice recognition systems to automated telephone directories, automatic speech recognition technology is becoming increasingly abundant in today’s technological world. Nonetheless, traditional audio-only ASR system performance degrades when employed in noisy environments such as moving vehicles. To improve system performance under these conditions, visual speech information can be incorporated into the ASR system, yielding what is known as audio-video speech recognition (AVASR). A majority of AVASR research focuses on lip parameters extraction within controlled environments, but these scenarios fail to meet the demanding requirements of most real-world applications. Within the visual unconstrained environment, AVASR systems must compete with constantly changing lighting conditions and background clutter as well as subject movement in three dimensions. This work proposes a robust still image lip localization algorithm capable of operating in an unconstrained visual environment, serving as a visual front end to AVASR systems. A novel Bhattacharyya-based face detection algorithm is used to compare candidate regions of interest with a unique illumination-dependent face model probability distribution function approximation. Following face detection, a lip-specific Gabor filter-based feature space is utilized to extract facial features and localize lips within the frame. Results indicate a 75% lip localization overall success rate despite the demands of the visually noisy environment.

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Lenz, Lutz Henning. "Automatic Tuning of Integrated Filters Using Neural Networks." PDXScholar, 1993. https://pdxscholar.library.pdx.edu/open_access_etds/4604.

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Component values of integrated filters vary considerably due to· manufacturing tolerances and environmental changes. Thus it is of major importance that the components of an integrated filter be electronically tunable. The method explored in this thesis is the transconductance-C-method. A method of realizing higher-order filters is to use a cascade structure of second-order filters. In this context, a method of tuning second-order filters becomes important The research objective of this thesis is to determine if the Neural Network methodology can be used to facilitate the filter tuning process for a second-order filter (realized via the transconductance-C-method). Since this thesis is, at least to the knowledge of the author, the first effort in this direction, basic principles of filters and of Neural Networks [1-22] are presented. A control structure is proposed which comprises three parts: the filter, the Neural Network, and a digital spectrum analyzer. The digital spectrum analyzer sends a test signal to the filter and measures the magnitude of the output at 49 frequency samples. The Neural Network part includes a memory that stores the 49 sampled values of the nominal spectrum. ·A comparator subtracts the latter values from the measured (actual) values, and feeds them as input to the Neural Network. The outputs of the Neural Network are the values of the percentage tuning amount The adjusting device, which is envisioned as a component of the filter itself, translates the output of the Neural Network to adjustments in the value of the filter's transconductances. Experimental results provide a demonstration that the Neural Network methodology can be usefully applied to the above problem context. A feedforward, singlehidden layer Backpropagation Network reduces the manufacturing errors of up to 85% for the pole frequency and of up to 41% for the quality factor down to less than approximately 5% each. It is demonstrated that the method can be iterated to further reduce the error.

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Angélico, Bruno Augusto. "Sistemas de banda ultralarga com pré-processamento." Universidade de São Paulo, 2010. http://www.teses.usp.br/teses/disponiveis/3/3142/tde-20082010-164755/.

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A resposta impulsiva do canal de um sistema de banda ultralarga típico é caracterizada pelo elevado número de percursos discerníveis. Dessa forma, para uma recepção eficiente, a energia espalhada nessas componentes multipercurso deve ser de alguma forma combinada. Considerando o enlace direto (downlink) de uma rede pessoal de curto alcance, assume-se que o ponto de acesso possui uma capacidade de processamento maior do que os dispositivos portáteis a ele conectados, tais como câmeras fotográficas, celulares e aparelhos de MP3. Este trabalho se concentra no estudo de esquemas de pré-processamento em ambientes mono e multiusuário, com vistas a combinar eficientemente a energia espalhada nas componentes multipercurso do canal e, consequentemente, combater a autointerferência e a interferência entre usuários, sem agregar muito custo computacional ao receptor (dispositivos portáteis da rede). Com isso, boa parte da complexidade é transferida para o transmissor (ponto de acesso), de forma que o receptor necessite apenas de um detector convencional, ou então de um detector convencional seguido de processamento adicional de complexidade moderada para mitigar a interferência residual.
The channel impulse response of a typical ultra wideband system is characterized by a large number of resolvable paths. For a efficient reception, the energy spread over the multipath components has to be somehow combined. Considering the downlink of a wireless personal area network, the access point is assumed to have a good hardware capacity when compared to the portable devices of the network, such as digital cameras, cell phones and MP3 players. This work focuses on preprocessing schemes that are able to combine efficiently the multipath components, and to combat self and multiuser interference without increasing the computational cost at the receiver (portable devices) substantially. Hence, most of the complexity is transferred to the transmitter (access point) in such a way that the receiver needs only a conventional detector or a conventional detector followed by a moderated complexity processing in order to mitigate the residual interference.

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Neto, Fernando Gonçalves de Almeida. "Análise de filtros digitais implementados em aritmética de ponto fixo usando cadeias de Markov." Universidade de São Paulo, 2011. http://www.teses.usp.br/teses/disponiveis/3/3142/tde-06052011-142814/.

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Uma forma de se reduzir o custo (em termos tanto de área de chip quanto de consumo de energia) de algoritmos de processamento de sinais é empregar aritmética de ponto fixo, usando o menor número de bits possível para se representar as variáveis e coeficientes necessários. Com isso, consegue-se reduzir a complexidade do hardware, levando a economias de energia e de área de chip em circuitos dedicados. A escolha do nível de quantização a que cada variável deve ser submetida depende de se conhecer o efeito da quantização de cada variável nas saídas do sistema, o que pode ser conseguido através de simulações (em geral lentas) ou por métodos analíticos. Este documento propõe avanços a uma nova metodologia de análise de algoritmos para processamento digital de sinais implementados em aritmética de ponto fixo, usando modelos baseados em cadeias de Markov. As contribuições desta dissertação são as seguintes: Filtros IIR de primeira e de segunda ordem são analisados via cadeia de Markov, pressupondo que a entrada possui uma função densidade de probabilidade conhecida. O modelo é desenvolvido de forma geral, de forma que pode ser considerada uma função de densidade de probabilidade qualquer. A saída dos filtros é usada para definir os estados da cadeia. O modelo via cadeia de Markov para o coeficiente do algoritmo LMS unidimensional é estendido para entrada correlacionada. Nesse caso, os estados passam a ser descritos em termos do coeficiente e do da entrada anterior. Um exemplo assumido função de densidade de probabilidade de entrada gaussiana para o filtro adaptativo é apresentado.
The implementation cost of signal processing algorithms may be reduced by using fixed-point arithmetic with the smallest possible word-length for each variable or parameter. This allows the designer to reduce hardware complexity, leading to economy of energy and chip area in dedicated circuits. The choice of word-length depends on the determination of the effect at the output of the quantization of each variable, which may be obtained through simulations (generally slow) or through analytical methods. This document proposes new advances to a new analysis method for digital signal processing algorithms implemented in fixed-point arithmetic, based on Markov chain models. Our contributions are the following: A Markov chain model is used to study first and second order IIR filters for an known input density probability function. The model is general and can be applied for any probability function. We use the output of the filters to define the states of the Markov chain. The unidimensional LMS Markov chain model is extended to correlated input. The states are defined by a pair considering the coefficient and the previous input and an example assuming Gaussian-distributed input is presented.

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Srinivasan, Venkatesh. "Programmable Analog Techniques For Precision Analog Circuits, Low-Power Signal Processing and On-Chip Learning." Diss., Georgia Institute of Technology, 2006. http://hdl.handle.net/1853/11588.

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In this work, programmable analog techniques using floating-gate transistors have been developed to design precision analog circuits, low-power signal processing primitives and adaptive systems that learn on-chip. Traditional analog implementations lack programmability with the result that issues such as mismatch are corrected at the expense of area. Techniques have been proposed that use floating-gate transistors as an integral part of the circuit of interest to provide both programmability and the ability to correct for mismatch. Traditionally, signal processing has been performed in the digital domain with analog circuits handling the interface with the outside world. Such a partitioning of responsibilities is inefficient as signal processing involves repeated multiplication and addition operations that are both very power efficient in the analog domain. Using programmable analog techniques, fundamental signal processing primitives such as multipliers have been developed in a low-power fashion while preserving accuracy. This results in a paradigm shift in signal processing. A co-operative analog/digital signal processing framework is now possible such that the partitioning of tasks between the analog and digital domains is performed in a power efficient manner. Complex signal processing tasks such as adaptive filtering that learn the weight coefficients are implemented by exploiting the non-linearities inherent with floating-gate programming. The resulting floating-gate synapses are compact, low-power and offer the benefits of non-volatile weight storage. In summary, this research involves developing techniques for improving analog circuit performance and in developing power-efficient techniques for signal processing and on-chip learning.

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Condori, Reynaldo Pampa. "Cancelamento de retorno local em aparelhos telefônicos para deficientes auditivos." Universidade de São Paulo, 2012. http://www.teses.usp.br/teses/disponiveis/3/3142/tde-04072013-171215/.

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Em algumas ocasiões o uso de aparelhos é necessário para melhorar a audição de deficientes auditivos. Para melhorar a clareza das ligações telefônicas dessas pessoas, há interesse em aparelhos telefônicos que desempenham função semelhante aos aparelhos de audição (hearing aids), amplificando adequadamente o sinal recebido do outro lado da conexão. Um dos maiores problemas com tais aparelhos é que a malha fechada que inclui o retorno local na híbrida telefônica e o acoplamento acústico entre o alto-falante e o microfone do aparelho pode se tornar instável devido à introdução da amplificação, dando origem a microfonia. Uma abordagem típica deste problema nos aparelhos de audição é feita mediante o cancelamento adaptativo do eco acústico. No caso sendo considerado aqui, porém, também é possível eliminar a microfonia fazendo o cancelamento adaptativo do retorno local na híbrida, abordagem que é mais simples, devido à resposta impulsiva mais curta a ser compensada e sua menor variação temporal. Portanto, esta dissertação tem como objetivo o desenvolvimento de um cancelador de retorno local usando filtragem adaptativa para fins de impedir a microfonia num aparelho telefônico para deficientes auditivos. Para uma efetiva compensação de um determinado grau de deficiência auditiva, será visto que pode ser desejável em algum momento introduzir até uma certa quantidade de ganho em alguma frequência, sendo que a viabilidade disso depende da efetividade do cancelamento do retorno local, é definida uma medida de desempenho dos filtros adaptativos usados em termos da amplificação máxima que pode ser introduzida graças ao cancelamento do retorno local sem provocar microfonia. Considerando esta medida de desempenho, foi definido um objetivo de 55 dB de amplificação máxima a ser alcançado pelo cancelamento do retorno local. Verificamos então que o uso dos algoritmos LMS (least mean square), e-NLMS (normalized least mean square ) e RLS (recursive least squares) não alcança este objetivo com sinal de fala como entrada. Visto isto, a adaptação antes da conversação é avaliada com sinal branco gaussiano como entrada, acrescentando-se um filtro notch para eliminar o tom de discar. Os resultados mostram que o algoritmo LMS é suficiente para alcançar o objetivo mencionado.
On some occasions the use of apparatus are needed to improve the deafness of hearing impaired. To improve the clarity of the telephonic connections of these people, there is interest in telephone sets that exert similar function to the hearing aids, properly amplifying the received signal of the another side of the connection. One of the main problems with such apparatus is that the closed loop that includes local return in the telephone hybrid and acoustic coupling between the capsule and the microphone of the telephone set would become unstable due to the introduction of amplification, producing howling. A typical approach to this problem in hearing aids is made by the adaptive acoustic echo cancellation. In this case, however, it is also possible to eliminate howling by the adaptive local return cancellation in the hybrid, which is a simpler approach, due to shorter impulse response to be compensated and their lower temporal variation. Therefore, the goal of this dissertation is to develop a local return canceller using adaptive filtering for the purpose of preventing howling in a telephone set for hearing impaired. For an effective compensation of a certain degree of hearing impairment, it will be seen that it may be desirable at some point to introduce up to a certain amount of gain at some frequency. Whereas the viability of this depends on the effectiveness of the cancellation of the local return, a performance measure of used adaptive filters in terms of the maximum amplification which may be released due to the cancellation of the local return is defined without causing howling. Considering this performance measure was set a goal of 55 dB of maximum gain to be achieved by the local return cancelling. We verify then that the use of the LMS (least mean square), e-NLMS (normalized least mean square) and RLS (recursive least squares) algorithms doesn\'t attain this goal with speech signal as input. Therefore, adaptation before the conversation is evaluated, with white gaussian signal as input, adding a notch filter to eliminate the dial tone. The results show that the LMS algorithm is sufficient to achieve the mentioned goal.

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Luwes, Nicolaas Johannes. "Massabepaling van bewegende voorwerpe op 'n vervoerband met behulp van DSP-tegnieke." Thesis, Bloemfontein : Central University of Technology, Free State, 2004. http://hdl.handle.net/11462/56.

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Thesis(M. Tech.) - Central University of Technology, Free State, 2004
Growing markets leads to an increase in production. In these modern industries, weight measurement is of high priority. Weight measurement instrumentation is used for quality control, as well as for effective process control. Ineffective instrumentation with inaccurate data will influence the production process and profit margins negatively. Experimental data is gathered from an angled load cell, placed as a crossover between two conveyer belts. A weight measurement instrument with the ability to acquire accurate measurement of individual, moving parts is produced with the aid of DSP techniques. This was accomplished by analyzing the frequency spectrum for the undesirable signals with the use of Wavelets transformations (WT) and Fourier transformations (FT). After these undesired signals were identified a digital filter was designed to remove the undesired signals. Repetition of performance is achieved by the automatic zeroing of the instrument after every individual measurement. This weight measurement instrumentation also has the ability to store data consisting of the amount of objects and their individual weights. This instrument can also determine the material of which an object is made of. This is done by calculating the friction coefficient. This function has the ability to effectively identify between iron and rubber components irrespective of their mass or area.

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Fernandes, Anderson Luiz. "Arquitetura híbrida com DSP e FPGA para implementação de controladores de filtros ativos de potência." Universidade Tecnológica Federal do Paraná, 2016. http://repositorio.utfpr.edu.br/jspui/handle/1/1785.

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A presença de cargas não-lineares em um ponto do sistema de distribuição pode deformar a forma de onda de tensão devido ao consumo de correntes não senoidais. O uso de filtros ativos de potência permite uma redução significativa do conteúdo harmônico da corrente de alimentação. Entretanto, as estruturas digitais de controle para estes filtros, particularmente o cálculo das correntes de referência, pode necessitar de processamento de alto desempenho. Neste trabalho se propõe o desenvolvimento de estruturas de controle com alto desempenho de processamento, para aplicação em filtros ativos de potência. Neste sentido, é considerada uma arquitetura que permite processamento paralelo utilizando dispositivos lógicos programáveis. A estrutura desenvolvida utiliza um modelo híbrido com um DSP e uma FPGA. O DSP é utilizado para aquisição de sinais de tensão e corrente, controladores adicionais relacionados a fundamental e acionamento PWM. A FPGA é utilizada para o processamento intensivo do sinal de compensação de harmônicas. Desta forma, através da análise experimental são obtidas reduções significativas nos tempos de processamento comparadas as abordagens tradicionais utilizando somente DSP. Os resultados experimentais validam a estrutura projetada e são comparados com outras arquiteturas relatadas na literatura.
The presence of non-linear loads at a point in the distribution system may deform voltage waveform due to the consumption of non-sinusoidal currents. The use of active power filters allows significant reduction of the harmonic content in the supply current. However, the processing of digital control structures for these filters may require high performance hardware, particularly for reference currents calculation. This work describes the development of hardware structures with high processing capability for application in active power filters. In this sense, it considers an architecture that allows parallel processing using programmable logic devices. The developed structure uses a hybrid model using a DSP and an FPGA. The DSP is used for the acquisition of current and voltage signals, calculation of fundamental current related controllers and PWM generation. The FPGA is used for intensive signal processing, such as the harmonic compensators. In this way, from the experimental analysis, significant reductions of the processing time are achieved when compared to traditional approaches using only DSP. The experimental results validate the designed structure and these results are compared with other ones from architectures reported in the literature.

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Yang, Zhenghong. "Joint time frequency analysis of Global Positioning System (GPS) multipath signals." Ohio : Ohio University, 1998. http://www.ohiolink.edu/etd/view.cgi?ohiou1176234303.

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Bengtsson, Fredrik, and Rikard Berglund. "Digital compensation of distortion in audio systems." Thesis, Linköping University, Department of Electrical Engineering, 2010. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-56392.

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The advancements of computational power in low cost FPGAs are giving the opportunityto implement real-time compensation of loudspeakers and audio systems. The need for expensive commercial audio systems is reduced when the fidelity ofmuch cheaper audio systems easily can be improved by real-time compensation. The topic of this thesis is to investigate and evaluate methods for digital compensationof distortion in audio systems. More specifically, a VHDL module isimplemented to, when necessary, alleviate the problem of drastically deterioratingfidelity of the bass appearing when the input power is too high.

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Akpa, Marcellin. "Tree structure filter bank for wideband signal processing." Thesis, University of Ottawa (Canada), 1995. http://hdl.handle.net/10393/10407.

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A N-parallel branches maximally decimated filter bank is generally implemented using the polyphase components implementation. In this case, a N-th band lowpass filter is designed and its polyphase components are derived to constitute the branch 'subfilters.' This approach uses a N x N FFT matrix that will be the source of the complex (numbers) operations. Obviously, when the number of branches is equal to 2, the computations remain real. In a tree structure filter bank, the computations remain real with or without polyphase implementation. When the polyphase implementation is used, the branch signals at each stage are computed using a set of 2 x 2 FFT matrices leading to real computations. In this thesis, a new implementation approach based on the tree structured is proposed. The derivation of the structure is based on the equivalent parallel structure implementation of the tree structured filter bank. It uses the polyphase components of a given half-band lowpass filter (real coefficients) followed by a N x N Hadamard matrix. The computations, as in the original tree structured filter bank, remain real. A simplified version of the structure is a 'tree structure' followed by an N x N Hadamard matrix. A comparison between this new structure and the N parallel branch maximally decimated filter bank is made based on reconstruction error, computation complexity and processing delay.

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Azurdia, Meza Cesar Augusto, and Mohamadi Yaqub Jon. "Implementation of the LMS Algorithm for Noise Cancellation on Speech Using the ARM LPC2378 Processor." Thesis, Växjö University, School of Mathematics and Systems Engineering, 2009. http://urn.kb.se/resolve?urn=urn:nbn:se:vxu:diva-5777.

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On this thesis project, the LMS algorithm has been applied for speech noise filteringand different behaviors were tested under different circumstances by using Matlabsimulations and the LPC2378 ARM Processor, which does the task of filtering in realtime. The thesis project is divided into two parts: the theoretical and practical part.

In the theoretical part there is a brief description of the different aspects of signalprocessing systems, filter theory, and a general description of the Least-Mean-SquareAdaptive Filter Algorithm.

In the practical part of the report a general description of the procedure will besummarized, the results of the tests that were conducted will be analyzed, a generaldiscussion of the problems that were encounter during the simulations will be mention,and suggestion for the problems will be given.

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Okullo-Oballa, Thomas Samuel. "Systolic realization of multirate digital filters." Thesis, [Hong Kong] : University of Hong Kong, 1988. http://sunzi.lib.hku.hk/hkuto/record.jsp?B12433998.

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Sridharan, M. K. "Subband Adaptive Filtering Algorithms And Applications." Thesis, Indian Institute of Science, 2000. http://hdl.handle.net/2005/266.

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In system identification scenario, the linear approximation of the system modelled by its impulse response, is estimated in real time by gradient type Least Mean Square (LMS) or Recursive Least Squares (RLS) algorithms. In recent applications like acoustic echo cancellation, the order of the impulse response to be estimated is very high, and these traditional approaches are inefficient and real time implementation becomes difficult. Alternatively, the system is modelled by a set of shorter adaptive filters operating in parallel on subsampled signals. This approach, referred to as subband adaptive filtering, is expected to reduce not only the computational complexity but also to improve the convergence rate of the adaptive algorithm. But in practice, different subband adaptive algorithms have to be used to enhance the performance with respect to complexity, convergence rate and processing delay. A single subband adaptive filtering algorithm which outperforms the full band scheme in all applications is yet to be realized. This thesis is intended to study the subband adaptive filtering techniques and explore the possibilities of better algorithms for performance improvement. Three different subband adaptive algorithms have been proposed and their performance have been verified through simulations. These algorithms have been applied to acoustic echo cancellation and EEG artefact minimization problems. Details of the work To start with, the fast FIR filtering scheme introduced by Mou and Duhamel has been generalized. The Perfect Reconstruction Filter Bank (PRFB) is used to model the linear FIR system. The structure offers efficient implementation with reduced arithmetic complexity. By using a PRFB with non adjacent filters non overlapping, many channel filters can be eliminated from the structure. This helps in reducing the complexity of the structure further, but introduces approximation in the model. The modelling error depends on the stop band attenuation of the filters of the PRFB. The error introduced due to approximation is tolerable for applications like acoustic echo cancellation. The filtered output of the modified generalized fast filtering structure is given by (formula) where, Pk(z) is the main channel output, Pk,, k+1 (z) is the output of auxiliary channel filters at the reduced rate, Gk (z) is the kth synthesis filter and M the number of channels in the PRFB. An adaptation scheme is developed for adapting the main channel filters. Auxiliary channel filters are derived from main channel filters. Secondly, the aliasing problem of the classical structure is reduced without using the cross filters. Aliasing components in the estimated signal results in very poor steady state performance in the classical structure. Attempts to eliminate the aliasing have reduced the computation gain margin and the convergence rate. Any attempt to estimate the subband reference signals from the aliased subband input signals results in aliasing. The analysis filter Hk(z) having the following antialiasing property (formula) can avoid aliasing in the input subband signal. The asymmetry of the frequency response prevents the use of real analysis filters. In the investigation presented in this thesis, complex analysis filters and real'synthesis filters are used in the classical structure, to reduce the aliasing errors and to achieve superior convergence rate. PRFB is traditionally used in implementing Interpolated FIR (IFIR) structure. These filters may not be ideal for processing an input signal for an adaptive algorithm. As third contribution, the IFIR structure is modified using discrete finite frames. The model of an FIR filter s is given by Fc, with c = Hs. The columns of the matrix F forms a frame with rows of H as its dual frame. The matrix elements can be arbitrary except that the transformation should be implementable as a filter bank. This freedom is used to optimize the filter bank, with the knowledge of the input statistics, for initial convergence rate enhancement . Next, the proposed subband adaptive algorithms are applied to acoustic echo cancellation problem with realistic parameters. Speech input and sufficiently long Room Impulse Response (RIR) are used in the simulations. The Echo Return Loss Enhancement (ERLE)and the steady state error spectrum are used as performance measures to compare these algorithms with the full band scheme and other representative subband implementations. Finally, Subband adaptive algorithm is used in minimization of EOG (Electrooculogram) artefacts from measured EEG (Electroencephalogram) signal. An IIR filterbank providing sufficient isolation between the frequency bands is used in the modified IFIR structure and this structure has been employed in the artefact minimization scheme. The estimation error in the high frequency range has been reduced and the output signal to noise ratio has been increased by a couple of dB over that of the fullband scheme. Conclusions Efforts to find elegant Subband adaptive filtering algorithms will continue in the future. However, in this thesis, the generalized filtering algorithm could offer gain in filtering complexity of the order of M/2 and reduced misadjustment . The complex classical scheme offered improved convergence rate, reduced misadjustment and computational gains of the order of M/4 . The modifications of the IFIR structure using discrete finite frames made it possible to eliminate the processing delay and enhance the convergence rate. Typical performance of the complex classical case for speech input in a realistic scenario (8 channel case), offers ERLE of more than 45dB. The subband approach to EOG artefact minimization in EEG signal was found to be superior to their fullband counterpart. (Refer PDF file for Formulas)

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Ödmark, Fredrik. "Model based pulse shaping for detection of gamma rays." Thesis, Luleå tekniska universitet, Institutionen för system- och rymdteknik, 2017. http://urn.kb.se/resolve?urn=urn:nbn:se:ltu:diva-66637.

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To analyse drill samples in a mine, a scanner that uses a gamma ray detector can be used. The scanner can analyse the drill sample to quickly see the elements present in the sample without destroying it. To improve the performance of the scanner, the electric signal from the detector needs to be less noisy, and different pulse shaping methods, filters and smoothers can be used on the sampled data to achieve an improved performance. In this master thesis, the electric noise model of the electronics around the detector was modeled, and analysed. Different pulse shaping method, filters and smoothers was also tested to see which method gave the best performance in FWHM sense. The Full Width at Half Maximum (FWHM) is the energy resolution of a detector, and is defined as the full width of a photopeak at the half maximum. The noise model of the schematic for the preamplifier was made by hand with support from MATLAB. The resulting noise model was compared between MATLAB and LTspice, and the conclusion is that the JFET is the main contributor of the significant noise, contributing to 98 % of the total noise at 10 GHz. The adopted filters and pulse shaping method are, matched filter, custom filter, CR-RC shaping, mean filter, median filter and clustering. The results from the tests indicated that custom filter with a FWHM of 1.96 keV and CR-RC with a FWHM of 1.67 keV shaping were more accurate than the matched filter with the FWHM of 5.1 keV. But the results also showed that it is important to take into account the waveform variance, due to inherent properties in the detector, with this consideration the FWHM of CR-RC shaper was improved from 2.29 keV to 1.67 keV. The clustering method was the most promising method but due to time constraints this method was never fully tested and no FWHM value was achieved.

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Owens, Peter. "Advanced signal processing of high resolution electrocardiograms." Thesis, University of Sussex, 1997. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.361399.

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Kavalov, Dimitar A. "Surface acoustic wave neural networks for RF signal processing." Thesis, Oxford Brookes University, 2002. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.249406.

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Nobbe, Andrea. "Pitch perception and signal processing in electric hearing." Diss., lmu, 2004. http://nbn-resolving.de/urn:nbn:de:bvb:19-31100.

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McWhorter, Francis L. "Novel structures for very fast adaptive filters." Ohio : Ohio University, 1990. http://www.ohiolink.edu/etd/view.cgi?ohiou1173322289.

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Wacey, Graham. "Algorithms and architectures for primitive operator digital signal processing." Thesis, University of Bristol, 1996. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.388043.

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Dissertations / Theses on the topic 'Signal processing Electric filters'

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