Dissertations / Theses on the topic 'Conditioning and processing electrical signal'
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Valero, Daniel. "Wireless Signal Conditioning." Thesis, University of North Texas, 2016. https://digital.library.unt.edu/ark:/67531/metadc862776/.
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This thesis presents a new approach to extend and reduce the transmission range in wireless systems. Conditioning is defined as purposeful electromagnetic interference that affects a wireless signal as it propagates through the air. This interference can be used constructively to enhance a signal and increase its energy, or destructively to reduce energy. The constraints and limitations of the technology are described as a system model, and a flow chart is used to describe the circuit process. Remaining theoretical in nature, practical circuit implementations are foregone in the interest of elementary simulations depicting the interactions of modulated signals as they experience phase mismatch. Amplitude modulation and frequency modulation are explored with using both positive and negative conditioning, and conclusions to whether one is more suitable than the other are made.
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Lee, Li 1975. "Distributed signal processing." Thesis, Massachusetts Institute of Technology, 2000. http://hdl.handle.net/1721.1/86436.
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Eldar, Yonina Chana 1973. "Quantum signal processing." Thesis, Massachusetts Institute of Technology, 2001. http://hdl.handle.net/1721.1/16805.
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Thesis (Ph. D.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, February 2002.Includes bibliographical references (p. 337-346).
This electronic version was submitted by the student author. The certified thesis is available in the Institute Archives and Special Collections.
Quantum signal processing (QSP) as formulated in this thesis, borrows from the formalism and principles of quantum mechanics and some of its interesting axioms and constraints, leading to a novel paradigm for signal processing with applications in areas ranging from frame theory, quantization and sampling methods to detection, parameter estimation, covariance shaping and multiuser wireless communication systems. The QSP framework is aimed at developing new or modifying existing signal processing algorithms by drawing a parallel between quantum mechanical measurements and signal processing algorithms, and by exploiting the rich mathematical structure of quantum mechanics, but not requiring a physical implementation based on quantum mechanics. This framework provides a unifying conceptual structure for a variety of traditional processing techniques, and a precise mathematical setting for developing generalizations and extensions of algorithms. Emulating the probabilistic nature of quantum mechanics in the QSP framework gives rise to probabilistic and randomized algorithms. As an example we introduce a probabilistic quantizer and derive its statistical properties. Exploiting the concept of generalized quantum measurements we develop frame-theoretical analogues of various quantum-mechanical concepts and results, as well as new classes of frames including oblique frame expansions, that are then applied to the development of a general framework for sampling in arbitrary spaces. Building upon the problem of optimal quantum measurement design, we develop and discuss applications of optimal methods that construct a set of vectors.
(cont.) We demonstrate that, even for problems without inherent inner product constraints, imposing such constraints in combination with least-squares inner product shaping leads to interesting processing techniques that often exhibit improved performance over traditional methods. In particular, we formulate a new viewpoint toward matched filter detection that leads to the notion of minimum mean-squared error covariance shaping. Using this concept we develop an effective linear estimator for the unknown parameters in a linear model, referred to as the covariance shaping least-squares estimator. Applying this estimator to a multiuser wireless setting, we derive an efficient covariance shaping multiuser receiver for suppressing interference in multiuser communication systems.
by Yonina Chana Eldar.
Ph.D.
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Vasconcellos, Brett W. (Brett William) 1977. "Parallel signal-processing for everyone." Thesis, Massachusetts Institute of Technology, 2000. http://hdl.handle.net/1721.1/9097.
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Thesis (M.Eng.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 2000.Includes bibliographical references (p. 65-67).
We designed, implemented, and evaluated a signal-processing environment that runs on a general-purpose multiprocessor system, allowing easy prototyping of new algorithms and integration with applications. The environment allows the composition of modules implementing individual signal-processing algorithms into a functional application, automatically optimizing their performance. We decompose the problem into four independent components: signal processing, data management, scheduling, and control. This simplifies the programming interface and facilitates transparent parallel signal processing. For tested applications, our system both runs efficiently on single-processors systems and achieves near-linear speedups on symmetric-multiprocessor (SMP) systems.
by Brett W. Vasconcellos.
M.Eng.
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Baran, Thomas A. (Thomas Anthony). "Conservation in signal processing systems." Thesis, Massachusetts Institute of Technology, 2012. http://hdl.handle.net/1721.1/74991.
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Thesis (Ph. D.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 2012.Cataloged from PDF version of thesis.
Includes bibliographical references (p. 205-209).
Conservation principles have played a key role in the development and analysis of many existing engineering systems and algorithms. In electrical network theory for example, many of the useful theorems regarding the stability, robustness, and variational properties of circuits can be derived in terms of Tellegen's theorem, which states that a wide range of quantities, including power, are conserved. Conservation principles also lay the groundwork for a number of results related to control theory, algorithms for optimization, and efficient filter implementations, suggesting potential opportunity in developing a cohesive signal processing framework within which to view these principles. This thesis makes progress toward that goal, providing a unified treatment of a class of conservation principles that occur in signal processing systems. The main contributions in the thesis can be broadly categorized as pertaining to a mathematical formulation of a class of conservation principles, the synthesis and identification of these principles in signal processing systems, a variational interpretation of these principles, and the use of these principles in designing and gaining insight into various algorithms. In illustrating the use of the framework, examples related to linear and nonlinear signal-flow graph analysis, robust filter architectures, and algorithms for distributed control are provided.
by Thomas A. Baran.
Ph.D.
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South, Colin R. "Signal processing in a loudspeaking telephone." Thesis, Aston University, 1985. http://publications.aston.ac.uk/8053/.
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One of the major problems associated with communication via a loudspeaking telephone (LST) is that, using analogue processing, duplex transmission is limited to low-loss lines and produces a low acoustic output. An architectural for an instrument has been developed and tested, which uses digital signal processing to provide duplex transmission between a LST and a telopnone handset over most of the B.T. network. Digital adaptive-filters are used in the duplex LST to cancel coupling between the loudspeaker and microphone, and across the transmit to receive paths of the 2-to-4-wire converter. Normal movement of a person in the acoustic path causes a loss of stability by increasing the level of coupling from the loudspeaker to the microphone, since there is a lag associated the adaptive filters learning about a non-stationary path, Control of the loop stability and the level of sidetone heard by the hadset user is by a microprocessoe, which continually monitors the system and regulates the gain. The result is a system which offers the best compromise available based on a set of measured parameters. A theory has been developed which gives the loop stability requirements based on the error between the parameters of the filter and those of the unknown path. The programme to develope a low-cost adaptive filter in LST produced a low-cost adaptive filter in LST produced a unique architecture which has a number of features not available in any similar system. These include automatic compensation for the rate of adaptation over a 36 dB range of output level, , 4 rates of adaptation (with a maximum of 465 dB/s), plus the ability to cascade up to 4 filters without loss o performance. A complex story has been developed to determine the adptation which can be achieved using finite-precision arithmatic. This enabled the development of an architecture which distributed the normalisation required to achieve optimum rate of adaptation over the useful input range. Comparison of theory and measurement for the adaptive filter show very close agreement. A single experimental LST was built and tested on connections to hanset telephones over the BT network. The LST demonstrated that duplex transmission was feasible using signal processing and produced a more comfortable means of communication beween people than methods emplying deep voice-switching to regulate the local-loop gain. Although, with the current level of processing power, it is not a panacea and attention must be directed toward the physical acoustic isolation between loudspeaker and microphone.
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Boufounos, Petros T. 1977. "Signal processing for DNA sequencing." Thesis, Massachusetts Institute of Technology, 2002. http://hdl.handle.net/1721.1/17536.
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Thesis (M.Eng. and S.B.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 2002.Includes bibliographical references (p. 83-86).
DNA sequencing is the process of determining the sequence of chemical bases in a particular DNA molecule-nature's blueprint of how life works. The advancement of biological science in has created a vast demand for sequencing methods, which needs to be addressed by automated equipment. This thesis tries to address one part of that process, known as base calling: it is the conversion of the electrical signal-the electropherogram--collected by the sequencing equipment to a sequence of letters drawn from ( A,TC,G ) that corresponds to the sequence in the molecule sequenced. This work formulates the problem as a pattern recognition problem, and observes its striking resemblance to the speech recognition problem. We, therefore, propose combining Hidden Markov Models and Artificial Neural Networks to solve it. In the formulation we derive an algorithm for training both models together. Furthermore, we devise a method to create very accurate training data, requiring minimal hand-labeling. We compare our method with the de facto standard, PHRED, and produce comparable results. Finally, we propose alternative HMM topologies that have the potential to significantly improve the performance of the method.
by Petros T. Boufounos.
M.Eng.and S.B.
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Ponnala, Lalit. "Analysis of Genetic Translation using Signal Processing." NCSU, 2007. http://www.lib.ncsu.edu/theses/available/etd-02072007-174200/.
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A series of free energy estimates can be calculated from the ribosome's progressive interaction with mRNA sequences during the process of translation elongation in eubacteria. A sinusoidal pattern of roughly constant phase has been detected in these free energy signals. Frameshifts of the +1 type occur when the ribosome skips an mRNA base in the 5'-3' direction, and can be associated with local phase-shifts in the free energy signal. We propose a mathematical model that captures the mechanism of frameshift based on the information content of the signal parameters and the relative abundance of tRNA in the bacterial cell. The model shows how translational speed can modulate translational accuracy to accomplish programmed +1 frameshifts and could have implications for the regulation of translational efficiency. Results are presented using experimentally verified frameshift genes across eubacteria.
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Wang, Yingying. "ADVANCED ANALOG SIGNAL PROCESSING FOR WIRELESS COMMUNICATIONS." Case Western Reserve University School of Graduate Studies / OhioLINK, 2020. http://rave.ohiolink.edu/etdc/view?acc_num=case1585776428631869.
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Ali-Bakhshian, Mohammad. "Digital processing of analog information adopting time-mode signal processing." Thesis, McGill University, 2013. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=114237.
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As CMOS technologies advance to 22-nm dimensions and below, constructing analog circuits in such advanced processes suffers many limitations, such as reduced signal swings, sensitivity to thermal noise effects, loss of accurate switching functions, to name just a few. Time-Mode Signal Processing (TMSP) is a technique that is believed to be well suited for solving many of these challenges. It can be defined as the detection, storage, and manipulation of sampled analog information using time-mode variables. One of the important advantages of TMSP is the ability to realize analog functions using digital logic structures. This technique has a long history of application in electronics; however, due to lack of some fundamental functions, the use of TM variables has been mostly limited to intermediate stage processing and it has been always associated with voltage/current-to-time and time-to-voltage/current conversion. These conversions necessitate the inclusion of analog blocks that contradict the digital advantage of TMSP. In this thesis, an intensive research has been presented that provides an appropriate foundation for the development of TMSP as a general processing tool. By proposing the new concept of delay interruption, a completely new asynchronous approach for the manipulation of TM variables is suggested. As a direct result of this approach, practical techniques for storage, addition and subtraction of time-mode variables are presented. To Extend the digital implementation of TMSP to a wider range of applications, the comprehensive design of a unity gain dual-path time-to-time integrator (accumulator) is demonstrated. This integrator is then used to implement a digital second-order delta-sigma modulator. Finally, to demonstrate the advantage of TMSP, a very low power and compact tunable interface for capacitive sensors is presented that is composed of a number of delay blocks associated with typical logic gates. All the proposed theories are supported by experimental results and post-layout simulations.The emphasis on the digital construction of the proposed circuits has been the first priority of this thesis. Having the building blocks implemented with a digital structure, provides the feasibility of a simple, synthesizable, and reconfigurable design where affordable circuit calibrations can be adopted to remove the effects of process variations.Les technologies CMOS progressant vers les procédés 22 nm et au delà, la abrication des circuits analogiques dans ces technologies se heurte a de nombreuses limitations. Entre autres limitations on peut citer la réduction d'amplitude des signaux, la sensibilité aux effets du bruit thermique et la perte de fonctions précises de commutation. Le traitement de signal en mode temps (TMSP pour Time-Mode Signal Processing) est une technique que l'on croit être bien adapté pour résoudre un grand nombre de problèmes relatifs a ces limitations. TMSP peut être défini comme la détection, le stockage et la manipulation de l'information analogique échantillonnée en utilisant des quantités de temps comme variables. L'un des avantages importants de TMSP est la capacité à réaliser des fonctions analogiques en utilisant des structures logiques digitales. Cette technique a une longue histoire en terme d'application en électronique. Cependant, en raison du manque de certaines fonctions fondamentales, l'utilisation de variables en mode temps a été limitée à une utilisation comme étape intermédiaire dans le traitement d'un signal et toujours dans le contexte d'une conversion tension/courant-temps et temps-tension/courant. Ces conversions nécessitent l'inclusion de blocs analogiques qui vont a l'encontre de l'avantage numérique des TMSP. Cette thèse fournit un fondement approprié pour le développement de TMSP comme outil général de traitement de signal. En proposant le concept nouveau d'interruption de retard, une toute nouvelle approche asynchrone pour la manipulation de variables en mode temps est suggéré. Comme conséquence directe de cette approche, des techniques pratiques pour le stockage, l'addition et la soustraction de variables en mode temps sont présentées. Pour étendre l'implémentation digitale de TMSP à une large gamme d'applications, la conception d'un intégrateur (accumulateur) à double voie temps- à -temps est démontrée. cet intégrateur est ensuite utilisé pour implémenter un modulateur delta-sigma de second ordre.Enfin, pour démontrer l'avantage de TMSP, une Interface de très basse puissance, compacte et réglable pour capteurs capacitifs est présenté. Cette interface est composé d'un certain nombre de blocs de retard associés à des portes logiques typiques. Toutes les théories proposées sont soutenues par des résultats expérimentaux et des simulations post-layout. L'implémentation digitale des circuits proposés a été la première priorité de cette thèse. En effet, une implémentation des bloc avec des structures digitales permet des conceptions simples, synthétisable et reconfigurables où des circuits de calibration très abordables peuvent être adoptées pour éliminer les effets des variations de process.
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Kolb, John. "SIGNAL PROCESSING ABOUT A DISTRIBUTED DATA ACQUISITION SYSTEM." International Foundation for Telemetering, 2002. http://hdl.handle.net/10150/605610.
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International Telemetering Conference Proceedings / October 21, 2002 / Town & Country Hotel and Conference Center, San Diego, CaliforniaBecause modern data acquisition systems use digital backplanes, it is logical for more and more data processing to be done in each Data Acquisition Unit (DAU) or even in each module. The processing related to an analog acquisition module typically takes the form of digital signal conditioning for range adjust, linearization and filtering. Some of the advantages of this are discussed in this paper. The next stage is powerful processing boards within DAUs for data reduction and third-party algorithm development. Once data is being written to and from powerful processing modules an obvious next step is networking and decom-less access to data. This paper discusses some of the issues related to these types of processing.
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Singer, Andrew C. (Andrew Carl). "Signal processing and communication with solitons." Thesis, Massachusetts Institute of Technology, 1996. http://hdl.handle.net/1721.1/11011.
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Thesis (Ph. D.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 1996.Includes bibliographical references (p. 137-142).
by Andrew Carl Singer.
Ph.D.
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Su, Guolong Ph D. Massachusetts Institute of Technology. "Polynomial decomposition algorithms in signal processing." Thesis, Massachusetts Institute of Technology, 2013. http://hdl.handle.net/1721.1/82383.
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Thesis (S.M.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 2013.Cataloged from PDF version of thesis.
Includes bibliographical references (p. 107-109).
Polynomial decomposition has attracted considerable attention in computational mathematics. In general, the field identifies polynomials f(x) and g(x) such that their composition f(g(x)) equals or approximates a given polynomial h(x). Despite potentially promising applications, polynomial decomposition has not been significantly utilized in signal processing. This thesis studies the sensitivities of polynomial composition and decomposition to explore their robustness in potential signal processing applications and develops effective polynomial decomposition algorithms to be applied in a signal processing context. First, we state the problems of sensitivity, exact decomposition, and approximate decomposition. After that, the sensitivities of the composition and decomposition operations are theoretically derived from the perspective of robustness. In particular, we present and validate an approach to decrease certain sensitivities by using equivalent compositions, and a practical rule for parameter selection is proposed to get to a point that is near the minimum of these sensitivities. Then, new algorithms are proposed for the exact decomposition problems, and simulations are performed to make comparison with existing approaches. Finally, existing and new algorithms for the approximate decomposition problems are presented and evaluated using numerical simulations.
by Guolong Su.
S.M.
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Naidoo, Thoneshan. "Signal and image processing for electrical resistance tomography." Master's thesis, University of Cape Town, 2002. http://hdl.handle.net/11427/5140.
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Bibliography: leaves 139-150.Electrical Resistance Tomography (ERT) is in essence an imaging technique.In ERT current is injected into and removed from a vessel via paired electrodes. The resulting voltage measurements are captured between the remaining electrode pairs. The principle behind ERT is to map these boundary voltages into a conductivity distribution that represents the domain of the vessel. The author has coded a versatile reconstruction algorithm based on the Newton-Raphson algorithm. The knowledge gained by implementing the algorithm is documented in this thesis. The literature covers the basic aspects of two-dimensional and three-dimensional ERT. It is hoped that this thesis will create a greater interest in ERT at the University of Cape Town (UCT) and also act as a building block for further developments. The thesis starts by presenting the basic concepts of ERT such as the underlying equations, the various boundary measurement strategies and a global perspective of ERT. The nature of this thesis is on software reconstruction and in so doing information on the incorporation of the Finite Element Method in ERT is provided. The thesis goes on to provide information about the reconstruction algorithms, which incorporate regularization. A novel aspect of this thesis involves the calibration and pre-processing of boundary voltages. These concepts were conceptualised and developed during formal communications with Dr. Wilkinson (2002) and Randal (2002). The calibration schemes try to eliminate the potential errors that can arise inthe captured data thus allowing for a clearer image to be reconstructed, Electrical Resistance Tomography. This thesis further develops the idea of parallelizing the Newton-Raphson algorithm to increase the speed of the algorithm. Various schemes on how this parallelization is achievable are put forward.
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Wheland, David Stanford. "Signal processing methods for brain connectivity." Thesis, University of Southern California, 2014. http://pqdtopen.proquest.com/#viewpdf?dispub=3610033.
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Although the human brain has been studied for centuries, and the advent of non-invasive brain imaging modalities in the last century in particular has led to significant advances, there is much left to discover. Current neuroscientific theory likens the brain to a highly interconnected network whose behavior can be better understood by determining its network connections. Correlation, coherence, Granger causality, and blind source separation (BSS) are frequently used to infer this connectivity. Here I propose novel methods to improve their inference from neuroimaging data. Correlation and coherence suffer from being unable to differentiate between direct and indirect connectivity. While partial correlation and partial coherence can mitigate this problem, standard methods for calculating these measures result in significantly reduced statistical inference power and require greater numbers of samples. To address these drawbacks I propose novel methods based on a graph pruning algorithm that leverage the connectivity sparsity of the brain to improve the inference of partial correlation and partial coherence. These methods are demonstrated in applications. In particular, partial correlation is explored in both cortical thickness data from structural MR images and resting state data from functional MR images, and partial coherence is explored in invasive electrophysiological measurements in non-human primates. Granger causality is able to differentiate between direct and indirect connectivity by default and like partial coherence is readily applicable to time series. However unlike partial coherence, it uses the temporal ordering implied by the time series to infer a type of causality on the connectivity. Despite its differences, the inference of Granger causality can also be improved using a similar graph pruning algorithm, and I describe such an extension here. The method is also applied to explore electrophysiological interactions in non-human primate data. BSS methods seek to decompose a dataset into a linear mixture of sources such that the sources best match some target property, such as independence. The second order blind identification (SOBI) BSS method has a number of properties particularly well-suited for data on the cerebral cortex and relies on the calculation of lagged covariance matrices. However while these lagged covariance matrices are readily available in one-dimensional data, they are not straightforward to calculate on the two-dimensional cortical manifold on which certain types of neuroimaging data lie. To address this, I propose a method for calculating the covariance matrices on the cortical manifold and demonstrate its application to cortical gray matter thickness and curvature data on the cerebral cortex.
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Zhao, Wentao. "Genomic applications of statistical signal processing." [College Station, Tex. : Texas A&M University, 2008. http://hdl.handle.net/1969.1/ETD-TAMU-2952.
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Lahlou, Tarek A. (Tarek Aziz). "Decentralized signal processing systems with conservation principles." Thesis, Massachusetts Institute of Technology, 2016. http://hdl.handle.net/1721.1/105946.
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Thesis: Ph. D., Massachusetts Institute of Technology, Department of Electrical Engineering and Computer Science, 2016.This electronic version was submitted by the student author. The certified thesis is available in the Institute Archives and Special Collections.
Cataloged from student-submitted PDF version of thesis.
Includes bibliographical references (pages 271-277).
In this thesis, a framework for designing fixed-point and optimization algorithms realized as asynchronous, distributed signal processing systems is developed with an emphasis on the system's stability, robustness, and variational properties. These systems are formed by connecting basic modules together via interconnecting networks. Several classes of systems are constructed using interconnecting networks that obey certain conservation principles where these principles specifically allow steady-state system variables to be interpreted as solutions to optimization problems in a generally non-convex class and provide local conditions on the individual modules to ensure that the variables tend to such solutions, including when the communication between modules is asynchronous and uncoordinated. A particular class of signal processing systems, referred to as scattering systems, is designed that can solve convex and non-convex optimization problems, and where convex problems do not require problem-specific tuning parameters. Connections between scattering systems and their gradient-based and proximal counterparts are also established. The primary contributions of this thesis broadly serve to assist with designing and implementing scattering systems, both by leveraging existing signal processing paradigms and by developing new results in signal processing theory. To demonstrate the utility of the framework, scattering algorithms implemented as web-services and decentralized processor networks are presented and used to solve problems related to optimum filter design, sparse signal recovery, supervised learning, and non-convex regression.
by Tarek Aziz Lahlou.
Ph. D.
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Demirtas, Sefa. "Functional composition and decomposition for signal processing." Thesis, Massachusetts Institute of Technology, 2014. http://hdl.handle.net/1721.1/89989.
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Thesis: Ph. D., Massachusetts Institute of Technology, Department of Electrical Engineering and Computer Science, 2014.Cataloged from PDF version of thesis.
Includes bibliographical references (pages 175-180).
Functional composition, the application of one function to the results of another function, has a long history in the mathematics community, particularly in the context of polynomials and rational functions. This thesis articulates and explores a general framework for the use of functional composition in the context of signal processing. Its many potential applications to signal processing include utilization of the composition of simpler or lower order subfunctions to exactly or approximately represent a given function or data sequence. Although functional composition currently appears implicitly in a number of established signal processing algorithms, it is shown how the more general context developed and exploited in this thesis leads to significantly improved results for several important classes of functions that are ubiquitous in signal processing such as polynomials, frequency responses and discrete multivariate functions. Specifically, the functional composition framework is exploited in analyzing, designing and extending modular filters, separating marginalization computations into more manageable subcomputations and representing discrete sequences with fewer degrees of freedom than their length and region of support with implications for sparsity and efficiency.
by Sefa Demirtas.
Ph. D.
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Welborn, Matthew Lee 1966. "Flexible signal processing algorithms for wireless communications." Thesis, Massachusetts Institute of Technology, 2000. http://hdl.handle.net/1721.1/86556.
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Thesis (Ph.D.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 2000.Includes bibliographical references (p. 129-132).
by Matthew Lee Welborn.
Ph.D.
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Agjee, F. M. D. E. (Faatima Moosa Dawood Ebrahim). "A wideband signal conditioning system for high voltage measurements using a transconductance topology." Thesis, Stellenbosch : Stellenbosch University, 2000. http://hdl.handle.net/10019.1/51633.
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Thesis (MScEng)--Stellenbosch University, 2000.ENGLISH ABSTRACT: Recent research has shown that standard substation capacitive voltage transformers (CVTs) and current transformers (CTs) can be used for the measurement of wideband high voltage phenomena by employing these apparatus in a transconductance topology. With this topology the voltage waveform can be obtained by integration of the ground return current in the earth straps of the CVT and the CT. This technique does, however, impose unique requirements on the amplification and integration instrumentation due to large dynamic range requirements and the strict offset specifications required for accurate integration. This thesis describes a programmable, wideband signal conditioning system for high voltage (HV) measurements using the transconductance topology. A suitable system topology, optimised to reduce the problems usually associated with grounding and electromagnetic interference (EMI) in HV environments, is proposed. This system consists of an analog signal conditioning subsystem, a digital signal conditioning subsystem and a high speed serial fibre-optic link. The analog signal conditioning subsystem conditions the signals from a sensor to levels suitable for the digitiser of the digital signal conditioning subsystem. The high bandwidth specification of the application made it necessary to consider both discrete and integrated implementation of the analog signal conditioning subsystem. Based on the simulated and laboratory test results of both implementations, the optimum design was chosen for the developed system. The digital signal conditioning subsystem, which performs the integration, as well as the serial optic-fibre link control logic was implemented using programmable logic array (PLA) technology. The digital data is transmitted across the fibre-optic link. This data is then converted back to an analog signal. Keywords: High voltage measurements, Transconductance topology.
AFRIKAANSE OPSOMMING: Onlangse navorsing het aangetoon dat standaard substasie kapasitiewe spanningstransformators en stroomtransformators gebruik kan word om wyeband hoogspanningsverskynsels te meet deur hierdie apparatuur in 'n transkonduktansie topologie aan te wend. Met hierdie topologie kan die spanningsgolfvorm verkry word deur die integrasie van die aardstrome in die aardverbindings van die kapasitiewe spanningstransformators en stroomtransformators. Hierdie tegniek stel egter unieke vereistes vir die versterkings- en integrasieinstrumentasie te wyte aan groot dinamiese bereik vereistes en die streng afset spesifikasies wat benodig word vir akkurate integrasie. Hierdie tesis beskryf 'n programmeerbare, wyeband seinkondisioneringstelsel vir hoogspanningsmetings deur van die transkonduktansie topologie gebruik te maak. 'n Geskikte stelseltopologie, wat geoptimiseer is om probleme, wat gewoonlik met aarding en elektromagnetiese interferensie in hoogspanningsomgewings geassosieer word, te verminder, is voorgestel. Die stelsel bestaan uit 'n analoog seinkondisioneringsubstelsel, 'n digitale seinkondisioneringsubstelsel en 'n hoëspoed seriële optiese vesel koppelvlak. Die analoog seinkondisioneringsubstelsel kondisioneer die seine vanaf 'n sensor na geskikte vlakke vir die versyferaar van die digitale seinkondisioneringsubstelsel. Die hoë bandwydte spesifikasie van die toepassing vereis die inagneming van beide diskrete en geïntegreerde implementerings van die analoog seinkondisioneringsubstelsel. Gebaseer op gesimuleerde en laboratorium toetsresultate van beide implementerings is die optimale ontwerp vir die ontwikkelde stelsel gekies. Die digitale seinkondisioneringsubstelsel wat die integrasie uitvoer, asook die seriële optiese vesel koppelvlak beheerlogika is geïmplementeer met behulp van programmeerbare logika skikking tegnologie. Die digitale data word gestuur oor die optiese vesel koppelvlak. Hierdie data word dan terug geskakel na 'n analoog sein. Sleutelwoorde: Hoogspanningsmetings, Transkonduktansie topologie.
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Hussein, Ghazanfar. "Optical signal processing using photorefractive crystals." Thesis, University of Southampton, 1992. https://eprints.soton.ac.uk/396383/.
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I describe in this thesis various techniques of optical signal processing using photorefractive BSO and BaTiO3 crystals. Operations of contrast manipulation, motion detection and parallel optical logic operations are demonstrated. Dynamic instabilities have also been investigated in photorefractive BaTiO3, in the mutually pumped geometry. Contrast manipulation of optical images has been performed via degenerate four wave mixing in BSO and BaTiO3 crystals. In BSO the technique adopted has the apparent drawback of intensity reduction, due to low reflectivities achieved, while using BaTiO3, selective enhancement is achieved for specific Fourier components. An improved versatile technique of polarization encoding of the object Fourier transform has also been implemented with, and without the inclusion of photorefractive crystals. Applications of this technique for phase contrast imaging, and observation in the field of aerodynamics, and Fourier transform synthesis, has been proposed and demonstrated. Optical motion detection using the differential response time of multiplexed gratings in photorefractive BSO has been demonstrated. The operation of velocity filtering has also been demonstrated using complementary gratings in a BSO crystal, in which specific features are only detected at particular speeds. All sixteen basic parallel optical logic operations have been demonstrated using polarization encoding in a phase conjugate Michelson interferometer with a crystal of BSO as a phase conjugate mirror. Finally dynamic instabilities in BaTiO3 in the 'Bird-wing' mutually-pumped configuration have also been investigated, and a phenomenological model is developed. Additionally various improvements and refinements have been proposed which will make these techniques more flexible and versatile.
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Sellergren, Albin, Tobias Andersson, and Jonathan Toft. "Signal processing through electroencephalography : Independent project in electrical engineering." Thesis, Uppsala universitet, Elektricitetslära, 2016. http://urn.kb.se/resolve?urn=urn:nbn:se:uu:diva-298771.
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This report is about a project where electroencephalography (EEG) wasused to control a two player game. The signals from the EEG-electrodeswere amplified, filtered and processed. Then the signals from the playerswere compared and an algorithm decided what would happen in the gamedepending on which signal was largest. The controls and the gaming mechanismworked as intended, however it was not possible to gather a signal fromthe brain with the method used in this project. So ultimately the goal wasnot reached.electroencephalography, EEG
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King, Graham A. "High performance computing systems for signal processing." Thesis, Southampton Solent University, 1996. http://ssudl.solent.ac.uk/2424/.
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The submission begins by demonstrating that the conditions required for consideration under the University's research degrees regulations have been met in full. There then follows a commentary which starts by explaining the origin of the research theme concerned and which continues by discussing the nature and significance of the work. This has been an extensive programme to devise new methods of improving the computational speed and efficiency required for effective implementation of FIR and IIR digital filters and transforms. The problems are analysed and initial experimental work is described which sought to quantify the performance to be derived from peripheral vector processors. For some classes of computation, especially in real time, it was necessary to tum to pure systolic array hardware engines and a large number of innovations are suggested, both in array architecture and in the creation of a new hybrid opto-electronic adder capable of improving the performance ofprocessing elements for the array. This significant and original research is extended further by including a class of computation involving a bit sliced co-processor. A means of measuring the performance of this system is developed and discussed. The contribution of the work has been evident in: software innovation for horizontal architecture microprocessors; improved multi-dimensional systolic array designs; the development of completely new implementations of processing elements in such arrays; and in the details of co-processing architectures for bit sliced microprocessors. The use of Read Only Memory in creating n-dimensional FIR or IIR filters, and in executing the discrete cosine transform is a further innovative contribution that has enabled researchers to re-examine the case for pre-calculated systems previously using stored squares. The Read Only Memory work has suggested that Read Only Memory chips may be combined in a way architecturally similar to systolic array processing elements. This led to original concepts of pipelining for memory devices. The work is entirely coherent in that it covers the application of these contributions to a set of common processes, producing a set of performance graded and scaleable solutions. In order that effective solutions are proposed it was necessary to demonstrate a solid underlying appreciation of the computational mechanics involved. Whilst the published papers within this submission assume such an understanding , two appendices are provided to demonstrate the essential groundwork necessary to underpin the work resulting in these publications. The improved results obtained from the programme were threefold: execution time; theoretical clocking speeds and circuit areas; and speed up ratios. In the case of the investigations involving vector signal processors the issue was one of quantifying the performance bounds of the architecture in performing specific combinations of signal processing functions. An important aspect of this work was the optimisation achieved in the programming of the device. The use of innovative techniques reduced the execution time for the complex combinational algorithms involved to sub 10 milliseconds. Given the real time constraints for typical applications and the aims for this research the work evolved toward dedicated hardware solutions. Systolic arrays were thus a significant area of investigation. In such systems meritorious criteria are concerned with achieving: a higher regularity in architectural structure; data exchanges only with nearest neighbour processing elements; minimised global distribution functions such as power supplies and clock lines; minimised latency; minimisation in the use of latches; the elimination of output adders; and the design of higher speed processing elements. The programme has made original and significant contributions to the art of effective array design culminating in systems calculated to clock at 100MHz when using 1 micron CMOS technology, whilst creating reductions in transistor count when compared with contemporary implementations. The improvements vary by specific design but are ofthe order of30-l00% speed advantage and 20-30% less real estate usage. The third type of result was obtained when considering operations best executed by dedicated microcode running on bit sliced engines. The main issues for this part of the work were the development of effective interactions between host processors and the bit sliced processors used for computationally intensive and repetitive functions together with the evaluation of the relative performance of new bit sliced microcode solutions. The speed up obtained relative to a range of state of the art microprocessors (68040, 80386, 32032) ranged from 2: 1 to 8: 1. The programme of research is represented by sixteen papers divided into three groups corresponding to the following stages in the work: problem definition and initial responses involving vector processors; the synthesis of higher performance solutions using dedicated hardware; and bit sliced solutions
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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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Agaskar, Ameya. "Problems in Signal Processing and Inference on Graphs." Thesis, Harvard University, 2015. http://nrs.harvard.edu/urn-3:HUL.InstRepos:17464767.
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Modern datasets are often massive due to the sharp decrease in the cost of collecting and storing data. Many are endowed with relational structure modeled by a graph, an object comprising a set of points and a set of pairwise connections between them. A ``signal on a graph'' has elements related to each other through a graph---it could model, for example, measurements from a sensor network. In this dissertation we study several problems in signal processing and inference on graphs.
We begin by introducing an analogue to Heisenberg's time-frequency uncertainty principle for signals on graphs. We use spectral graph theory and the standard extension of Fourier analysis to graphs. Our spectral graph uncertainty principle makes precise the notion that a highly localized signal on a graph must have a broad spectrum, and vice versa.
Next, we consider the problem of detecting a random walk on a graph from noisy observations. We characterize the performance of the optimal detector through the (type-II) error exponent, borrowing techniques from statistical physics to develop a lower bound exhibiting a phase transition. Strong performance is only guaranteed when the signal to noise ratio exceeds twice the random walk's entropy rate. Monte Carlo simulations show that the lower bound is quite close to the true exponent.
Next, we introduce a technique for inferring the source of an epidemic from observations at a few nodes. We develop a Monte Carlo technique to simulate the infection process, and use statistics computed from these simulations to approximate the likelihood, which we then maximize to locate the source.
We further introduce a logistic autoregressive model (ALARM), a simple model for binary processes on graphs that can still capture a variety of behavior. We demonstrate its simplicity by showing how to easily infer the underlying graph structure from measurements; a technique versatile enough that it can work under model mismatch.
Finally, we introduce the exact formula for the error of the randomized Kaczmarz algorithm, a linear system solver for sparse systems, which often arise in graph theory. This is important because, as we show, existing performance bounds are quite loose.Engineering and Applied Sciences - Engineering Sciences
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Chou, Kenneth C. (Kenneth Chien-ko). "A stochastic modelling approach to multiscale signal processing." Thesis, Massachusetts Institute of Technology, 1991. http://hdl.handle.net/1721.1/13851.
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Thesis (Ph. D.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 1991.Includes bibliographical references (leaves 260-265).
by Kenneth Chien-ko Chou.
Ph.D.
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Sheet, Lenny. "Noise measurement to 40PPM using digital signal processing." Thesis, Massachusetts Institute of Technology, 1990. http://hdl.handle.net/1721.1/26832.
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Singer, Amy M. (Amy Michelle). "Top-down design of digital signal processing systems." Thesis, Massachusetts Institute of Technology, 1996. http://hdl.handle.net/1721.1/40000.
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Thesis (M. Eng.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 1996.Includes bibliographical references (leaves 45-46).
by Amy M. Singer.
M.Eng.
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Bland, Ross E. (Ross Edward). "Acoustic and seismic signal processing for footsetp detection." Thesis, Massachusetts Institute of Technology, 2006. http://hdl.handle.net/1721.1/37052.
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Thesis (M. Eng.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 2006.Includes bibliographical references (leaves 83-84).
The problem of detecting footsteps using acoustic and seismic sensors is approached from three different angles in this thesis. First, accelerometer data processing systems are designed to make footsteps more apparent to a human operator listening to accelerometer recordings. These systems work by modulating footstep signal energy into the ear's most sensitive frequency bands. Second, linear predictive modeling is shown to be an effective means to detect footsteps in accelerometer and microphone data. The time evolution of the third order linear prediction coefficients leads to the classical binary hypothesis testing framework. Lastly, a new method for blindly estimating the filters of a SIMO channel is presented. This method is attractive because it allows for a more tractable performance analysis.
by Ross E. Bland.
M.Eng.
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Kem, Katherine (Katherine M. ). "Laboratory assignments for teaching introductory signal processing concepts." Thesis, Massachusetts Institute of Technology, 2017. http://hdl.handle.net/1721.1/119529.
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Thesis: M. Eng., Massachusetts Institute of Technology, Department of Electrical Engineering and Computer Science, 2017.This electronic version was submitted by the student author. The certified thesis is available in the Institute Archives and Special Collections.
Cataloged from student-submitted PDF version of thesis.
Includes bibliographical references (page 63).
This thesis proposes labs for a new, applications-based signal processing class. These labs span topics in audio, image, and video processing and will combine signal processing techniques with computational tools. The goal of these labs is to improve student understanding of signal processing concepts and show them the power of signal processing in everyday applications.
by Katherine Kem.
M. Eng.
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Mishra, Ekavali. "Signal processing approaches to analyzing patient cardiovascular state." Thesis, Massachusetts Institute of Technology, 2011. http://hdl.handle.net/1721.1/66448.
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Thesis (M. Eng.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 2011.Cataloged from PDF version of thesis.
Includes bibliographical references (p. 51-52).
There is a wealth of unanalyzed data stored in patient records that could yield insight into a patient's cardiovascular state during surgery and causes of fluctuations in hemodynamics. Recent work suggests that time spent outside a certain blood pressure range corresponds to an increased risk of adverse outcomes after surgery. An analysis of blood pressures recorded during surgery could also be tied to patient fluid responsiveness, pulse pressure variability (PPV) can be a predictor of fluid responsiveness in surgical patients. Thus, a comparison of physiological variables such as cardiac output (CO), total peripheral resistance (TPR), and PPV of patients who experience adverse outcomes to those who do not could help explain the link between adverse outcomes and intraoperative blood pressure variations. Data from patients undergoing cardiothoracic surgery was used to investigate intraoperative hemodynamics. Patients were separated into two groups: those who experienced adverse outcomes within 30 days of surgery (cases) and those who did not (controls). A comparison of blood pressure values extracted from patient data revealed that cases had higher systolic and lower diastolic values during surgery. CO and TPR were computed from these data but a comparison of variability for the two groups yielded no conclusive results.
by Ekavali Mishra.
M.Eng.
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Eghbali, Amir. "Contributions to Flexible Multirate Digital Signal Processing Structures." Licentiate thesis, Linköping : Department of Electrical Engineering, Linköping University, 2009. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-17182.
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Awad, Hazem. "Competitive optics circuits for all-optical signal-processing applications." Thesis, University of Ottawa (Canada), 2006. http://hdl.handle.net/10393/27221.
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This thesis details the simulation and experimental validation of several all-optical signal-processing configurations that utilise competitive optics principles. Competitive optics refers to any number of optical systems where different optical modes or wavelengths compete for a limited system resource, such as the optical gain of some common gain medium, in order to receive amplification. The gain medium can utilise different materials including photorefractive materials (e.g. BaTO3 crystals) and saturated gain material (e.g. semiconductor optical amplifiers). Competitive optics configurations are capable of sophisticated all-optical signal processing functions ranging from all-optical wavelength conversion to optical logic and storage.
This thesis will present a series of simulated competitive optics configurations that utilise a semiconductor ring laser a basic competitive optics structure. These simulations will prove the viability and validity of competitive optics configurations that utilise saturated gain material, specifically the Semiconductor Optical Amplifier. The thesis will demonstrate the application of the Lotka-Volterra mathematical model of competitive interactions to the modelling of some of aforementioned configurations. Finally, experimental investigations of different semiconductor ring lasers configurations are presented and analysed from a competitive optics point of view.
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Taillefer, Christopher. "Analog-to-digital conversion via time-mode signal processing." Thesis, McGill University, 2008. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=18669.
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Conventional voltage-mode analog-to-digital converters use voltage amplifiers, voltage comparators, and switch capacitor networks to perform their signal processing. When compared to digital circuitry, these analog circuit blocks consume significant power, occupy large silicon areas, and operate at relatively slow data processing speeds. A signal processing methodology is proposed that performs analog-to-digital conversion on voltage signals while implementing all the circuits in a digital CMOS logic style. This methodology, called time-mode signal processing, uses time-difference variables as an intermediate signal between the input voltage and digital output. The resulting silicon devices offer very compact, low power, high-speed, and robust analog-to-digital converter alternatives. There are five main analog-to-digital converter topologies: flash, successiveapproximation, pipeline, delta-sigma, and integrating converters. Each converter topology is presented in the context of the time-mode signal processing methodology. The circuits that implement each time-mode data converter are described and when appropriate system-level, transistor-level, and experimental results are revealed. Three integrated circuits (IC) were fabricated in a 0.18-µm CMOS technology to demonstrate the feasibility of the time-mode ADC methodology. The first IC implemented the time-mode comparator and a time-mode flash ADC. The timemode delta-sigma ADC design was demonstrated in the second IC. Two circuits were implemented in the third IC: a differential-input time-mode delta-sigma ADC and a cyclic (or algorithmic) ADC.Les convertisseurs conventionnels pour changer la tension analogique à une tension numérique emploient les amplificateurs de tension, les comparateurs de tension, et les résaux de condensateur sélectionable pour acquir leur traitement de signal. En comparaison le circuit des modules analogues vis-à-vis le circuit numérique nous constatons une augmentation de puissance, une superficie de silicium moins compacte, et un traitement de données beaucoup plus lent. Une méthodologie est proposée pour le traitement du signal qui établi la conversion analogue à numérique sur les signaux de tension et tout en mettant en oeuvre tous les circuits dans un format numérique de type circuit à semiconducteur oxyde-métal à symétrie complémentaire (CMOS). Cette méthodologie reconnue sur le nom de technique-temporelle donne un traitement de signal par domaine temporel en employant la variance de cadence entre les temps comme un signal intermédiare entre la tension d'entrée et la tension de sortie numérique. Les formats numériques de type circuit semiconducteur nous offrent une alternative en temps convertisseur d'analogue à numérique avec l'avantage d'une unité compact, robuste, un coût de puissance réduit, et une haute-vitesse efficace. Il existe cinq topologies principales dans les convertisseurs analogiques à numérique: flash, approximations successives, pipeline, delta-sigma, convertisseurs intégrés. Dans chacune des topologies mentionnées ci-dessus, le traitement de signal par technique-temporelle est une méthode réconnue. Les circuits employés par chaque convertisseur de donnée par technique temporelle sont décrits lorsque le niveau du système est approprié, le niveau du transitor, et les données expérimentales sont identifiés. Trois circuits intégrés (CI) ont été conçus et fabriqués, avec une technologie de 0,18-µm CMOS pour démontrer la possibilité de la méthodologie du techniquetemporelle convertisseur analogique-numéri
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Noor, Fazal. "Inverse and Eigenspace decomposition algorithms for statistical signal processing." Thesis, McGill University, 1993. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=39489.
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In this work, a number of advances are described which we feel lead to better understanding and solution of the eigenvalue and inverse eigenvalue problems for Hermitian Toeplitz matrices. First, a unitary matrix is derived which transforms a Hermitian Toeplitz matrix into a real Toeplitz plus Hankel matrix. Some properties of this transformation are also presented. Second, we solve the inverse eigenvalue problem for Hermitian Toeplitz matrices. Specifically, we present a method for the construction of a Hermitian Toeplitz matrix from an arbitrary set of real eigenvalues. The procedure utilizes the discrete Fourier transform to first construct a real symmetric negacyclic matrix from the specified eigenvalues. The algorithm presented is computationally efficient. Finally, we derive a new order recursive algorithm and modify Trench's algorithm, both for eigenvalue decomposition. The former development is of mathematical interest; whereas, the latter is clearly of practical interest. The modifications proposed to Trench's algorithm are to employ noncontiguous intervals and to include a procedure to detect multiple eigenvalues. The goals of the modification are to improve the rate of convergence. The modified algorithm presented utilizes three root searching techniques: the Pegasus method, the modified Rayleigh quotient iteration with bisection shifts (MRQI-B), and the MRQI with Pegasus shifts (MRQI-P). Simulation results are provided for large matrices of orders 50, 100, 200, and 500. Application of the algorithms to Pisarenko's harmonic decomposition, an important signal processing problem, is presented. Fortran programs of the modified method are also provided.
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Guttman, Michael. "Sampled-data IIR filtering via time-mode signal processing." Thesis, McGill University, 2010. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=86770.
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In this work, the design of sampled-data infinite impulse response filters based on time-mode signal processing circuits is presented. Time-mode signal processing (TMSP), defined as the processing of sampled analog information using time-difference variables, has become one of the more popular emerging technologies in circuit design. As TMSP is still relatively new, there is still much development needed to extend the technology into a general signal-processing tool. In this work, a set of general building block will be introduced that perform the most basic mathematical operations in the time-mode. By arranging these basic structures, higher-order time-mode systems, specifically, time-mode filters, will be realized. Three second-order time-mode filters (low-pass, band-reject, high-pass) are modeled using MATLAB, and simulated in Spectre to verify the design methodology. Finally, a damped integrator and a second-order low-pass time-mode IIR filter are both implemented using discrete components.Dans ce mémoire, la conception de filtres de données-échantillonnées ayant une réponse impulsionnelle infinie basée sur le traitement de signal en mode temporel est présentée. Le traitement de signal dans le domaine temporel (TSDT), définie comme étant le traitement d'information analogique échantillonnée en utilisant des différences de temps comme variables, est devenu une des techniques émergentes de conception de circuits des plus populaires. Puisque le TSDT est toujours relativement récent, il y a encore beaucoup de développements requis pour étendre cette technologie comme un outil de traitement de signal général. Dans cette recherche, un ensemble de blocs d'assemblage capable de réaliser la plupart des opérations mathématiques dans le domaine temporel sera introduit. En arrangeant ces structures élémentaires, des systèmes en mode temporel d'ordre élevé, plus spécifiquement des filtres en mode temporel, seront réalisés. Trois filtres de deuxième ordre dans le domaine temporel (passe-bas, passe-bande et passe-haut) sont modélisés sur MATLAB et simulé sur Spectre afin de vérifier la méthodologie de conception. Finalement, un intégrateur amorti et un filtre passe-bas IIR de deuxième ordre en mode temporel sont implémentés avec des composantes discrètes.
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Tran, Merry Thi. "Applications of Digital Signal Processing with Cardiac Pacemakers." PDXScholar, 1992. https://pdxscholar.library.pdx.edu/open_access_etds/4582.
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Because the voltage amplitude of a heart beat is small compared to the amplitude of exponential noise, pacemakers have difficulty registering the responding heart beat immediately after a pacing pulse. This thesis investigates use of digital filters, an inverse filter and a lowpass filter, to eliminate the effects of exponential noise following a pace pulse. The goal was to create a filter which makes recognition of a haversine wave less dependent on natural subsidence of exponential noise. Research included the design of heart system, pacemaker, pulse generation, and D sensor system simulations. The simulation model includes the following components: \ • Signal source, A MA TLAB generated combination of a haversine signal, exponential noise, and myopotential noise. The haversine signal is a test signal used to simulate the QRS complex which is normally recorded on an ECG trace as a representa tion of heart function. The amplitude is approximately 10 mV. Simulated myopotential noise represents a uniformly distributed random noise which is generated by skeletal muscle tissue. The myopotential noise has a frequency spectrum extending from 70 to 1000Hz. The amplitude varies from 2 to 5 mV. Simulated exponential noise represents the depolarization effects of a pacing pulse as seen at the active cardiac lead. The amplitude is about -1 volt, large in comparison with the haversine signal. • AID converter, A combination of sample & hold and quantizer functions translate the analog signal into a digital signal. Additionally, random noise is created during quantization. • Digital filters, An inverse filter removes the exponential noise, and a lowpass filter removes myopotential noise. • Threshold level detector, A function which detects the strength and amplitude of the output signal was created for robustness and as a data sampling device. The simulation program is written for operation in a DOS environment. The program generates a haversine signal, myopotential noise (random noise), and exponential noise. The signals are amplified and sent to an AID converter stage. The resultant digital signal is sent to a series of digital filters, where exponential noise is removed by an inverse digital filter, and myopotential noise is removed by the Chebyshev type I lowpass digital filter. The output signal is "detected" if its waveform exceeds the noise threshold level. To determine what kind of digital filter would remove exponential noise, the spectrum of exponential noise relative to a haversine signal was examined. The spectrum of the exponential noise is continuous because the pace pulse is considered a non-periodic signal (assuming the haversine signal occurs immediately after a pace pulse). The spectrum of the haversine is also continuous, existing at every value of frequency co. The spectrum of the haversine is overlapped by the spectrum of and amplitude of the exponential, which is several orders of magnitude larger. The exponential cannot be removed by conventional filters. Therefore, an inverse filter approach is used to remove exponential noise. The transfer function of the inverse filter of the model has only zeros. This type of filter is called FIR, all-zero, non recursive, or moving average. Tests were run using the model to investigate the behavior of the inverse filter. It was found that the haversine signal could be clearly detected within a 5% change in the time constant of the exponential noise. Between 5% and 15% of change in the time constant, the filtered exponential amplitude swamps the haversine signal. The sensitivity of the inverse filter was also studied: when using a fixed exponential time constant but changing the location of the transfer function, the effect of the exponential noise on the haversine is minimal when zeros are located between 0.75 and 0.85 of the unit circle. After the source signal passes the inverse filter, the signal consists only of the haversine signal, myopotential noise, and some random noise introduced during quantization. To remove these noises, a Chebyshev type I lowpass filter is used.
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Etumi, Adel. "Current signal processing-based techniques for transformer protection." Thesis, Cardiff University, 2016. http://orca.cf.ac.uk/94716/.
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Transformer is an expensive device and one of the most important parts in a power system. Internal faults can cause a transformer to fail and thus, it is necessary for it to be protected from these faults. Protection doesn’t mean that it prevents damage to the protected transformer but it is to minimize the damage to the transformer as much as possible, which consequently minimizes the subsequent outage time and repair cost. Therefore, fast and reliable protection system should be used for limiting damages to the transformer by rapidly disconnecting the faulty transformer from the network, which also leads to the elimination of the stresses on the system itself and preventing damage to adjacent equipment. The main aim of this thesis is to propose transformer protection technique that is fast and highly sensitive to internal faults that occur inside the transformer, to overcome the problems of current transformer saturation and inrush current, and to make it immune to the external faults (through faults) that occur outside of the transformer protection zone. The current transformer saturation and inrush current are significant problems since they cause malfunction of the protection system, which consequently will disconnect the transformer because they are considered faults. This improper disconnection of transformer is not desirable as it shortens its life time. So the proposed protection technique was designed to be fast and to avoid maloperation caused by saturation and inrush current. The proposed protection technique was based on current signal processing. Three methods, namely the application of correlation coefficients, current change ratio (CCR) and percentage area difference (PAD) were proposed based on practical and simulation tests. These techniques were successfully proved by carrying out tests on Simulink models using MATLAB/SIMULINK program and on a practical laboratory model. In transformer transient state, the response time for the methods that were used to address the problem of inrush condition, was 10 ms for CCR when transformer was on no-load and 5 ms for PAD when the transformer was on-load. This response time Current Signal Processing-Based Techniques for Transformer Protection v is faster than the most popular method relying on second harmonic, which needs at least one cycle (20ms in 50 Hz systems) to recognize the condition. In transformer steady state, it was proved that the proposed correlation method was capable of detecting the internal faults successfully within a very short time, ranging from 0.8 to 2.5 ms according to the type and severity of the fault and in addition was able to overcome the problem of current transformer (CT) saturation. The contribution of this research is the development of a transformer protection technique, which is simple in design, fast and reliable in fault detection and at the same time capable of overcoming the problems of current transformer saturation and inrush current.
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Lipstreu, William F. "Digital Signal Processing Laboratory Using Real-Time Implementations of Audio Applications." Cleveland, Ohio : Case Western Reserve University, 2009. http://rave.ohiolink.edu/etdc/view?acc_num=case1240836810.
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Said, Maya Rida 1976. "Signal processing in biological cells : proteins, networks, and models." Thesis, Massachusetts Institute of Technology, 2005. http://hdl.handle.net/1721.1/30165.
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Thesis (Sc. D.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 2005.Includes bibliographical references (p. 202-210).
This thesis introduces systematic engineering principles to model, at different levels of abstraction the information processing in biological cells in order to understand the algorithms implemented by the signaling pathways that perform the processing. An example of how to emulate one of these algorithms in other signal processing contexts is also presented. At a high modeling level, the focus is on the network topology rather than the dynamical properties of the components of the signaling network. In this regime, we examine and analyze the distribution and properties of the network graph. Specifically, we present a global network investigation of the genotype/phenotype data-set recently developed for the yeast Saccharomyces cerevisiae from exposure to DNA damaging agents, enabling explicit study of how protein-protein interaction network characteristics may be associated with phenotypic functional effects. The properties of several functional yeast networks are also compared and a simple method to combine gene expression data with network information is proposed to better predict pathophysiological behavior. At a low level of modeling, the thesis introduces a new framework for modeling cellular signal processing based on interacting Markov chains. This framework provides a unified way to simultaneously capture the stochasticity of signaling networks in individual cells while computing a deterministic solution which provides average behavior. The use of this framework is demonstrated on two classical signaling networks: the mitogen activated protein kinase cascade and the bacterial chemotaxis pathway. The prospects of using cell biology as a metaphor for signal processing are also considered in a preliminary way by presenting a surface mapping algorithm based on bacterial chemotaxis.
by Maya Rida Said.
Sc.D.
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Tokmouline, Timur. "A signal oriented stream processing system for pipeline monitoring." Thesis, Massachusetts Institute of Technology, 2006. http://hdl.handle.net/1721.1/37106.
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Thesis (M. Eng.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 2006.Includes bibliographical references (p. 115-117).
In this thesis, we develop SignalDB, a framework for composing signal processing applications from primitive stream and signal processing operators. SignalDB allows the user to focus on the signal processing task and avoid needlessly spending time on learning a particular application programming interface (API). We use SignalDB to express acoustic and pressure transient methods for water pipeline monitoring as query plans consisting of signal processing operators.
by Timur Tokmouline.
M.Eng.
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Gruhl, Daniel F. "LibDsp, an object oriented C++ digital signal processing library." Thesis, Massachusetts Institute of Technology, 1995. http://hdl.handle.net/1721.1/37539.
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Thesis (M. Eng.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 1995.Includes bibliographical references (leaves 194-195).
by Daniel F. Gruhl.
M.Eng.
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Alam, Shaif-ul. "Advanced fibre circuitry for all-optical signal processing." Thesis, University of Southampton, 2000. https://eprints.soton.ac.uk/15501/.
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This thesis presents results of new types of fibre lasers and oscillators as well as a new all-fibre nonlinear modulator with nearly instantaneous response time. The development of a simple and stable, passively mode-locked source of picosecond pulses is described in chapter 2. Here the mode locking of the laser was obtained by using the combined effect of frequency-shifted feedback and a nonlinear amplifying loop mirror. The new cavity configuration allowed tuning of the laser continuously over 25 nm of the erbium gain bandwidth by using a bulk diffraction grating. The shortest reported pulse width (1.2 ps) from this type of laser has been demonstrated. A complete characterisation of the laser, including its mode locking build-up time, is presented. Cascaded nonlinearity in quadratic nonlinear materials is the basis in realising nonlinearity free optical amplifiers. Experimental results on cascaded nonlinearity in a periodically poled lithium niobate sample are presented in chapter 3. A nonlinear phase shift of more than 1.5p was obtained from only a 4 mm long sample due to its large effective nonlinear refractive index coefficient (~1 x 10-13 cm2/W). Experiments on the nonlinear phase shift compensation in an optical fibre and amplifier are presented in chapter 4. Phase shift compensation of more than 1.5p has been successfully demonstrated. Chapter 5 presents theoretical investigations on the modulation of optical signals using the stimulated Raman scattering (SRS) process in silica fibre. Numerical results reveal that to modulate signals in this scheme it is necessary to consider other competing nonlinear effects such as cascaded SRS, modulation instability etc. actively. Signal modulation as fast as 250 - 300 GHz can be realised using this intensity modulator. Chapter 6 describes the experimental results on the modulation of optical signal using SRS in optical fibre. With this Raman intensity modulator, bit-by-bit modulation of 10 Gbit/s simulated data stream has been demonstrated. An extinction ratio (modulation depth) of more than 15 dB was realised. The proposed intensity modulator can also be used as a time domain scalpel and can create a dark pulse in a bright background.
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Lin, Lu. "Adaptive signal processing in subbands using sigma-delta modulation technique." Thesis, University of Ottawa (Canada), 1994. http://hdl.handle.net/10393/6532.
Full textAbstract:
In this thesis, the use of subbanding and sigma-delta modulation in interference/noise cancellation is intensively studied and a sigma-delta modulated subbanded adaptive interference/noise cancellation system is proposed. The filter bank is fully sigma-delta modulated. The output signal from the filter bank is then used to produce the input to the adaptive filter. The adaptive filter is partially sigma-delta modulated. The output is demodulated at the final stage. Maintaining the sigma-delta modulated signal representation throughout the system results in considerable savings in complexity. The performance of the proposed system is studied and compared to the regular non sigma-delta modulated case regarding complexity, convergence speed and steady state error. The effect of the oversampling rate used in the sigma-delta modulation as well as the quality of the demodulator is also considered. It is shown that in the case of interference cancellation a comb filter is sufficient, while in the case of noise canceller a good quality demodulator is essential. The thesis concludes by highlighting the tradeoffs between the hardware complexity reduction and the overall system performance.
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Balraj, Navaneethakrishnan. "AUTOMATED ACCIDENT DETECTION IN INTERSECTIONS VIA DIGITAL AUDIO SIGNAL PROCESSING." MSSTATE, 2003. http://sun.library.msstate.edu/ETD-db/theses/available/etd-10212003-102715/.
Full textAbstract:
The aim of this thesis is to design a system for automated accident detection in intersections. The input to the system is a three-second audio signal. The system can be operated in two modes: two-class and multi-class. The output of the two-class system is a label of ?crash? or ?non-crash?. In the multi-class system, the output is the label of ?crash? or various non-crash incidents including ?pile drive?, ?brake?, and ?normal-traffic? sounds. The system designed has three main steps in processing the input audio signal. They are: feature extraction, feature optimization and classification. Five different methods of feature extraction are investigated and compared; they are based on the discrete wavelet transform, fast Fourier transform, discrete cosine transform, real cepstrum transform and Mel frequency cepstral transform. Linear discriminant analysis (LDA) is used to optimize the features obtained in the feature extraction stage by linearly combining the features using different weights. Three types of statistical classifiers are investigated and compared: the nearest neighbor, nearest mean, and maximum likelihood methods. Data collected from Jackson, MS and Starkville, MS and the crash signals obtained from Texas Transportation Institute crash test facility are used to train and test the designed system. The results showed that the wavelet based feature extraction method with LDA and maximum likelihood classifier is the optimum design. This wavelet-based system is computationally inexpensive compared to other methods. The system produced classification accuracies of 95% to 100% when the input signal has a signal-to-noise-ratio of at least 0 decibels. These results show that the system is capable of effectively classifying ?crash? or ?non-crash? on a given input audio signal.
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Cosgrave, Joseph Anthony. "Acoustic-optic monitoring of electrical power equipment using chromatic signal processing." Thesis, University of Liverpool, 1996. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.263845.
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Andrikogiannopoulos, Nikolas I. "RF phase modulation of optical signals and optical/electrical signal processing." Thesis, Massachusetts Institute of Technology, 2006. http://hdl.handle.net/1721.1/42930.
Full textAbstract:
Thesis (S.M.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 2006.This electronic version was submitted by the student author. The certified thesis is available in the Institute Archives and Special Collections.
Includes bibliographical references (p. 125-127).
Analog RF phase modulation of optical signals has been a topic of interest for many years, mainly focusing on Intensity Modulation Direct Detection (IMDD). The virtues of coherent detection combined with the advantages of Frequency Modulation, however, have not been explored thoroughly. By employing Frequency Modulation Coherent Detection (FMCD), the wide optical transmission bandwidth of optical fiber can be traded for higher signal-to-noise performance. In this thesis, we derive the FM gain over AM modulation -- the maximum achievable signal-to-noise ratio (by spreading the signal's spectrum) for specific carrier-to-noise ratio. We then employ FMCD for a scheme of remote antennas for which we use optical components and subsystem to perform signal processing such as nulling of interfering signals. The performance of optical processing on different modulation schemes are compared, and some important conclusions are reported relating to the use of conventional FMCD, FMCD with optical discriminator (FMCD O-D), and IMDD. Specifically, the superiority of conventional FMCD is shown; and, on the other hand, the inferiority of FMCD O-D is shown (same performance as IMDD) because of the use of an O-D. Finally, the remote antenna scheme is generalized for N antennas and N users.
by Nikolas I. Andrikogiannopoulos.
S.M.
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Song, William S. "A fault-tolerant multiprocessor architecture for digital signal processing applications." Thesis, Massachusetts Institute of Technology, 1988. http://hdl.handle.net/1721.1/14427.
Full textAbstract:
Thesis (Ph. D.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 1989.Includes bibliographical references.
Partly funded by US Air Force Office of Scientific Research. AFOSR-86-0164 Partly funded by Draper Laboratories.
by William S. Song.
Ph.D.
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Kunz, Eben A. "Low cost analog signal processing for massive radio telescope arrays." Thesis, Massachusetts Institute of Technology, 2012. http://hdl.handle.net/1721.1/77079.
Full textAbstract:
Thesis (M. Eng.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 2012.Cataloged from PDF version of thesis.
Includes bibliographical references (p. 38).
Measurement and analysis of redshifted 21cm hydrogen emissions is a developing technique for studying the early universe. The primary time of interest corresponds to a signal in the the 100-200MHz frequency band. The Omniscope is a new type of radio telescope array being developed at MIT which images the entire sky in this band at low resolution using spatial Fourier transforms. In order to gain the maximum benefit from this type of telescope, a regular array of more than 10,000 antennas will eventually be necessary. I detail a low cost analog signal path which was developed to test and refine the signal processing and imaging pathways of the Omniscope. This signal path begins at the output of a preexisting antenna design and ends with digitization.
by Eben A. Kunz.
M.Eng.
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Olshevsky, Alexander. "Efficient information aggregation strategies for distributed control and signal processing." Thesis, Massachusetts Institute of Technology, 2010. http://hdl.handle.net/1721.1/62427.
Full textAbstract:
Thesis (Ph. D.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 2010.Cataloged from PDF version of thesis.
Includes bibliographical references (p. 129-136).
This thesis will be concerned with distributed control and coordination of networks consisting of multiple, potentially mobile, agents. This is motivated mainly by the emergence of large scale networks characterized by the lack of centralized access to information and time-varying connectivity. Control and optimization algorithms deployed in such networks should be completely distributed, relying only on local observations and information, and robust against unexpected changes in topology such as link failures. We will describe protocols to solve certain control and signal processing problems in this setting. We will demonstrate that a key challenge for such systems is the problem of computing averages in a decentralized way. Namely, we will show that a number of distributed control and signal processing problems can be solved straightforwardly if solutions to the averaging problem are available. The rest of the thesis will be concerned with algorithms for the averaging problem and its generalizations. We will (i) derive the fastest known averaging algorithms in a variety of settings and subject to a variety of communication and storage constraints (ii) prove a lower bound identifying a fundamental barrier for averaging algorithms (iii) propose a new model for distributed function computation which reflects the constraints facing many large-scale networks, and nearly characterize the general class of functions which can be computed in this model.
by Alexander Olshevsky.
Ph.D.