Dissertations / Theses: 'Adaptive acoustics'

1

Dessalermos, Spyridon. "Undersea acoustic propagation channel estimation." Thesis, Monterey, Calif. : Springfield, Va. : Naval Postgraduate School ; Available from National Technical Information Service, 2005. http://library.nps.navy.mil/uhtbin/hyperion/05Jun%5FDessalermos.pdf.

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Thesis (M.S. in Electrical Engineering and M.S. in Applied Physics)--Naval Postgraduate School, June 2005.
Thesis Advisor(s): Joseph Rice, Roberto Cristi. Includes bibliographical references (p. 117-119). Also available online.

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Hermand, Jean-Pierre. "Environmentally-Adaptive Signal Processing in Ocean Acoustics." Doctoral thesis, Universite Libre de Bruxelles, 1993. http://hdl.handle.net/2013/ULB-DIPOT:oai:dipot.ulb.ac.be:2013/212734.

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Creasy, Miles Austin. "Adaptive Collocated Feedback for Noise Absorption in Acoustic Enclosures." Thesis, Virginia Tech, 2006. http://hdl.handle.net/10919/45209.

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This thesis focuses on adaptive feedback control for low frequency acoustic energy absorption in acoustic enclosures. The specific application chosen for this work is the reduction of high interior sound pressure levels (SPL) experienced during launch within launch vehicle payload fairings. Two acoustic enclosures are used in the research: the first being a symmetric cylindrical duct and the other being a full scale model of a payload fairing. The symmetric cylindrical duct is used to validate the ability of the adaptive controller to compensate for large changes in the interior acoustical properties. The payload fairing is used to validate that feedback control, for a large geometry, does absorb acoustic energy. The feedback controller studied in this work is positive position feedback (PPF) used in conjunction with high and low pass Butterworth filters. An algorithm is formed from control experiments for setting the filter parameters of the PPF and Butterworth filters from non-adaptive control simulations and tests of the duct and payload fairing. This non-adaptive control shows internal SPL reductions of 2.2 dB in the cylindrical duct for the frequency range from 100 to 500 Hz and internal SPL reductions of 4.2 dB in the full scale fairing model for the frequency range from 50 to 250 Hz. The experimentally formed control algorithm is then used as the basis for an adaptive controller that uses the collocated feedback signal to actively tune the control parameters. The cylindrical duct enclosure with a movable end cap is used to test the adaptation properties of the controller. The movable end cap allows the frequencies of the acoustic modes to vary by more than 20 percent. Experiments show that a 10 percent change in the frequencies of the acoustic modes cause the closed-loop system to go unstable with a non-adaptive controller. The closed-loop system with the adaptive controller maintains stability and reduces the SPL throughout the 20 percent change of the acoustic modes' frequencies with a 2.3 dB SPL reduction before change and a 1.7 dB SPL reduction after the 20 percent change.
Master of Science

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Srinivas, Vivek. "Adaptive, Wave Guiding Acoustic Arrays using Circularly Symmetric Reconfigurable Structures." The Ohio State University, 2020. http://rave.ohiolink.edu/etdc/view?acc_num=osu1587130205893861.

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5

Fuller, Ryan Michael. "Adaptive Noise Reduction Techniques for Airborne Acoustic Sensors." Wright State University / OhioLINK, 2012. http://rave.ohiolink.edu/etdc/view?acc_num=wright1355361066.

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6

Freij, G. J. "Enhanced sequential adaptive linear prediction for speech encoding." Thesis, University of Liverpool, 1985. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.356268.

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Wong, Lawrence Yik-Lung. "Adaptive system modelling for active attenuation of sound." Thesis, Open University, 1992. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.316967.

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Al-Kindi, Manal Jamil. "Implementation of adaptive noise cancellation in the diving environment." Thesis, University of Strathclyde, 1988. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.304949.

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9

Gillespie, Andrew Fleming Ralph. "The application of adaptive transversal filtering to active noise control." Thesis, London South Bank University, 1992. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.316973.

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10

John, Ranjit Yohannan. "Adaptive filtering and the identification of tones in broadband noise." Thesis, University of Southampton, 1991. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.315347.

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Higley, William J. "Self-adaptive processes for the mitigation of coherent multipath in ocean acoustics." Connect to a 24 p. preview or request complete full text in PDF format. Access restricted to UC campuses, 2007. http://wwwlib.umi.com/cr/ucsd/fullcit?p3259053.

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Thesis (Ph. D.)--University of California, San Diego, 2007.
Title from first page of PDF file (viewed June 21, 2007). Available via ProQuest Digital Dissertations. Vita. Includes bibliographical references.

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Round, David Peter. "Application of DSP methods to sound reproduction." Thesis, Bangor University, 1996. https://research.bangor.ac.uk/portal/en/theses/application-of-dsp-methods-to-sound-reproduction(89e56009-ab09-4054-9a6b-e5181507a3f4).html.

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Courtis, N. J. "Some aspects of speech intelligibility enhancement with particular regard to adaptive filtering and room acoustics." Thesis, University of Hertfordshire, 1985. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.356313.

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Drumm, Ian. "The development and application of an adaptive beam tracing algorithm to predict the acoustics of auditoria." Thesis, University of Salford, 1998. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.244883.

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Fung, Chi Fung. "On-line dynamical system modelling using radial basis function networks in adaptive non-linear noise cancellation." Thesis, University of Sheffield, 1996. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.389790.

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Lewis, Matthew Robert S. M. Massachusetts Institute of Technology. "Evaluation of vector sensors for adaptive equalization in underwater acoustic communication." Thesis, Massachusetts Institute of Technology, 2014. http://hdl.handle.net/1721.1/93793.

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Thesis: S.M., Joint Program in Applied Ocean Science and Engineering (Massachusetts Institute of Technology, Department of Mechanical Engineering; and the Woods Hole Oceanographic Institution), 2014.
Cataloged from PDF version of thesis.
Includes bibliographical references (pages 123-125).
Underwater acoustic communication is an extremely complex field that faces many challenges due to the time-varying nature of the ocean environment. Vector sensors are a proven technology that when utilizing their directional sensing capabilities allows us to minimize the effect of interfering noise sources. A traditional pressure sensor array has been the standard for years but suffers at degraded signal to noise ratios (SNR) and requires maneuvers or a lengthly array aperture to direction find. This thesis explores the effect of utilizing a vector sensor array to steer to the direction of signal arrival and the effect it has on equalization of the signal at degraded SNRs. It was demonstrated that utilizing a single vector sensor we were able steer to the direction of arrival and improve the ability of an equalizer to determine the transmitted signal. This improvement was most prominent when the SNR was degraded to levels of 0 and 10 dB where the performance of the vector sensor outperformed that of the pressure sensor in nearly 100% of cases. Finally, this performance improvement occurred with a savings in computational expense.
by Matthew Robert Lewis.
S.M.

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Yellepeddi, Atulya. "Direct-form adaptive equalization for underwater acoustic communication." Thesis, Massachusetts Institute of Technology, 2012. http://hdl.handle.net/1912/5281.

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Thesis (S.M.)--Joint Program in Applied Ocean Science and Engineering (Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science; and the Woods Hole Oceanographic Institution), 2012.
Cataloged from PDF version of thesis.
Includes bibliographical references (p. 139-143).
Adaptive equalization is an important aspect of communication systems in various environments. It is particularly important in underwater acoustic communication systems, as the channel has a long delay spread and is subject to the effects of time- varying multipath fading and Doppler spreading. The design of the adaptation algorithm has a profound influence on the performance of the system. In this thesis, we explore this aspect of the system. The emphasis of the work presented is on applying concepts from inference and decision theory and information theory to provide an approach to deriving and analyzing adaptation algorithms. Limited work has been done so far on rigorously devising adaptation algorithms to suit a particular situation, and the aim of this thesis is to concretize such efforts and possibly to provide a mathematical basis for expanding it to other applications. We derive an algorithm for the adaptation of the coefficients of an equalizer when the receiver has limited or no information about the transmitted symbols, which we term the Soft-Decision Directed Recursive Least Squares algorithm. We will demonstrate connections between the Expectation-Maximization (EM) algorithm and the Recursive Least Squares algorithm, and show how to derive a computationally efficient, purely recursive algorithm from the optimal EM algorithm. Then, we use our understanding of Markov processes to analyze the performance of the RLS algorithm in hard-decision directed mode, as well as of the Soft-Decision Directed RLS algorithm. We demonstrate scenarios in which the adaptation procedures fail catastrophically, and discuss why this happens. The lessons from the analysis guide us on the choice of models for the adaptation procedure. We then demonstrate how to use the algorithm derived in a practical system for underwater communication using turbo equalization. As the algorithm naturally incorporates soft information into the adaptation process, it becomes easy to fit it into a turbo equalization framework. We thus provide an instance of how to use the information of a turbo equalizer in an adaptation procedure, which has not been very well explored in the past. Experimental data is used to prove the value of the algorithm in a practical context.
by Atulya Yellepeddi.
S.M.

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Offermans, Nicolas. "Towards adaptive mesh refinement in Nek5000." Licentiate thesis, KTH, Mekanik, 2017. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-217501.

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The development of adaptive mesh refinement capabilities in the field of computational fluid dynamics is an essential tool for enabling the simulation of larger and more complex physical problems. While such techniques have been known for a long time, most simulations do not make use of them because of the lack of a robust implementation. In this work, we present recent progresses that have been made to develop adaptive mesh refinement features in Nek5000, a code based on the spectral element method. These developments are driven by the algorithmic challenges posed by future exascale supercomputers. First, we perform the study of the strong scaling of Nek5000 on three petascale machines in order to assess the scalability of the code and identify the current bottlenecks. It is found that strong scaling limit ranges between 5, 000 and 220, 000 degrees of freedom per core depending on the machine and the case. The need for synchronized and low latency communication for efficient computational fluid dynamics simulation is also confirmed. Additionally, we present how Hypre, a library for linear algebra, is used to develop a new and efficient code for performing the setup step required prior to the use of an algebraic multigrid solver for preconditioning the pressure equation in Nek5000. Finally, the main objective of this work is to develop new methods for estimating the error on a numerical solution of the Navier–Stokes equations via the resolution of an adjoint problem. These new estimators are compared to existing ones, which are based on the decay of the spectral coefficients. Then, the estimators are combined with newly implemented capabilities in Nek5000 for automatic grid refinement and adaptive mesh adaptation is carried out. The applications considered so far are steady and two-dimensional, namely the lid-driven cavity at Re = 7, 500 and the flow past a cylinder at Re = 40. The use of adaptive mesh refinement techniques makes mesh generation easier and it is shown that a similar accuracy as with a static mesh can be reached with a significant reduction in the number of degrees of freedom.

QC 20171114

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Shi, Kun. "Nonlinear acoustic echo cancellation." Diss., Atlanta, Ga. : Georgia Institute of Technology, 2008. http://hdl.handle.net/1853/26704.

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Thesis (Ph.D)--Electrical and Computer Engineering, Georgia Institute of Technology, 2009.
Committee Chair: G. Tong Zhou; Committee Co-Chair: Xiaoli Ma; Committee Member: David V. Anderson; Committee Member: James Stevenson Kenney; Committee Member: Liang Peng; Committee Member: William D. Hunt. Part of the SMARTech Electronic Thesis and Dissertation Collection.

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Schlicher, Jeremy T. "A Multi-Genre Adaptive Performance Hall." University of Cincinnati / OhioLINK, 2006. http://rave.ohiolink.edu/etdc/view?acc_num=ucin1148489621.

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21

Касянчик, Юрій Олександрович. "Мультифункціональний зал з адаптивною акустикою." Master's thesis, КПІ ім. Ігоря Сікорського, 2019. https://ela.kpi.ua/handle/123456789/30467.

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Дипломна робота містить основну частину на 75 аркушах, 63 ілюстрацій. Метою роботи є аналіз існуючої літератури, засвоєння теоретичних знань і реалізація їх на проектованому прикладі. А саме система адаптивної акустики в поєднанні з системою Wave Field Synthesis(WFS) в мультифункціональному залі. Головною функцією якого є відтворення різних програм, від мовних і музичних до відтворення звуку за стандартом Dolby. Після підбору всіх вхідних данних, розмірів приміщення і вибору акустичної системи проведено аналіз в програмному пакеті Matlab з використанням змодельованих графіків і масивів данних. По всій роботі, з врахуванням всіх результатів зроблені висновки.
The thesis contains the main part of 75 sheets, 63 illustrations. The purpose of the work is to analyze the existing literature, assimilate theoretical knowledge and implement it on a projected example. Namely, the system of adaptive acoustics in combination with the Wave Field Synthesis (WFS) in the multifunctional room. The main function of which is the playback of various programs, from speech and music to Dolby sound. After selecting all of the input data, the size of the premises and the choice of the speaker system, the analysis was performed in the Matlab software package using simulated graphs and arrays. Throughout the work, all the conclusions are drawn.

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Videla, Marió Javier Andrés. "Spline-based methods with adaptive refinement for problems of acoustics and fracture mechanics of thin plates." Tesis, Universidad de Chile, 2018. http://repositorio.uchile.cl/handle/2250/170004.

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Tesis para optar al grado de Magíster en Ciencias de la Ingeniería, Mención Mecánica
Both the CAD software and FEM software have a significant impact on engineering nowadays. Even though both are powerful tools for design and analysis, the main drawback is that CAD geometries and Finite Element models do not entirely match, which results in the necessity to re-parameterize the geometry many times during the solution cycle in FEM. Isogeometric Analysis (IGA) was proposed to fulfill this gap and create the direct link between the CAD design and FEM analysis. The main idea of IGA is to substitute the shape functions used in FEM by the shape functions used in the CAD software. In particular, one of the main drawbacks of NURBS basis functions, and therefore of IGA, is the lack of local refinement, which makes them computationally highly expensive in applications that demands a non-uniform refinement of the geometry. Polynomial splines over Hierarchical T-meshes (PHT-splines) were introduced by Deng et al. as a type of spline that allows local refinement and adaptability by means of a polynomial basis capable of parameterizing the geometry. In this work, we demonstrate the application of PHT-splines for two type of problems: time-harmonic acoustic problems, modeled by the Helmholtz equation, and fracture mechanics of thin plate problems, modeled by the Kirchhoff-Love theory. Solutions of the Helmholtz equation have two features: global oscillations associated with the wave number and local gradients caused by geometrical irregularities. The results show that after a sufficient number of degrees of freedom is used to approximate global oscillations, adaptive refinement can capture local features of the solution. The residual-based and recovery-based error estimators are compared and the performance of $p$-refinement is investigated. Moreover, an eXtended Geometry Independent Field approximaTion (XGIFT) formulation based on Polynomials Splines Over Hierarchical T-meshes (PHT-splines) for modeling both static and dynamic fracture mechanic problems for plates described by the Kirchhoff-Love theory is presented. Adaptive refinement is employed using a recovery-based error estimator. Results show that adaptive refinement can capture local features of the solution around the crack tip, improving results in both static and dynamic examples. In both cases, the simulations are done in the context of recently introduced Geometry Independent Field approximaTion (GIFT), where PHT-splines are only used to approximate the solution, while the computational domain is parameterized with NURBS. This approach builds on the natural adaptation ability of PHT-splines and avoids the re-parameterization of the NURBS geometry during the solution refinement process.

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Puikkonen, Panu Tapani. "Development of an Adaptive Equalization Algorithm Using Acoustic Energy Density." BYU ScholarsArchive, 2009. https://scholarsarchive.byu.edu/etd/1686.

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Sound pressure equalization of audio signals using digital signal processors has been a subject of ongoing study for many years. The traditional approach is to equalize sound at a point in a listening environment, but because of its specific dependence on the room frequency response between a source and receiver position, this equalization generally causes the spectral response to worsen significantly at other locations in the room. This work presents both a time-invariant and a time-varying implementation of an adaptive acoustic energy density equalization filter for a one-dimensional sound field. Energy density equalization addresses the aforementioned challenge and others that relate to sound equalization. The theory and real-time implementation of time-invariant sound pressure and energy density equalizers designed using the least-squares method are presented, and their performances are compared. An implementation of a time-varying energy density equalizer is also presented. Time-invariant equalization results based on real-time measurements in a plane-wave tube are presented. A sound pressure equalizer results in a nearly flat spectral magnitude at the point of equalization. However, it causes the frequencies corresponding to spatial nulls at that point to be undesirably boosted elsewhere in the sound field, where those nulls do not exist at the same frequencies. An energy density equalization filter identifies and compensates for all resonances and other global spectral effects of the tube and loudspeaker. It does not attempt to equalize the spatially varying frequency nulls caused by local pressure nodes at the point of equalization. An implementation of a time-varying energy density equalizer is also presented. This method uses the filtered-x filter update to adjust the filter coefficients in real-time to adapt to changes in the sound field. Convergence of the filter over time is demonstrated as the closed end of the tube is opened, then closed once again. Thus, the research results demonstrate that an acoustic energy density filter can be used to time-adaptively equalize global spectral anomalies of a loudspeaker and a one-dimensional sound field.

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Kamaldar, Mohammadreza. "DISCRETE-TIME ADAPTIVE CONTROL ALGORITHMS FOR REJECTION OF SINUSOIDAL DISTURBANCES." UKnowledge, 2018. https://uknowledge.uky.edu/me_etds/129.

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We present new adaptive control algorithms that address the problem of rejecting sinusoids with known frequencies that act on an unknown asymptotically stable linear time-invariant system. To achieve asymptotic disturbance rejection, adaptive control algorithms of this dissertation rely on limited or no system model information. These algorithms are developed in discrete time, meaning that the control computations use sampled-data measurements. We demonstrate the effectiveness of algorithms via analysis, numerical simulations, and experimental testings. We also present extensions to these algorithms that address systems with decentralized control architecture and systems subject to disturbances with unknown frequencies.

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Nadakuditi, Rajesh Rao. "A channel subspace post-filtering approach to adaptive equalization." Thesis, Massachusetts Institute of Technology, 2002. http://hdl.handle.net/1721.1/87613.

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Thesis (S.M.)--Joint Program in Oceanography/Applied Ocean Science and Engineering (Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science; and the Woods Hole Oceanographic Institution), 2002.
Includes bibliographical references (p. 151-154).
by Rajesh Rao Naduditi.
S.M.

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Anderson, Michael-John Peter. "The amalgamation of acoustic and digital audio techniques for the creation of adaptable sound output for musical theatre." Diss., University of Pretoria, 2019. http://hdl.handle.net/2263/76720.

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There are many facets that influence the quality of a musical theatre production. The visual appeal is created from the décor, costumes and lighting, whereas the plot, pace, and relationship a listener develops with the characters are fundamental to the performance quality. However, one often overlooked factor is the impact of sound quality. The perception of sound quality is subjective but is greatly impacted by the environment in which the listener finds themselves. If the projection of the music is underwhelming in depth and expression, or the balance of the dynamics and timbre are badly mixed, this can jeopardise the production’s success, regardless of the quality of the composition or the visual aspects. The production budget for a musical performance can be prohibitive. As a result, prerecorded music is often used as an alternative substitute to live musicians. However, the subjective authenticity of a musical may be jeopardized by the exclusion of live musicians and create additional challenges and performance limitations. One such challenge is the environment in which music will be played. Recorded music is usually created in a single format such as compact disc or for broadcasting, and the cost of recording be can just as expensive as a live performance, especially on large scale works. Time and budget constraints may impact the sound quality. In addition to this, the varying acoustic properties of potential venues may emphasise sonic gaps and flaws contributing to a listener’s negative perception of the sound quality, resulting in a compromised experience of the performance as a whole. This mixed method dissertation offers a systematic explanation to potentially resolve these challenges and limitations by conceptualising established knowledge of sound, audio and acoustics to formulate a framework for adaptive sound. These concepts are put into practice by creating a specifically designed audio recording that is experimented with in multiple theatre scenarios to successfully achieve optimal adaptation of the sound for the theatre environment.
Dissertation (MMus)--University of Pretoria, 2019.
Music
MMus
Unrestricted

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Nunes, Ronaldo Fernandes. "Um estudo do controle ativo de ruidos em dutos usando o algoritmo do minimo erro medio quadratico com referencia filtrada." [s.n.], 1999. http://repositorio.unicamp.br/jspui/handle/REPOSIP/263223.

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Orientador: Jose Maria Campos dos Santos
Dissertação (mestrado) - Universidade Estadual de Campinas, Faculdade de Engenharia Mecanica
Made available in DSpace on 2018-07-25T19:56:09Z (GMT). No. of bitstreams: 1 Nunes_RonaldoFernandes_M.pdf: 5604429 bytes, checksum: 3e2d42b8a7421a42a727f76195057b6e (MD5) Previous issue date: 1999
Resumo: Neste trabalho foi estudado o controle ativo de ruído em dutos usando técnicas de filtragem adaptativa. Foi utilizado o método de controle adaptativo do Mínimo Erro Médio Quadrático (Least Mean Square - LMS) normalizado com sinal de referência filtrado, NFXLMS. Três geometrias de dutos montados em uma bancada experimental para os casos de controle monocanal, multierro e multiexcitação foram verificados. Os sinais de perturbação investigados foram: tonal (seno), aleatório e uma com composição de duas senoidais no caso multiexcitação. Avaliações das impedâncias analíticas e experimentais foram verificadas no estudo de uma configuração do duto para o caso de controle monocanal. Simulações e experimentos foram realizados para diversas geometrias de dutos e diferentes tipos de excitação. Parâmetros do algoritmo, como o número de pesos do filtro adaptativo, faixa de frequência de perturbação, fator de convergência do algoritmo e freqüência de amostragem foram investigados nos casos tratados. Os resultados encontrados nos experimentos realizados para o controlador monocanal e multierro confirmaram as avaliações efetuadas nas simulações. Para o caso multiexcitação, limitações da placa de processamento de sinal não permitiram a obtenção de resultados conclusivos
Abstract: In this work was investigated the active control of noise in ducts using the techniques of adaptive filtering. The normalized, filtered reference Least Mean Square algorithm control method - NXLMS was used. Three shapes of duct system in a supported experimental test rig for the cases of control mono-channel, multi-error, and multi-input were verified. The investigated disturbance signals were: tonal (sine), random, and a composition of two sinusoids for the case of multi-input. Evaluations of analytical and experimental impedances were verified in a configuration study of mono-channel duct contraI case. Simulations and experiments were accomplished for several duct shapes and different types of excitations. AIgorithm parameters, such as the weight number of adaptive filter, the disturbance frequency range, the convergence factor of the algorithm, and the sampling frequency were investigated in the treated cases. The experimental results obtained for the mono-channel and multi-error controller confirm the evaluations obtained in the simulations. For the multi-input case, limitations of the DSP board didn't allow to obtain conclusive results
Mestrado
Mecanica dos Sólidos e Projeto Mecanico
Mestre em Engenharia Mecânica

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Downs, Matthew C. "Adaptive Control Applied to the Cal Poly Spacecraft Attitude Dynamics Simulator." DigitalCommons@CalPoly, 2010. https://digitalcommons.calpoly.edu/theses/231.

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The goal of this thesis is to use the Cal Poly Spacecraft Attitude Dynamics Simulator to provide proof of concept of two adaptive control theories developed by former Cal Poly students: Nonlinear Direct Model Reference Adaptive Control and Adaptive Output Feedback Control. The Spacecraft Attitude Dynamics Simulator is a student-built air bearing spacecraft simulator controlled by four reaction wheels in a pyramidal arrangement. Tests were performed to determine the effectiveness of the two adaptive control theories under nominal operating conditions, a “plug-and-play” spacecraft scenario, and under simulated actuator damage. Proof of concept of the adaptive control theories applied to attitude control of a spacecraft is provided. The adaptive control theories are shown to attain similar or improved performance over a Full State Feedback controller. However, the measurement capabilities of the simulator need to be improved before strong comparisons between the adaptive controllers and Full State Feedback can be achieved.

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Sun, Guohua. "Active Control of Impact Acoustic Noise." University of Cincinnati / OhioLINK, 2013. http://rave.ohiolink.edu/etdc/view?acc_num=ucin1413542213.

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Bapat, Milind S. "New Developments in Fast Boundary Element Method." University of Cincinnati / OhioLINK, 2012. http://rave.ohiolink.edu/etdc/view?acc_num=ucin1331296947.

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Polston, James D. "DECENTRALIZED ADAPTIVE CONTROL FOR UNCERTAIN LINEAR SYSTEMS: TECHNIQUES WITH LOCAL FULL-STATE FEEDBACK OR LOCAL RELATIVE-DEGREE-ONE OUTPUT FEEDBACK." UKnowledge, 2013. http://uknowledge.uky.edu/me_etds/24.

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This thesis presents decentralized model reference adaptive control techniques for systems with full-state feedback and systems with output feedback. The controllers are strictly decentralized, that is, each local controller uses feedback from only local subsystems and no information is shared between local controllers. The full-state feedback decentralized controller is effective for multi-input systems, where the dynamics matrix and control-input matrix are unknown. The decentralized controller achieves asymptotic stabilization and command following in the presence of sinusoidal disturbances with known spectrum. We present a construction technique of the reference-model dynamics such that the decentralized controller is effective for systems with arbitrarily large subsystem interconnections. The output-feedback decentralized controller is effective for single-input single-output subsystems that are minimum phase and relative degree one. The decentralized controller achieves asymptotic stabilization and disturbance rejection in the presence of an unknown disturbance, which is generated by an unknown Lyapunov-stable linear system.

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Rose, John Frederick William. "Visually adaptive virtual acoustic imaging." Thesis, University of Southampton, 2004. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.399992.

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Cheung, Mei Yi S. M. Massachusetts Institute of Technology. "Autonomous adaptive acoustic relay positioning." Thesis, Massachusetts Institute of Technology, 2013. http://hdl.handle.net/1721.1/85504.

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Thesis: S.M., Massachusetts Institute of Technology, Department of Mechanical Engineering, 2013.
Cataloged from PDF version of thesis.
Includes bibliographical references (pages 75-79).
We consider the problem of maximizing underwater acoustic data transmission by adaptively positioning an autonomous mobile relay so as to learn and exploit spatial variations in channel performance. The acoustic channel is the main practical method of underwater wireless communication and improving channel throughput and reliability is key to improving the capabilities of underwater vehicles. Predicting the performance of the acoustic channel in the shallow-water environment is challenging and usually requires extensive modeling of the environment. However, a mobile relay can learn about the unknown channel as it transmits. The relay must balance searching unknown sites to gain more information, which may pay off in the future, and exploiting already-visited sites for immediate reward. This is a classic exploration vs. exploitation problem that is well-described by a multi-armed bandit formulation with an elegant solution in the form of Gittins indices. For an autonomous ocean vehicle traveling between distant waypoints, however, switching costs are significant. The multi-armed bandit with switching costs has no optimal index policy, so we have developed an adaptation of the Gittins index rule with limited policy enumeration and asymptotic performance bounds. We describe extensive shallow-water field experiments conducted in the Charles River (Boston, MA) with autonomous surface vehicles and acoustic modems, and use the field data to assess performance of the MAB decision policies and comparable heuristics. We find the switching-costs-aware algorithm offers superior real-time performance in decision-making and efficient learning of the unknown field.
by Mei Yi Cheung.
S.M.

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Vuksanovic, Branislav. "Electronically controlled acoustic shadows." Thesis, University of Huddersfield, 1998. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.285586.

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Active Noise Control (ANC) is an old concept which has generated increased interest over the past 10-15 years. Using the principle of destructive interference of waves, an inverse pressure wave - "anti-sound wave" is generated in order to attenuate the undesired noise. To achieve substantial cancellation of sound, performance of the cancelling sources must be accurately monitored and controlled. This has only become possible with the rapid development of digital signal processing theory and hardware. Most of the early work in the area of ANC has been done in duct silencing using single channel feed forward and feedback control arrangements. Providing that the sound wavelength is large enough (Le. frequency low enough) in comparison with the cross-sectional dimensions of the duct, spherical sound waves can be adequately approximated with plane waves. The problem is then reduced from three to two dimensions, which provides the possibilities for better understanding of the basic mechanisms of active noise cancellation and study of various adaptive control algorithms. The aim of the present work is to systematically investigate ANC methods for outdoor applications, through the development of Electronically Controlled Acoustic Shadow (ECAS) systems. In this work, the problem is fully three-dimensional. Multichannel ANC methods are proposed to be used, to reduce the noise emitted by large vibrating structures, such as power transformers, in the open air. The adopted approach is to design an active sound wall to create a controlled "anti-sound" shadow. In this way unwanted sound can be reduced in the direction of a complaint area. The potential applications for outdoor ANC systems are considerable. There is need to reduce low frequency sound, which is very hard to reduce using conventional methods - very heavy and expensive structures are required. This opens up the whole field of reducing noise from heavy rotating machinery, such as large generators/motors, factory machinery and mills (many of which have to operate 24 hours per day to remain competitive - which in turn causes noise problems). This work is divided into two main parts. First part considers computer modelling, simulations and theoretical investigation of Electronically Controlled Acoustic Shadows (ECAS) systems. It is demonstrated, that these shadows can be superior to acoustic shadows generated naturally by solid barriers. Detailed analysis predicts that deep shadows (> 1 00 dB) are po.ssible, indicating that practical shadows (>20 dB) are potentially achievable. The object of second part of the work is to investigate practical ECAS systems and establish their performance. In Chapters 2 and 3 (PART 1) the system performance at the fundamental, 100Hz frequency of transformer noise is analysed. To investigate the influence of a large number of parameters on the active wall performance, computer modelling of the primary and secondary (cancelling) sources is developed. The acoustic radiation from this primary source distribution is computed in the far field over a given control angle (both azimuthal and elevation angles). Angles between the 150 and 600 in azimuth and 150 to 300 in elevation are co~'sidered. Phase and amplitude of the secondary sources are than computed through the matrix algebra using exact solution of the least squares problem to minimise the sound at the sensor array. Using this modelling important properties of the acoustic shadows generated by active walls are established, and the basic theory to explain these shadows is formulated. No such theory existed previously. The concept of generating an acoustic shadow in the direction of the complaint area, has resulted in the acoustic properties of a 15°xI5° reference shadow being established in detail. It appears that any arbitrary shadow at this frequency can then be constructed by an addition of the~~ reference shadows, the shadow depth depending on the density of the cancellers per unit angle. Deep shadows in access of 100 dB are predicted, making practical shadows from real sources a possibility. It is now feasible to predict and optimise the future performance of proposed active wall configurations using the computer modelling and developed theory. Further. in the first part of the document (Chapter 4). acoustic interference across high frequency finite Source distributions is studied. The basic theory of non compact sources is considered and the possibility of continuous source representation with a finite number of discrete sources is discussed. The concept of non discreteness or poor discrete representation is established. Here, the .~coustic wavelength is considered small compared to the separation distance between discrete sources. The extent of the near field from these discrete source arrays is also established. where the simplified far field radiation equation breaks down. Finally, in Chapter 4. the optimisation and performance of cancelling arrays to create acoustic shadows from non compact. discrete representation of finite source distributions is investigated.

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35

Wage, Kathleen E. "Adaptive estimation of acoustic normal modes." Thesis, Massachusetts Institute of Technology, 1994. http://hdl.handle.net/1721.1/12096.

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Huo, Jiaquan. "Subband acoustic echo cancellation." Curtin University of Technology, Australian Telecommunications Research Institute, 2004. http://espace.library.curtin.edu.au:80/R/?func=dbin-jump-full&object_id=15802.

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The main theme of this thesis is the control of acoustic echoes for modem voice communication systems by means of echo cancellation. Two important issues in acoustic echo cancellation, namely the efficient adaptation of the echo cancellation filter and the reliable adaptation of the echo cancellation filter in double talk environment, are investigated. The delayless subband adaptive filter architecture is studied. Efficient implementation of the analysis filter bank and the time domain filtering are derived. The transforming of the subband filter weights to a fullhand counterpart is examined. It is shown that the weight transform is a synthesis filtering procedure. Two new weight transform schemes that deliver substantial performance improvements are proposed. The open-loop optimal subband filter impulse responses are shown to be non-causal and several anti-causal laps in the subband filters are required to model this non-causality. Because of the inevitable double talk detection errors, adaptive filtering algorithms with built-in double talk robustness measures are needed for the reliable operation of the echo canceller. The basic idea of robust adaptive filtering is examined. A comparison of different existing time domain robust adaptive filtering algorithms demonstrates that excellent trade-off between the convergence and the tracking properties and the double talk robustness of the adaptive filtering algorithm can he achieved by using Huher’s method for both the update of the echo cancellation filter and the estimation of scale. A delayless closed-loop robust sub- hand adaptive filter is proposed.
By independently adapting the scale estimates and normalizing the adaptation in each subband, significant improvement in terms of the convergence and tracking speed over the time domain robust NLMS algorithm can be obtained without sacrificing the double talk robustness. Moreover, it is demonstrated that by using different thresholds in the update of the echo cancellation filter and the scales, the robust algorithms converge and track echo path variation as fast as their non-robust counter part while still maintaining a sufficiently low sensitivity to double talk detection errors. The application of two path adaptive filters to acoustic echo cancellation is examined. An analysis of the original two path adaptive filtering algorithm shows that it suffers from two kinds of performance degradation due to the divergence of the background filter during double talk, namely the slow tracking of echo path variation and the false filter coefficient copying after double talk. A robust two path adaptive filter is proposed to mitigate these problems.

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Fee, D. T. "Dereverberation of acoustic signals via adaptive filtering." Thesis, Queen's University Belfast, 2006. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.438629.

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38

Chen, Wu-Nan. "Multiple microphone voice activity detection and adaptive noise cancellation." Thesis, University of the West of Scotland, 2001. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.365083.

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Herbordt, Wolfgang. "Combination of robust adaptive beamforming with acoustic echo cancellation for acoustic human-machine interfaces." [S.l. : s.n.], 2004. http://deposit.ddb.de/cgi-bin/dokserv?idn=972527907.

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40

Birkett, A. Neil. "Nonlinear adaptive filtering with application to acoustic echo cancellation." Thesis, National Library of Canada = Bibliothèque nationale du Canada, 1997. http://www.collectionscanada.ca/obj/s4/f2/dsk2/ftp03/NQ26845.pdf.

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41

Alvarez-Tinoco, Antonio Mario. "Adaptive algorithms for the active attenuation of acoustic noise." Thesis, Heriot-Watt University, 1985. http://hdl.handle.net/10399/1607.

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42

Lyness, A. D. "Adaptive compensation for acoustic impairments using multichannel surround sound." Thesis, Queen's University Belfast, 2004. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.411128.

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43

Birkett, A. Neil (Alexander Neil) Carleton University Dissertation Engineering Systems and Computer. "Nonlinear adaptive filtering with application to acoustic echo cancellation." Ottawa, 1997.

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44

Chen, Teyan. "Novel adaptive signal processing techniques for underwater acoustic communications." Thesis, University of York, 2011. http://etheses.whiterose.ac.uk/1925/.

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The underwater acoustic channel is characterized by time-varying multipath propagation with large delay spreads of up to hundreds of milliseconds, which introduces severe intersymbol interference (ISI) in digital communication system. Many of the existing channel estimation and equalization techniques used in radio frequency wireless communication systems might be practically inapplicable to underwater acoustic communication due to their high computational complexity. The recursive least squares (RLS)-dichotomous coordinate descent (DCD) algorithm has been recently proposed and shown to perform closely to the classical RLS algorithm while having a significantly lower complexity. It is therefore a highly promising channel estimation algorithm for underwater acoustic communications. However, predicting the convergence performance of the RLS-DCD algorithm is an open issue. Known approaches are found not applicable, as in the RLS-DCD algorithm, the normal equations are not exactly solved at every time instant and the sign function is involved at every update of the filter weights. In this thesis, we introduce an approach for convergence analysis of the RLS-DCD algorithm based on computations with only deterministic correlation quantities. Equalization is a well known method for combatting the ISI in communication channels. Coefficients of an adaptive equalizer can be computed without explicit channel estimation using the channel output and known pilot signal. Channel-estimate (CE) based equalizers which re-compute equalizer coefficients for every update of the channel estimate, can outperform equalizers with the direct adaptation. However, the computational complexity of CE based equalizers for channels with large delay spread, such as the underwater acoustic channel, is an open issue. In this thesis, we propose a low-complexity CE based adaptive linear equalizer, which exploits DCD iterations for computation of equalizer coefficients. The proposed technique has as low complexity as O(Nu(K+M)) operations per sample, where K and M are the equalizer and channel estimator length, respectively, and Nu is the number of iterations such that Nu << K and Nu << M. Moreover, when using the RLS-DCD algorithm for channel estimation, the computation of equalizer coefficients is multiplication-free and division-free, which makes the equalizer attractive for hardware design. Simulation results show that the proposed adaptive equalizer performs close to the minimum mean-square-error (MMSE) equalizer with perfect knowledge of the channel. Decision feedback equalizers (DFEs) can outperform LEs, provided that the effect of decision errors on performance is negligible. However, the complexity of existing CE based DFEs normally grows squarely with the feedforward filter (FFF) length K. In multipath channels with large delay spread and long precursor part, such as in underwater acoustic channels, the FFF length K needs to be large enough to equalize the precursor part, and it is usual that K > M. Reducing the complexity of CE based DFEs in such scenarios is still an open issue. In this thesis, we derive two low complexity approaches for computing CE based DFE coefficients. The proposed DFEs operate together with partial-update channel estimators, such as the RLS-DCD channel estimator, and exploit complex-valued DCD iterations to efficiently compute the DFE coefficients. In the first approach, the proposed DFE has a complexity of O(Nu l log 2l) real multiplications per sample, where l is the equalizer delay and Nu is the number of iterations such that Nu << l. In the second proposed approach, DFE has a complexity as low as O(Nu K)+O(Nu B) + O(Nu M) operations per sample, where B is the feedback filter (FBF) length and Nu << M. Moreover, when the channel estimator also exploits the DCD iterations, e.g. such as in the RLS-DCD adaptive filter, the second approach is multiplication-free and division-free, which makes the equalizer attractive for hardware implementation. Simulation results show that the proposed DFEs perform close to the RLS CE based DFE, where the CE is obtained using the classical RLS adaptive filter and the equalizer coefficients are computed according to the MMSE criterion. Localization is an important problem for many underwater communication systems, such as underwater sensor networks. Due to the characteristics of the underwater acoustic channel, localization of underwater acoustic sources is challenging and needs to be accurate and computationally efficient. The matched-phase coherent broadband matched-field (MF) processor has been previously proposed and shown to outperform other advanced broadband MF processors for underwater acoustic source localization. It has been previously proposed to search the matched phases using the simulated annealing, which is well known for its ability for solving global optimization problems while having high computational complexity. This prevents simultaneous processing of many frequencies, and thus, limits the processor performance. In this thesis, we introduce a novel iterative technique based on coordinate descent optimization, the phase descent search (PDS), for searching the matched phases. We show that the PDS algorithm obtains matched phases similar to that obtained by the simulated annealing, and has significantly lower complexity. Therefore, it enables to search phases for a large number of frequencies and significantly improves the processor performance. The proposed processor is applied to experimental data for locating a moving acoustic source and shown to provide accurate localization of the source well matched to GPS measurements.

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Finefield, John K. "Investigation of combined feedback and adaptive control of cylinder vibrations." Thesis, This resource online, 1992. http://scholar.lib.vt.edu/theses/available/etd-10062009-020152/.

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Moat, Trevor P. B. M. "Orthogonal adaptive digital filters with applications to acoustic system identification." Thesis, National Library of Canada = Bibliothèque nationale du Canada, 1998. http://www.collectionscanada.ca/obj/s4/f2/dsk2/tape17/PQDD_0025/MQ27022.pdf.

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Puikkonen, Panu. "Development of an adaptive equalization algorithm using acoustic energy density /." Diss., CLICK HERE for online access, 2009. http://contentdm.lib.byu.edu/ETD/image/etd2899.pdf.

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48

Mulholland, P. J. "Adaptive filters and their application to an adaptive receiving array for an underwater acoustic data link." Thesis, University of Newcastle Upon Tyne, 1988. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.234508.

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Pushparajah, Sudarini Jayani. "Frequency domain adaptive algorithms for acoustic echo cancellation during double talk." Thesis, Imperial College London, 2000. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.392886.

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Hart, Joanna Elizabeth. "Multirate sub-band structures with application to adaptive acoustic echo cancellation." Thesis, Imperial College London, 1999. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.300759.

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Dissertations / Theses on the topic 'Adaptive acoustics'

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