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Quintero, Florez Victor. "Noisy channel-output feedback in the interference channel." Thesis, Lyon, 2017. http://www.theses.fr/2017LYSEI128/document.
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Dans cette thèse, le canal Gaussien à interférence à deux utilisateurs avec voie de retour dégradée par un bruit additif (GIC-NOF) est étudié sous deux perspectives : les réseaux centralisés et décentralisés. Du point de vue des réseaux centralisés, les limites fondamentales du GIC-NOF sont caractérisées par la région de capacité. L’une des principales contributions de cette thèse est une approximation à un nombre constant de bits près de la région de capacité du GIC-NOF. Ce résultat est obtenu grâce à l’analyse d’un modèle de canal plus simple, le canal linéaire déterministe à interférence à deux utilisateurs avec voie de retour dégradée par un bruit additif (LDIC-NOF). L’analyse pour obtenir la région de capacité du LDIC-NOF fournit les idées principales pour l’analyse du GIC-NOF. Du point de vue des réseaux décentralisés, les limites fondamentales du GIC-NOF sont caractérisées par la région d’η-équilibre de Nash (η-EN). Une autre contribution de cette thèse est une approximation de la région η-EN du GIC-NOF, avec η > 1. Comme dans le cas centralisé, le cas décentralisé LDIC-NOF (D-LDIC-NOF) est étudié en premier et les observations sont appliquées dans le cas décentralisé GIC-NOF (D-GIC-NOF). La contribution finale de cette thèse répond à la question suivante : “À quelles conditions la voie de retour permet d’agrandir la région de capacité, la région η-EN du GIC-NOF ou du D-GIC-NOF ? ”. La réponse obtenue est de la forme : L’implémentation de la voie de retour de la sortie du canal dans l’émetteur-récepteur i agrandit la région de capacité ou la région η-EN si le rapport signal sur bruit de la voie de retour est supérieure à SNRi* , avec i ∈ {1, 2}. La valeur approximative de SNRi* est une fonction de tous les autres paramètres du GIC-NOF ou du D-GIC-NOFIn this thesis, the two-user Gaussian interference channel with noisy channel-output feedback (GIC-NOF) is studied from two perspectives: centralized and decentralized networks. From the perspective of centralized networks, the fundamental limits of the two-user GICNOF are characterized by the capacity region. One of the main contributions of this thesis is an approximation to within a constant number of bits of the capacity region of the two-user GIC-NOF. This result is obtained through the analysis of a simpler channel model, i.e., a two-user linear deterministic interference channel with noisy channel-output feedback (LDIC-NOF). The analysis to obtain the capacity region of the two-user LDIC-NOF provides the main insights required to analyze the two-user GIC-NOF. From the perspective of decentralized networks, the fundamental limits of the two-user decentralized GIC-NOF (D-GIC-NOF) are characterized by the η-Nash equilibrium (η-NE) region. Another contribution of this thesis is an approximation of the η-NE region of the two-user GIC-NOF, with η> 1. As in the centralized case, the two-user decentralized LDIC-NOF (D-LDIC-NOF) is studied first and the lessons learnt are applied in the two-user D-GIC-NOF. The final contribution of this thesis consists in a closed-form answer to the question: “When does channel-output feedback enlarge the capacity or η-NE regions of the two-user GIC-NOF or two-user D-GIC-NOF?”. This answer is of the form: Implementing channel-output feedback in transmitter-receiver i enlarges the capacity or η-NE regions if the feedback SNR is beyond SNRi* , with i ∈ {1, 2}. The approximate value of SNRi* is shown to be a function of all the other parameters of the two-user GIC-NOF or two-user D-GIC-NOF
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Sollund, Tomas. "Dirty-paper coding over noisy feedback channels with ISI." Thesis, McGill University, 2009. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=97783.
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This work generalizes the Liu and Elia dirty-paper coding scheme to include feedback channels with additive white Gaussian noise (AWGN) and intersymbol interference (ISI). An optimal scheme is derived which achieves lossless interference cancellation, performs arbitrarily close to the feedback capacity for a fixed encoder dimension, and yields a probability of error which decays to zero doubly exponentially with the number of channel uses. This scheme is further modified to mitigate inherent numerical instability problems, avoiding the need to solve a non-convex optimization problem, and at the same time minimizing the coding delay for a given probability of symbol error, at a cost of larger transmission power. Noise and parametric model uncertainties for the channels are considered, and knowledge of upper bounds on the noise variances are shown to yield a guaranteed performance level for the coding scheme, where the performance measures are probability of symbol error vs. number of channel uses, as well as probability of symbol error vs. signal-to-noise ratio (SNR). Computer simulations are conducted to validate the results.Cette thèse généralise la technique du Dirty Paper Coding de Liu et Elia en incluant un canal de renvoi avec bruit blanc Gaussien additif (BBGA) et brouillage intersymbole. Il en découle une approche optimale qui permet l’élimination sans perte de l’interférence et dont la performance peut être aussi proche que voulue de la capacité du model avec renvoi pour une taille de codeur donnée. De plus, la probabilité d’erreur associée à cette technique décroît doublement exponentiellement vers zéro en fonction du nombre d’usages du canal. Une modification additionnelle est apportée à notre algorithme pour éviter de résoudre un problème d’optimisation non-convexe tout en minimisant le délai de codage pour une probabilité d’erreur de symbole donnée, et ceci au prix d’un accroissement de la puissance de transmission requise. Cette modification permet en outre de rendre l’algorithme plus robuste face aux divers problèmes d’instabilités numériques. En prenant en compte aussi bien le bruit que les incertitudes du modèle paramétrique du canal, nous montrons que le fait de connaître les limites supérieures des variances du bruit, permet de garantir un certain niveau de performance pour la technique de codage. Ici les mesures de performance sont la probabilité d’erreur de symbole en fonction du nombre d’usages du canal et la probabilité d’erreur de symbole en fonction du rapport signal-bruit. Des simulations par ordinateur sont ensuite conduites pour valider nos résultats.
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Nagasubramanian, Karthik. "Code design for erasure channels with limited or noisy feedback." [College Station, Tex. : Texas A&M University, 2007. http://hdl.handle.net/1969.1/ETD-TAMU-2065.
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Tukhlina, Natalia. "Feedback control of complex oscillatory systems." Phd thesis, Universität Potsdam, 2008. http://opus.kobv.de/ubp/volltexte/2008/1854/.
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In the present dissertation paper an approach which ensures an efficient control of such diverse systems as noisy or chaotic oscillators and neural ensembles is developed. This approach is implemented by a simple linear feedback loop. The dissertation paper consists of two main parts.
One part of the work is dedicated to the application of the suggested technique to a population of neurons with a goal to suppress their synchronous collective dynamics.
The other part is aimed at investigating linear feedback control of coherence of a noisy or chaotic self-sustained oscillator.
First we start with a problem of suppressing synchronization in a large population of interacting neurons.
The importance of this task is based on the hypothesis that emergence of pathological brain activity in the case of Parkinson's disease and other neurological disorders is caused by synchrony of many thousands of neurons.
The established therapy for the patients with such disorders is a permanent high-frequency electrical stimulation via the depth microelectrodes, called Deep Brain Stimulation (DBS). In spite of efficiency of such stimulation, it has several side effects and mechanisms underlying DBS remain unclear.
In the present work an efficient and simple control technique is suggested. It is designed to ensure suppression of synchrony in a neural ensemble by a minimized stimulation that vanishes as soon as the tremor is suppressed. This vanishing-stimulation technique would be a useful tool of experimental neuroscience; on the other hand, control of collective dynamics in a large population of units represents an interesting physical problem. The main idea of suggested approach is related to the classical problem of oscillation theory, namely the interaction between a self-sustained (active) oscillator and a passive load (resonator). It is known that under certain conditions the passive oscillator can suppress the oscillations of an active one. In this thesis a much more complicated case of active medium, which itself consists of thousands of oscillators is considered. Coupling this medium to a specially designed passive oscillator, one can control the collective motion of the ensemble, specifically can enhance or suppress it. Having in mind a possible application in neuroscience, the problem of suppression is concentrated upon.
Second, the efficiency of suggested suppression scheme is illustrated by considering more complex case, i.e. when the population of neurons generating the undesired rhythm consists of two non-overlapping subpopulations: the first one is affected by the stimulation,
while the collective activity is registered from the second one. Generally speaking, the second population can be by itself both active and passive; both cases are considered here. The possible applications of suggested technique are discussed.
Third, the influence of the external linear feedback on coherence of a noisy or chaotic self-sustained oscillator is considered. Coherence is one of the main properties of self-oscillating systems and plays a key role in the construction of clocks, electronic generators, lasers, etc. The coherence of a noisy limit cycle oscillator in the context of phase dynamics is evaluated by the phase diffusion constant, which is in its turn proportional to the width of the spectral peak of oscillations. Many chaotic oscillators can be described within the framework of phase dynamics, and, therefore, their coherence can be also quantified by the way of the phase diffusion constant. The analytical theory for a general linear feedback, considering noisy systems in the linear and Gaussian approximation is developed and validated by numerical results.In der vorliegenden Dissertation wird eine Näherung entwickelt, die eine effiziente Kontrolle verschiedener Systeme wie verrauschten oder chaotischen Oszillatoren und Neuronenensembles ermöglicht. Diese Näherung wird durch eine einfache lineare Rückkopplungsschleife implementiert. Die Dissertation besteht aus zwei Teilen. Ein Teil der Arbeit ist der Anwendung der vorgeschlagenen Technik auf eine Population von Neuronen gewidmet, mit dem Ziel ihre synchrone Dynamik zu unterdrücken. Der zweite Teil ist auf die Untersuchung der linearen Feedback-Kontrolle der Kohärenz eines verrauschten oder chaotischen, selbst erregenden Oszillators gerichtet. Zunächst widmen wir uns dem Problem, die Synchronisation in einer großen Population von aufeinander wirkenden Neuronen zu unterdrücken. Da angenommen wird, dass das Auftreten pathologischer Gehirntätigkeit, wie im Falle der Parkinsonschen Krankheit oder bei Epilepsie, auf die Synchronisation großer Neuronenpopulation zurück zu führen ist, ist das Verständnis dieser Prozesse von tragender Bedeutung. Die Standardtherapie bei derartigen Erkrankungen besteht in einer dauerhaften, hochfrequenten, intrakraniellen Hirnstimulation mittels implantierter Elektroden (Deep Brain Stimulation, DBS). Trotz der Wirksamkeit solcher Stimulationen können verschiedene Nebenwirkungen auftreten, und die Mechanismen, die der DBS zu Grunde liegen sind nicht klar. In meiner Arbeit schlage ich eine effiziente und einfache Kontrolltechnik vor, die die Synchronisation in einem Neuronenensemble durch eine minimierte Anregung unterdrückt und minimalinvasiv ist, da die Anregung stoppt, sobald der Tremor erfolgreich unterdrückt wurde. Diese Technik der "schwindenden Anregung" wäre ein nützliches Werkzeug der experimentellen Neurowissenschaft. Desweiteren stellt die Kontrolle der kollektiven Dynamik in einer großen Population von Einheiten ein interessantes physikalisches Problem dar. Der Grundansatz der Näherung ist eng mit dem klassischen Problem der Schwingungstheorie verwandt - der Interaktion eines selbst erregenden (aktiven) Oszillators und einer passiven Last, dem Resonator. Ich betrachte den deutlich komplexeren Fall eines aktiven Mediums, welches aus vielen tausenden Oszillatoren besteht. Durch Kopplung dieses Mediums an einen speziell hierür konzipierten, passiven Oszillator kann man die kollektive Bewegung des Ensembles kontrollieren, um diese zu erhöhen oder zu unterdrücken. Mit Hinblick auf eine möglichen Anwendung im Bereich der Neurowissenschaften, konzentriere ich mich hierbei auf das Problem der Unterdrückung. Im zweiten Teil wird die Wirksamkeit dieses Unterdrückungsschemas im Rahmen eines komplexeren Falles, bei dem die Population von Neuronen, die einen unerwünschten Rhythmus erzeugen, aus zwei nicht überlappenden Subpopulationen besteht, dargestellt. Zunächst wird eine der beiden Subpopulationen durch Stimulation beeinflusst und die kollektive Aktivität an der zweiten Subpopulation gemessen. Im Allgemeinen kann sich die zweite Subpopulation sowohl aktiv als auch passiv verhalten. Beide Fälle werden eingehend betrachtet. Anschließend werden die möglichen Anwendungen der vorgeschlagenen Technik besprochen. Danach werden verschiedene Betrachtungen über den Einfluss des externen linearen Feedbacks auf die Kohärenz eines verrauschten oder chaotischen selbst erregenden Oszillators angestellt. Kohärenz ist eine Grundeigenschaft schwingender Systeme und spielt ein tragende Rolle bei der Konstruktion von Uhren, Generatoren oder Lasern. Die Kohärenz eines verrauschten Grenzzyklus Oszillators im Sinne der Phasendynamik wird durch die Phasendiffusionskonstante bewertet, die ihrerseits zur Breite der spektralen Spitze von Schwingungen proportional ist. Viele chaotische Oszillatoren können im Rahmen der Phasendynamik beschrieben werden, weshalb ihre Kohärenz auch über die Phasendiffusionskonstante gemessen werden kann. Die analytische Theorie eines allgemeinen linearen Feedbacks in der Gaußschen, als auch in der linearen, Näherung wird entwickelt und durch numerische Ergebnisse gestützt.
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Chande, Vinay. "Progressive source-channel coding for multimedia transmission over noisy and lossy channels with and without feedback." College Park, Md. : University of Maryland, 2004. http://hdl.handle.net/1903/1752.
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Thesis (Ph. D.) -- University of Maryland, College Park, 2004.Thesis research directed by: Electrical Engineering. Title from t.p. of PDF. Includes bibliographical references. Published by UMI Dissertation Services, Ann Arbor, Mich. Also available in paper.
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Duke, Cole Victor. "Analog Feedback Control of Broadband Fan Noise." BYU ScholarsArchive, 2012. https://scholarsarchive.byu.edu/etd/3646.
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Active noise control (ANC) has been implemented using analog filters to reduce broadband noise from a small axial cooling fan. Previous work successfully attenuated narrow-band, tonal portions of the noise using a digital controller. The practical performance limits of this system were reached and it was desirable to attenuate the noise further. Additional research, therefore, sought to attenuate broadband noise from the fan using a digital controller, but performance was limited by the group delay inherent in the digital signal processor (DSP). Current research attempts to further attenuate broadband noise and improve performance of the system by combining the tonal controller with an analog feedback controller. An analog controller is implemented in parallel with the digital controller without degrading the performance of either individual controller. Broadband noise is attenuated in a certain frequency region, but at the expense of increasing noise in adjacent frequency regions. Results show that a single-input single-output (SISO) controller is preferable to a multiple-input multiple-output (MIMO) controller for this system.
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Boglione, Luciano. "Low noise microwave feedback amplifier design with simultaneous signal and noise matching." Thesis, University of Leeds, 1998. http://etheses.whiterose.ac.uk/900/.
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This thesis looks into the problem of simultaneous signal and noise match at the input port of low noise amplifiers; feedback LNAs are considered because previous works show that they can achieve the simultaneous match condition. The investigation analyses the influence of both parallel and series feedback elements on the amplifier. Matrices are used to describe signal and noise parameters of each component of the model - parallel admittance, series impedance, active device. This approach allows the analysis to be applied to a wide range of networks, as long as noise and signal matrices are available. For this reason, the results are not limited to active devices in the microwave region of the spectrum but they are applicable to any linear 2-port circuit. The noise parameters of feedback networks are investigated thoroughly. Analytical expressions are worked out as functions of the feedback immittances and have been used to support experimental evidence previously published. A duality property for feedback networks is pointed out; new circles for constant equivalent noise resistance are devised; optimum values for the feedback impedance are determined; an investigation of a well-known noise model is carried out and its validity is extended. Based on the closed form expressions of the noise parameters, an original analytical procedure for the design of the optimum noise source reflection coefficient is presented. To the author's knowledge, no technique was available before. The design for simultaneous signal and noise match is now possible, because the input reflection coefficient can be set independently by properly choosing the load. Different devices are considered and their different behaviour is highlighted. A remarkable feature of the new design technique is to avoid the need of input matching when designing low noise amplifiers. Finally, experimental results are also presented and the performance of aI GHz single stage BJT LNA is shown. The fundamental achievement is that the noise figure of the LNA is equal to its minimum value within the measurement uncertainty.
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Pawełczyk, Marek. "Feedback control of acoustic noise at desired locations." Praca habilitacyjna, Wydawnictwo Politechniki Śląskiej, 2005. https://delibra.bg.polsl.pl/dlibra/docmetadata?showContent=true&id=10178.
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Longtin, André. "Nonlinear oscillations, noise and chaos in neural delayed feedback." Thesis, McGill University, 1989. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=74311.
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Bifurcations and complex oscillations in the human pupil light reflex (PLR) are studied. Autonomous pupil area oscillations are produced by substituting electronically controllable nonlinear feedback for the normal negative feedback of this reflex. A physiologically sound theoretical framework in which to study pupillary oscillations is developed. The model, framed as a delay-differential equation (DDE), agrees quantitatively with the simpler periodic behaviors and qualitatively with the complex behaviors. Much of the aperiodicity in the data can be ascribed to noise and transients rather than to chaos. The critical behavior of the PLR at oscillation onset is different with piecewise constant rather than smooth negative feedback. In the former, relative fluctuations in period are larger than those in amplitude, and vice versa in the latter. Properties of the time solutions and densities of a stochastic DDE are used to explain this experimental result. The Hopf bifurcation in this system is postponed by both additive and multiplicative colored noise. Theoretical insight into the behavior of stationary densities of DDE's and the origin of the postponement is given, and implications for analyzing bifurcations in neural delayed feedback systems are discussed.
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Raja, Ahmad Raja Mohd Kamil. "Minimum effort active noise control with feedback inclusion architecture." Thesis, University of Sheffield, 2006. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.443903.
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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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Saxena, Bhavaye. "Noise Characteristics for Random Fiber Lasers with Rayleigh Distributed Feedback." Thesis, Université d'Ottawa / University of Ottawa, 2014. http://hdl.handle.net/10393/31766.
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Frequency and intensity noise are characterized for Erbium-Doped Fiber and Brillouin random lasers based on Rayleigh distributed feedback mechanism. We propose a theoretical model for the frequency noise of an Er-doped fiber random lasers using the property of random phase modulations from multiple scattering points in ultra-long fibers. We find that the Rayleigh feedback suppresses the noise at higher frequencies by introducing a Lorentzian envelope over the thermal frequency noise of a long fiber cavity. The theoretical model and measured frequency noise agree quantitatively with two fitting parameters. A similar model, which also includes additional acoustic fluctuations and a distributed gain profile in the fiber, has been speculated for the Brillouin random laser. These random laser exhibits a frequency noise level of < 6 Hz^2/Hz at 2 kHz, which is lower than what is found in conventional narrow-linewidth EDF fiber lasers and Nonplanar Ring Laser oscillators (NPRO) by a factor of 166 and 2 respectively.
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Klein, Thierry Etienne 1971. "Capacity of Gaussian noise channels with side information and feedback." Thesis, Massachusetts Institute of Technology, 2001. http://hdl.handle.net/1721.1/8984.
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Thesis (Ph.D.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 2001.Includes bibliographical references (p. 295-306).
In wireless communication systems, the communication channel is often modeled as a fading multiaccess channel with time-varying multipath. Because of the mobility of the users and the varying number of users, the transmission conditions are constantly changing. These characteristics of the channel, as well as the ever-growing number of users competing for limited resources, call for an efficient use of the available power and bandwidth resources. In order to achieve a maximum system efficiency, the transmission conditions have to be constantly monitored and used to update the transmission strategies. The transmitters usually have some form of information regarding the channel behavior. The main question of interest is to determine how to best use this information in order to increase the transmission rates, decrease the error probability and conserve resources. In this thesis, we present an information-theoretic analysis of the single-user channel with side information and feedback. The first part of this work is devoted to the discrete-time, finite-state additive white Gaussian noise channel. Under perfect instantaneous side-information, the capacity achieving power allocation is determined by the well-known water-filling procedure. The basic water-filling power allocation is extended to include minimal rate and / or maximal power constraints. Various imperfections in the side-information and their influence on capacity and the power allocation are studied next. These imperfections include delayed side-information, errors in the transmission of side-information, and errors in the estimation of the channel conditions. We also present a universal power control algorithm that does not assume knowledge of the statistical behavior of the channel. The feedback capacity of the finite-state channel is investigated and upper and lower bounds are derived. In the second part of the thesis, we concentrate on the feedback capacity of the colored Gaussian noise channel. The capacity of this channel is still an open problem and most research work has focused on finding upper bounds. We propose a new, tighter lower bound on the feedback capacity. This bound is very general in the sense that it can be applied to any noise covariance matrix and can be computed for both the finite and the infinite time horizon cases. A sufficient condition is obtained on the average available power such that feedback strictly increases capacity. An interesting and intriguing consequence is reached that sheds new light on the role of the feedback. Specifically it is shown that feedback should not be used to cancel out the noise process, but rather to transmit information about the noise to the receiver. Finally, two problems related to the Gaussian multi-access channel are examined. First, new bounds on the maximum sum rate and the capacity region with feedback are derived. Several previously published bounds can be viewed as special cases of these bounds. Second, we compute the optimal power allocation vector that guarantees that a fixed target rate tuple is achievable, while at the same time minimizing the sum of the transmit powers used by the individual users. The optimal decoding order is shown to be independent of the target rate tuple.
by Thierry Etienne Klein.
Ph.D.
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Green, Matthew J. "Feedback Applications in Active Noise Control for Small Axial Cooling Fans." Diss., CLICK HERE for online access, 2006. http://contentdm.lib.byu.edu/ETD/image/etd1539.pdf.
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Amoêdo, David Jorge Tiago. "A 1.2 V low noise amplifier with double feedback for high gain and low noise figure." Master's thesis, Faculdade de Ciências e Tecnologia, 2013. http://hdl.handle.net/10362/11040.
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Dissertação para obtenção do Grau de Mestre em
Engenharia Eletrotécnica e de ComputadoresIn this thesis we present a balun low noise amplifier (LNA) in which the gain is boosted using a double feedback structure. The circuit is based in a Balun LNA with noise and distortion cancellation. The LNA is based in two basic stages: common-gate (CG) and common-source (CS). We propose to replace the resistors by active loads, which have two inputs that will be used to provide the feedback (in the CG and CS stages). This proposed methodology will boost the gain and reduce the NF (Noise Figure). Simulation results, with a 130 nm CMOS technology, show that the gain is 19.65 dB and the NF is less than 2.17 dB. The total power dissipation is only 5 mW (since no extra blocks are required), leading to an FOM (Figure of Merit) of 3.13 mW-1 from a nominal 1.2 supply.
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Li, Shih-Hung. "Progressive learning of endpoint feedback systems with model uncertainty and sensor noise." Thesis, Massachusetts Institute of Technology, 1996. http://hdl.handle.net/1721.1/38158.
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Kawamura, Yoji. "Spatiotemporal chaos in coupled oscillator systems with global feedback or external noise." 京都大学 (Kyoto University), 2007. http://hdl.handle.net/2433/136746.
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Xiong, Zhijie. "Radio Frequency Low Noise and High Q Integrated Filters in Digital CMOS Processes." Diss., Georgia Institute of Technology, 2004. http://hdl.handle.net/1853/5043.
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Radio Frequency Low Noise and High Q
Integrated Filters in Digital CMOS Processes
Zhijie Xiong
149 pages
Directed by Dr. Phillip E. Allen
Presented in this work is a novel design technique for CMOS integration of RF high Q integrated filters using positive feedback and current mode approach. Two circuits are designed in this work: a 100MHz low-noise and high Q bandpass filter suited for an FM radio front-end, and a 2.4GHz low-noise and high-Q bandpass filter suited for a Bluetooth front-end. Current-mode approach and positive feedback design techniques are successfully used in the design of both circuits. Both circuits are fabricated through a 0.18um CMOS process provided by National Semiconductor Corp.
The 100MHz circuit achieves 3.15uV RF sensitivity with 26dB SNR, and the total current consumption is 12mA. The center frequency of the filter is tunable from 80MHz to 110MHz, and the Q value is tunable from 0.5 to 28.9. 1 dB compression point is measured as -34.0dBm, combined with noise measurement results, a dynamic range of 54.1 dB results. Silicon area of the core circuit is 0.4 square millimeters.
The center frequency of the 2.4GHz circuit is tunable from 2.4GHz to 2.5GHz, and the Q value is tunable from 20 to 120. The 1 dB compression dynamic range of the circuit is 50dB. Integrated spiral inductors are developed for this design. Patterned ground shields are laid out to reduce inductor loss through substrate, especially eddy current loss when the circuit is fabricated on epi wafers. Accumulation mode MOS varactors are designed to tune the frequency response. Silicon area of the core circuit is 1 square millimeter.
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Weststrate, Marnus. "LC-ladder and capacitive shunt-shunt feedback LNA modelling for wideband HBT receivers." Thesis, University of Pretoria, 2011. http://hdl.handle.net/2263/26615.
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Although the majority of wireless receiver subsystems have moved to digital signal processing over the last decade, the low noise amplifier (LNA) remains a crucial analogue subsystem in any design being the dominant subsystem in determining the noise figure (NF) and dynamic range of the receiver as a whole. In this research a novel LNA configuration, namely the LC-ladder and capacitive shunt-shunt feedback topology, was proposed for use in the implementation of very wideband LNAs. This was done after a thorough theoretical investigation of LNA configurations available in the body of knowledge from which it became apparent that for the most part narrowband LNA configurations are applied to wideband applications with suboptimal results, and also that the wideband configurations that exist have certain shortcomings. A mathematical model was derived to describe the new configuration and consists of equations for the input impedance, input return loss, gain and NF, as well as an approximation of the worst case IIP3. Compact design equations were also derived from this model and a design strategy was given which allows for electronic design automation of a LNA using this configuration. A process for simultaneously optimizing the circuit for minimum NF and maximum gain was deduced from this model and different means of improving the linearity of the LNA were given. This proposed design process was used successfully throughout this research. The accuracy of the mathematical model has been verified using simulations. Two versions of the LNA were also fabricated and the measured results compared well with these simulations. The good correlation found between the calculated, simulated and measured results prove the accuracy of the model, and some comments on how the accuracy of the model could be improved even further are provided as well. The simulated results of a LNA designed for the 1 GHz to 18 GHz band in the IBM 8HP process show a gain of 21.4 dB and a minimum NF of only 1.7 dB, increasing to 3.3 dB at the upper corner frequency while maintaining an input return loss below -10 dB. After steps were taken to improve the linearity, the IIP3 of the LNA is -14.5 dBm with only a small degradation in NF now 2.15 dB at the minimum. The power consumption of the respective LNAs are 12.75 mW and 23.25 mW and each LNA occupies a chip area of only 0.43 mm2. Measured results of the LNA fabricated in the IBM 7WL process had a gain of 10 dB compared to an expected simulated gain of 20 dB, however significant path loss was introduced by the IC package and PCB parasitics. The S11 tracked the simulated response very well and remained below -10 dB over the feasible frequency range. Reliable noise figure measurements could not be obtained. The measured P1dB compression point is -22 dBm. A 60 GHz LNA was also designed using this topology in a SiGe process with ƒT of 200 GHz. A simulated NF of 5.2 dB was achieved for a gain of 14.2 dB and an input return loss below -15 dB using three amplifier stages. The IIP3 of the LNA is -8.4 dBm and the power consumption 25.5 mW. Although these are acceptable results in the mm-wave range it was however found that the wideband nature of this configuration is redundant in the unlicensed 60 GHz band and results are often inconsistent with the design theory due to second order effects. The wideband results however prove that the LC-ladder and capacitive shunt-shunt feedback topology is a viable means for especially implementing LNAs that require a very wide operating frequency range and also very low NF over that range.Thesis (PhD(Eng))--University of Pretoria, 2011.
Electrical, Electronic and Computer Engineering
unrestricted
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Zangi, Kambiz Casey. "Optimal feedback control formulation of the active noise cancellation problem : pointwise and distributed." Thesis, Massachusetts Institute of Technology, 1994. http://hdl.handle.net/1721.1/12215.
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Thesis (Ph. D.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 1994.Includes bibliographical references (p. 151-156).
by Kambiz C. Zangi.
Ph.D.
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White, Andrew. "On Implemintation of Loudspeakers for Feedback Control, Open-Air, Active Noise Control Headsets." Thesis, Virginia Tech, 1999. http://hdl.handle.net/10919/35936.
Full textAbstract:
The loudspeakers used in active noise reduction (ANR) headsets are generally identical to loudspeakers used in commercial headphones. Unfortunately, the frequency response characteristics of these loudspeakers are not particularly well suited for open-air active noise control (ANC). Open-air headsets float outside the ear with no contact between the system and the user and allow for regular conversation with others in the environment.
This study has identified three limitations on the closed-loop performance of open-air headsets: the distribution of gain and phase in the loudspeaker's open-loop frequency response function, manufacturing variations in loudspeakers that can deviate from design specifications by up to 40%, and the variations in acoustic impedance coupling (ear-to-speaker) among users. This thesis explores the mechanisms that underlie these limitations with the goal of designing open-air headsets that are robust to manufacturing and user variations. Methods are introduced on ways to minimize the effects of manufacturing and user variations and are proven by experiment. With these variations minimized, the controller's design is only limited by the frequency response of the loudspeaker. A comprehensive examination of techniques to model moving-coil loudspeakers is presented followed by detailed studies on how each parameter affects the system's frequency response. A review of frequency domain control system design is then included to help the reader understand loop-shaping techniques. Finally, a compensator is designed for an open-air ANR headset using loop-shaping techniques and the robustness of the closed-loop performance is verified experimentally.Master of Science
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Shafer, Benjamin Michael. "Error sensor placement for active control of an axial cooling fan /." Diss., CLICK HERE for online access, 2007. http://contentdm.lib.byu.edu/ETD/image/etd2119.pdf.
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Sagers, Jason Derek. "Analog Feedback Control of an Active Sound Transmission Control Module." Diss., CLICK HERE for online access, 2008. http://contentdm.lib.byu.edu/ETD/image/etd2461.pdf.
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Forsgren, Fredrik. "Active Noise Control in Forest Machines." Thesis, Umeå universitet, Institutionen för fysik, 2011. http://urn.kb.se/resolve?urn=urn:nbn:se:umu:diva-48661.
Full textAbstract:
Achieving a low noise level is of great interest to the forest machine industry. Traditionally this is obtained by using passive noise reduction, i.e. by using materials for sound isolation and sound absorption. Especially designs to attenuate low frequency noise tend to be bulky and impractical from an installation point of view. An alternative solution to the problem is to use active noise control (ANC). The basic principle of ANC is to generate an anti-noise signal designed to destructively interfere with the unwanted noise. In this thesis two algorithms (Feedback FxLMS and Feedforward FxLMS) are implemented and evaluated for use in the ANC-system. The ANC-system is tuned to the specific environment in the driver’s cabin of a Komatsu forest machine. The algorithms are first tested in a simulated environment and then in real-time inside a forest machine. Simulations are made both in Matlab and in C using both generated signals and recorded signals. The C code is implemented on the Analog Devices Blackfin DSP card BF526. The result showed a significantly reduction of the sound pressure level (SPL) in the driver’s cabin. The noise attenuation obtained using the Feedback FxLMS was approximately 14 dB for a tonal 100 Hz signal and 11 dB using recorded engine noise from a forest machine at 850 rpm.
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Zylich, Brian Matthew. "Training Noise-Robust Spoken Phrase Detectors with Scarce and Private Data: An Application to Classroom Observation Videos." Digital WPI, 2019. https://digitalcommons.wpi.edu/etd-theses/1289.
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We explore how to automatically detect specific phrases in audio from noisy, multi-speaker videos using deep neural networks. Specifically, we focus on classroom observation videos that contain a few adult teachers and several small children (< 5 years old). At any point in these videos, multiple people may be talking, shouting, crying, or singing simultaneously. Our goal is to recognize polite speech phrases such as "Good job", "Thank you", "Please", and "You're welcome", as the occurrence of such speech is one of the behavioral markers used in classroom observation coding via the Classroom Assessment Scoring System (CLASS) protocol. Commercial speech recognition services such as Google Cloud Speech are impractical because of data privacy concerns. Therefore, we train and test our own custom models using a combination of publicly available classroom videos from YouTube, as well as a private dataset of real classroom observation videos collected by our colleagues at the University of Virginia. We also crowdsource an additional 1152 recordings of polite speech phrases to augment our training dataset. Our contributions are the following: (1) we design a crowdsourcing task for efficiently labeling speech events in classroom videos, (2) we develop a neural network-based architecture for speech recognition, robust to noise and overlapping speech, and (3) we explore methods to synthesize new and authentic audio data, both to increase the training set size and reduce the class imbalance. Finally, using our trained polite speech detector, (4) we investigate the relationship between polite speech and CLASS scores and enable teachers to visualize their use of polite language.
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Werner, Maike [Verfasser]. "Experimental Study on Tonal Self-Noise Generation by Aeroacoustic Feedback on a Side Mirror / Maike Werner." München : Verlag Dr. Hut, 2018. http://d-nb.info/1164293540/34.
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Munyai, Pandelani Reuben Mulalo. "On the improvement of phase noise in wideband frequency synthesizers." Diss., University of Pretoria, 2017. http://hdl.handle.net/2263/63003.
Full textAbstract:
Wireless communication systems are based on frequency synthesizers that generate carrier signals,
which are used to transmit information. Frequency synthesizers use voltage controlled oscillators
(VCO) to produce the required frequencies within a specified period of time. In the process of generating
frequency, the VCO and other electronic components such as amplifiers produce some unwanted
short-term frequency variations, which cause frequency instability within the frequency of
interest known as phase noise (PN). PN has a negative impact on the performance of the overall wireless
communication system. A literature study conducted on this research reveals that the existing PN
cancellation techniques have some limitations and drawbacks that require further attention.
A new PN correction technique based on the combination of least mean square (LMS) adaptive filtering
and single-loop single-bit Sigma Delta (SD) modulator is proposed. The new design is also based
on the Cascaded Resonator Feedback (CRFB) architecture. The noise transfer function (NTF) of the
architecture was formulated in way that made it possible to stabilize the frequency fluctuations within
the in-band (frequency of interest) by locating its poles and zeros within the unit circle.
The new design was simulated and tested on a commercially available software tool called Agilent Advanced Design System (ADS). Simulation results show that the new technique achieves better
results when compared with existing techniques as it achieves a 104 dB signal-to-noise (SNR), which
is an improvement of 9 dB when compared with the existing technique accessed from the latest
publications. The new design also achieves a clean signal with minimal spurious tones within the inband
with a phase noise level of -141 dBc/Hz (lower phase noise level by 28 dBc/Hz) when compared
with the existing techniques.Thesis (MEng)--University of Pretoria, 2017.
Electrical, Electronic and Computer Engineering
MEng
Unrestricted
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Nguyen, Lan K. "Dynamical modelling of feedback gene regulatory networks." Diss., Lincoln University, 2009. http://hdl.handle.net/10182/1340.
Full textAbstract:
Living cells are made up of networks of interacting genes, proteins and other bio-molecules. Simple interactions between network components in forms of feedback regulations can lead to complex collective dynamics. A key task in cell biology is to gain a thorough understanding of the dynamics of intracellular systems and processes. In this thesis, a combined approach of mathematical modelling, computational simulation and analytical techniques, has been used to obtain a deeper insight into the dynamical aspects of a variety of feedback systems commonly encountered in cells. These systems range from model system with detailed available molecular knowledge to general regulatory motifs with varying network structures. Deterministic as well as stochastic modelling techniques have been employed, depending primarily on the specific questions asked. The first part of the thesis focuses on dissecting the principles behind the regulatory design of the Tryptophan Operon system in Escherichia coli. It has evolved three negative feedback loops, namely repression, attenuation and enzyme inhibition, as core regulator mechanisms to control the intracellular level of tryptophan amino acid, which is taken up for protein synthesis. Despite extensive experimental knowledge, the roles of these seemingly redundant loops remain unclear from a dynamical point of view. We aim to understand why three loops, rather than one, have evolved. Using a large-scale perturbation/response analysis through modelling and simulations and novel metrics for transient dynamics quantification, it has been revealed that the multiple negative feedback loops employed by the tryptophan operon are not redundant. In fact, they have evolved to concertedly give rise to a much more efficient, adaptive and stable system, than any single mechanism would provide. Since even the full topology of feedback interactions within a network is insufficient to determine its behavioural dynamics, other factors underlying feedback loops must be characterised to better predict system dynamics. In the second part of the thesis, we aim to derive these factors and explore how they shape system dynamics. We develop an analytical approach for stability and bifurcation analysis and apply it to class of feedback systems commonly encountered in cells. Our analysis showed that the strength and the Hill coefficient of a feedback loop play key role in determining the dynamics of the system carrying the loop. Not only that, the position of the loop was also found to be crucial in this decision. The analytical method we developed also facilitates parameter sensitivity analysis in which we investigate how the production and degradation rates affect system dynamics. We find that these rates are quite different in the way they shape up system behaviour, with the degradation rates exhibiting a more intricate manner. We demonstrated that coupled-loop systems display greater complexity and a richer repertoire of behaviours in comparison with single-loop ones. Different combinations of the feedback strengths of individual loops give rise to different dynamical regimes. The final part of the thesis aims to understand the effects of molecular noise on dynamics of specific systems, in this case the Tryptophan Operon. We developed two stochastic models for the system and compared their predictions to those given by the deterministic model. By means of simulations, we have shown that noise can induce oscillatory behaviour. On the other hand, incorporating noise in an oscillatory system can alter the characteristics of oscillation by shifting the bifurcation point of certain parameters by a substantial amount. Measurement of fluctuations reveals that that noise at the transcript level is most significant while noise at the enzyme level is smallest. This study highlights that noise should not be neglected if we want to obtain a complete understanding of the dynamic behaviour of cells.
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Adiseno. "Design Aspects of Fully Integrated Multiband Multistandard Front-End Receivers." Doctoral thesis, KTH, Microelectronics and Information Technology, IMIT, 2003. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-3581.
Full textAbstract:
In this thesis, design aspects of fully integrated multibandmultistandard front-end receivers are investigated based onthree fundamental aspects: noise, linearity and operatingfrequency. System level studies were carried out to investigatethe effects of different modulation techniques, duplexing andmultiple access methods on the noise, linearity and selectivityperformance of the circuit. Based on these studies and thelow-cost consideration, zero-IF, low-IF and wideband-IFreceiver architectures are promising architectures. These havea common circuit topology in a direct connection between theLNA and the mixer, which has been explored in this work toimprove the overall RF-to-IF linearity. One front-end circuitapproach is used to achieve a low-cost solution, leading to anew multiband multistandard front-end receiver architecture.This architecture needs a circuit whose performance isadaptable due to different requirements specified in differentstandards, works across several RF-bands and uses a minimumamount ofexternal components.
Five new circuit topologies suitable for a front-endreceiver consisting of an LNA and mixer (low-noise converter orLNC) were developed. A dual-loop wide-band feedback techniquewas applied in all circuits investigated in this thesis. Threeof the circuits were implemented in 0.18 mm RF-CMOS and 25 GHzbipolar technologies. Measurement results of the circuitsconfirmed the correctness of the design approach.
The circuits were measured in several RF-bands, i.e. in the900 MHz, 1.8 GHz and 2.4 GHz bands, with S11 ranging from9.2 dB to17 dB. The circuits have a typicalperformance of 18-20 dB RF-to-IF gain, 3.5-4 dB DSB NF and upto +4.5 dBm IIP3. In addition, the circuit performance can beadjusted by varying the circuits first-stage biascurrent. The circuits may work at frequencies higher than 3GHz, as only 1.5 dB of attenuation is found at 3 GHz and nopeaking is noticed. In the CMOS circuit, the extrapolated gainat 5 GHz is about 15 dB which is consistent with the simulationresult. The die-area of each of the circuits is less than 1mm2.
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Zhou, Limin. "ASSESSING AND MITIGATING AIRBORNE NOISE FROM POWER GENERATION EQUIPMENT." UKnowledge, 2013. http://uknowledge.uky.edu/me_etds/22.
Full textAbstract:
This dissertation examines the assessment and mitigation of airborne noise from power generation equipment.
The first half of the dissertation investigates the diagnosis and treatment of combustion oscillations in boilers. Sound is produced by the flame and is reflected downstream from the combustion chamber. The reflected sound waves perturb the mixture flow or equivalence ratio increasing the heat release pulsations and the accompanying sound produced by the flame. A feedback loop model for determining the likelihood of and diagnosing combustion oscillations was reviewed, enhanced, and then validated. The current work applies the feedback loop stability model to two boilers, which exhibited combustion oscillations. Additionally, a feedback loop model was developed for equivalence ratio fluctuations and validated. For the first boiler, the combustion oscillation problem is primarily related to the geometry of the burner and the intake system. For the second boiler, the model indicated that the combustion oscillations were due to equivalence ratio fluctuations. Principles for both measuring and simulating the acoustic impedance are summarized. An approach for including the effect of structural-acoustic coupling was developed. Additionally, a method for determining the impedance above the plane wave cut-off frequency, using the acoustic FEM, of the boiler was proposed.
The second half of the dissertation examines the modeling of bar silencers. Bar silencers are used to mitigate the airborne noise from large power generation equipment (especially gas turbines). Due to the large dimensions of the full cross section, a small representative cell is isolated from the entire array for analysis purposes. To predict the acoustical performance of the isolated cell for different geometric configurations, a numerical method based on the direct mixed-body boundary element method (BEM) was used. An analytical solution for a simplified circular geometry was also derived to serve as a comparison tool for the BEM. Additionally, a parametric study focusing on the effects of flow resistivity, perforate porosity, length of bars, and cross-sectional area ratio was performed. A new approach was proposed to evaluate the transmission loss based on a reciprocal work identity. Moreover, extension of the transmission loss computation above the plane wave cut-off frequency was demonstrated.
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Siviero, Diego Azevedo. "Aplicação das metodologias feedback e feedforward no controle ativo do ruido transmitido por uma placa." [s.n.], 2007. http://repositorio.unicamp.br/jspui/handle/REPOSIP/263247.
Full textAbstract:
Orientadores: Jose Roberto de França Arruda, Juan Francisco Camino dos SantosDissertação (mestrado) - Universidade Estadual de Campinas, Faculdade de Engenharia Mecanica
Made available in DSpace on 2018-08-09T13:00:18Z (GMT). No. of bitstreams: 1 Siviero_DiegoAzevedo_M.pdf: 1349512 bytes, checksum: dd6f2edf66df39c704de24d42a9ed55c (MD5) Previous issue date: 2007
Resumo: Dado o contínuo processo de otimização ao no setor aeroespacial é cada vez mais priorizada nos projetos a busca pelo decréscimo de massa dos sistemas secundários ao vôo, como por exemplo, no sistema responsável pelo controle de ruídos internos. Isto tem provocado um aumento de interesse no desenvolvimento de placas inteligentes, ou ¿smart-plates¿, que consistem em elementos estruturais com atuadores e sensores agregados as suas superfícies, para o controle de suas próprias vibrações estruturais, possibilitando, no futuro, a redução nas dimensões dos elementos passivos de contenção de ruído hoje em uso, principalmente no trabalho com sinais de baixa freqüência. Cresce paralelamente a necessidade de se do definir qual a melhor estratégia de controle para estas estruturas inteligentes. Este estudo descreve a implementação de dois tipos distintos de controladores em uma placa de LEXAN com o objetivo de aumentar a perda transmissão de ruídos. O primeiro controlador utilizado é tipo H2, uma estratégia de controle que utiliza a realimentação da saída (feedback) como referencia para a ação de controle. O segundo controlador é o Filtered-X LMS, uma estratégia por alimentação direta (feedforward) que utiliza um sinal correlacionado ao distúrbio como referencia para o controle. A resposta da planta em malha fechada, com cada controlador, é medida por um microfone com a finalidade de s88e determinar o desempenho atingido pelas diferentes metodologias. Um enfoque maior será dado ao controlador Filtered-X LMS, que também será detalhado e aplicado a um sistema numérico de dutos
Abstract: Due to the continuous optimization process in the aerospace industry, the search for lighter secondary flight systems has been intensively investigated in recent years, for instance, the system responsible for the control of internal noise. This leads to a growing interest in the development of smart panels, which consist of structural elements with actuators and sensors attached to their surfaces, in order to control the structural vibration. This leads to a reduction of the members of passive elements used to attenuate noise mainly at low frequencies. The interest in these smart structures grows along with the necessity of defining the best control strategy. This thesis describes the implementation of two distinct controllers on a LEXAN smart plate, with the purpose of increasing the transmission loss. The first is an H2 dynamic output feedback controller, a strategy that uses the system's output as a reference to the control action. The second controller is the Filtered-X LMS, a strategy that uses a signal correlated with the disturbance as a reference to the control. The response of the closed-loop systems, using each controller, is measured using a microphone. This determines the performance achieved by the different methodologies. More emphasis will be given to the Filtered-X LMS controller, which is also applied to a vibroacoustic problem in a duct
Mestrado
Mecanica dos Sólidos e Projeto Mecanico
Mestre em Engenharia Mecânica
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Deng, Jie. "Rear Axle Gear Whine Noise Abatement via Active Vibration Control of the Rear Subframe." University of Dayton / OhioLINK, 2015. http://rave.ohiolink.edu/etdc/view?acc_num=dayton1447772359.
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Tokatli, Ahmet. "Design And Implementation Of A Dsp Based Active Noise Controler For Headsets." Master's thesis, METU, 2004. http://etd.lib.metu.edu.tr/upload/12605429/index.pdf.
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The design of a battery-powered, portable headphone active noise control
system with TI TMS320C5416 DSP is described. The preliminary implementation
of the system on a C5416 DSK is also explained. The problems of fixed-point
implementation are described and solutions are proposed. Sign-sign Fx-LMS
algorithm with a dead-zone is introduced and used as the adaptation algorithm.
Effective use of dynamic range to improve the accuracy in filtering operations is
discussed. Details of the designed battery-powered DSP board are given and board
software development process is explained. The DSK system and designed
portable system is compared against two commercially available analog systems
under three different types of noisescomposition of tones, drill noise and propeller plane cabin noise. The results reveal that adaptive system has better overall performance.
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Crawford, Jackie H. III. "Factors that limit control effectiveness in self-excited noise driven combustors." Diss., Georgia Institute of Technology, 2012. http://hdl.handle.net/1853/43647.
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A full Strouhal number thermo-acoustic model is purposed for the feedback control of self excited noise driven combustors. The inclusion of time delays in the volumetric heat release perturbation models create unique behavioral characteristics which are not properly reproduced within current low Strouhal number thermo acoustic models. New analysis tools using probability density functions are introduced which enable exact expressions for the statistics of a time delayed system. Additionally, preexisting tools from applied mathematics and control theory for spectral analysis of time delay systems are introduced to the combustion community. These new analysis tools can be used to extend sensitivity function analysis used in control theory to explain limits to control effectiveness in self-excited combustors. The control effectiveness of self-excited combustors with actuator constraints are found to be most sensitive to the location of non-minimum phase zeros. Modeling the non-minimum phase zeros correctly require accurate volumetric heat release perturbation models. Designs that removes non-minimum phase zeros are more likely to have poles in the right hand complex plane. As a result, unstable combustors are inherently more responsive to feedback control.
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Frank, Hannes [Verfasser]. "High Order Large Eddy Simulation for the Analysis of Tonal Noise Generation via Aeroacoustic Feedback Effects at a Side Mirror / Hannes Frank." München : Verlag Dr. Hut, 2017. http://d-nb.info/1147674434/34.
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Wickert, Mark, Shaheen Samad, and Bryan Butler. "AN ADAPTIVE BASEBAND EQUALIZER FOR HIGH DATA RATE BANDLIMITED CHANNELS." International Foundation for Telemetering, 2006. http://hdl.handle.net/10150/604050.
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ITC/USA 2006 Conference Proceedings / The Forty-Second Annual International Telemetering Conference and Technical Exhibition / October 23-26, 2006 / Town and Country Resort & Convention Center, San Diego, CaliforniaMany satellite payloads require wide-band channels for transmission of large amounts of data to users on the ground. These channels typically have substantial distortions, including bandlimiting distortions and high power amplifier (HPA) nonlinearities that cause substantial degradation of bit error rate performance compared to additive white Gaussian noise (AWGN) scenarios. An adaptive equalization algorithm has been selected as the solution to improving bit error rate performance in the presence of these channel distortions. This paper describes the design and implementation of an adaptive baseband equalizer (ABBE) utilizing the latest FPGA technology. Implementation of the design was arrived at by first constructing a high fidelity channel simulation model, which incorporates worst-case signal impairments over the entire data link. All of the modem digital signal processing functions, including multirate carrier and symbol synchronization, are modeled, in addition to the adaptive complex baseband equalizer. Different feedback and feed-forward tap combinations are considered as part of the design optimization.
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Kromer, Justus Alfred. "Noise in adaptive excitable systems and small neural networks." Doctoral thesis, Humboldt-Universität zu Berlin, Mathematisch-Naturwissenschaftliche Fakultät, 2017. http://dx.doi.org/10.18452/17683.
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Neuronen sind erregbare Systeme. Ihre Antwort auf Anregungen oberhalb eines bestimmten Schwellwertes sind Pulse. Häufig wird die Pulserzeugung von verschiedenen Rückkopplungsmechanismen beeinflusst, die auf langsamen Zeitskalen agieren. Das kann zu Phänomenen wie Feuerraten-Adaptation, umgekehrter Feuerraten-Adaptation oder zum Feuern von Pulsen in Salven führen. Weiterhin sind Neuronen verschiedenen Rauschquellen ausgesetzt und wechselwirken mit anderen Neuronen, in neuronalen Netzen. Doch wie beeinflusst das Zusammenspiel von Rückkopplungsmechanismen, Rauschen und der Wechselwirkung mit anderen Neuronen die Pulserzeugung? Diese Arbeit untersucht, wie die Pulserzeugung in rauschgetriebenen erregbaren Systemen von langsamen Rückkopplungsmechanismen und der Wechselwirkung mit anderen erregbaren Systemen beeinflusst wird. Dabei wird die Pulserzeugung in drei Szenarien betrachtet: (i) in einem einzelnen erregbaren System, das um einen langsamen Rückkopplungsmechanismus erweitert wurde, (ii) in gekoppelten erregbaren Systemen und (iii) in stark gekoppelten salvenfeuernden Neuronen. In jedem dieser Szenarien wird die Pulsstatistik mit Hilfe von analytischen Methoden und Computersimulationen untersucht. Das wichtigste Resultat im ersten Szenario ist, dass das Zusammenspiel von einer stark anregenden Rückkopplung und Rauschen zu rauschkontrollierter Bistabilität führt. Das erlaubt es dem System zwischen verschiedenen Modi der Pulserzeugung zu wechseln. In (ii) wird die Pulserzeugung stark von der Wahl der Kopplungsstärken und der Anzahl der Verbindungen beeinflusst. Analytische Näherungen werden abgeleitet, die einen Zusammenhang zwischen der Anzahl der Verbindungen und der Pulsrate, sowie der Pulszugvariabilität herstellen. In (iii) wird festgestellt, dass eine hemmende Rückkopplung zu sehr unregelmäßigem Verhalten der isolierten Neuronen führt, wohingegen eine starke Kopplung mit dem Netzwerk ein regelmäßigeres Feuern von Salven hervorruft.Neurons are excitable systems. Their responses to excitations above a certain threshold are spikes. Usually, spike generation is shaped by several feedback mechanisms that can act on slow time scales. These can lead to phenomena such as spike-frequency adaptation, reverse spike-frequency adaptation, or bursting. In addition to these, neurons are subject to several sources of noise and interact with other neurons, in the connected complexity of a neural network. Yet how does the interplay of feedback mechanisms, noise as well as interaction with other neurons affect spike generation? This thesis examines how spike generation in noise-driven excitable systems is influenced by slow feedback processes and coupling to other excitable systems. To this end, spike generation in three setups is considered: (i) in a single excitable system, which is complemented by a slow feedback mechanism, (ii) in a set of coupled excitable systems, and (iii) in a set of strongly-coupled bursting neurons. In each of these setups, the statistics of spiking is investigated by a combination of analytical methods and computer simulations. The main result of the first setup is that the interplay of strong positive (excitatory) feedback and noise leads to noise-controlled bistability. It enables excitable systems to switch between different modes of spike generation. In (ii), spike generation is strongly affected by the choice of the coupling strengths and the number of connections. Analytical approximations are derived that relate the number of connections to the firing rate and the spike train variability. In (iii), it is found that negative (inhibitory) feedback causes very irregular behavior of the isolated bursters, while strong coupling to the network regularizes the bursting.
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Johansson, Sven. "Active Control of Propeller-Induced Noise in Aircraft : Algorithms & Methods." Doctoral thesis, Karlskrona, Ronneby : Blekinge Institute of Technology, 2000. http://urn.kb.se/resolve?urn=urn:nbn:se:bth-00171.
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In the last decade acoustic noise has become more and more regarded as a problem. In cars, boats, trains and aircraft, low-frequency noise reduces comfort. Lightweight materials and more powerful engines are used in high-speed vehicles, resulting in a general increase in interior noise levels. Low-frequency noise is annoying and during periods of long exposure it causes fatigue and discomfort. The masking effect which low-frequency noise has on speech reduces speech intelligibility. Low-frequency noise is sought to be attenuated in a wide range of applications in order to improve comfort and speech intelligibility. The use of conventional passive methods to attenuate low-frequency noise is often impractical since considerable bulk and weight are required; in transportation large weight is associated with high fuel consumption. In order to overcome the problems of ineffective passive suppression of low-frequency noise, the technique of active noise control has become of considerable interest. The fundamental principle of active noise control is based on secondary sources producing ``anti-noise.'' Destructive interference between the generated and the primary sound fields results in noise attenuation. Active noise control systems significantly increase the capacity for attenuating low-frequency noise without major increase in volume and weight. This doctoral dissertation deals with the topic of active noise control within the passenger cabin in aircraft, and within headsets. The work focuses on methods, controller structures and adaptive algorithms for attenuating tonal low-frequency noise produced by synchronized or moderately synchronized propellers generating beating sound fields. The control algorithm is a central part of an active noise control system. A multiple-reference feedforward controller based on the novel actuator-individual normalized Filtered-X Least-Mean-Squares algorithm is introduced, yielding significant attenuation of such period noise. This algorithm is of the LMS-type, and owing to the novel normalization it can also be regarded as a Newton-type algorithm. The new algorithm combines low computational complexity with high performance. For that reason the algorithm is suitable for use in systems with a large number of control sources and control sensors in order to reduce the computional power required by the control system. The computational power of the DSP hardware is limited, and therefore algorithms with high computational complexity allow fewer control sources and sensors to be used, often with reduced noise attenuation as a result. In applications, such as controlling aircraft cabin noise, where a large multiple-channel system is needed to control the relative complex interior sound field, it is of great importance to keep down the computational complexity of the algorithm so that a large number of loudspeakers and microphones can be used. The dissertation presents theoretical work, off-line computer experiments and practical real-time experiments using the actuator-individual normalized algorithm. The computer experiments are principally based on real-life cabin noise data recorded during flight in a twin-engine propeller aircraft and in a helicopter. The practical experiments were carried out in a full-scale fuselage section from a propeller aircraft.Buller i vår dagliga miljö kan ha en negativ inverkan på vår hälsa. I många sammanhang, i tex bilar, båtar och flygplan, förekommer lågfrekvent buller. Lågfrekvent buller är oftast inte skadligt för hörseln, men kan vara tröttande och försvåra konversationen mellan personer som vistas i en utsatt miljö. En dämpning av bullernivån medför en förbättrad taluppfattbarhet samt en komfortökning. Att dämpa lågfrekvent buller med traditionella passiva metoder, tex absorbenter och reflektorer, är oftast ineffektivt. Det krävs stora, skrymmande absorbenter för att dämpa denna typ av buller samt tunga skiljeväggar för att förhindra att bullret transmitteras vidare från ett utrymme till ett annat. Metoder som är mera lämpade vid dämpning av lågfrekvent buller är de aktiva. De aktiva metoderna baseras på att en vågrörelse som ligger i motfas med en annan överlagras och de släcker ut varandra. Bullerdämpningen erhålls genom att ett ljudfält genereras som är lika starkt som bullret men i motfas med detta. De aktiva bullerdämpningsmetoderna medför en effektiv dämpning av lågfrekvent buller samtidigt som volymen, tex hos bilkupen eller båt/flygplanskabinen ej påverkas nämnvärt. Dessutom kan fordonets/farkostens vikt reduceras vilket är tacksamt för bränsleförbrukningen. I de flesta tillämpningar varierar bullrets karaktär, dvs styrka och frekvensinnehåll. För att följa dessa variationer krävs ett adaptivt (självinställande) reglersystem som styr genereringen av motljudet. I propellerflygplan är de dominerande frekvenserna i kabinbullret relaterat till propellrarnas varvtal, man känner alltså till frekvenserna som skall dämpas. Man utnyttjar en varvtalssignal för att generera signaler, så kallade referenssignaler, med de frekvenser som skall dämpas. Dessa bearbetas av ett reglersystem som generar signaler till högtalarna som i sin tur generar motljudet. För att ställa in högtalarsignalerna så att en effektiv dämpning erhålls, används mikrofoner utplacerade i kabinen som mäter bullret. För att åstadkomma en effektiv bullerdämpning i ett rum, tex i en flygplanskabin, behövs flera högtalare och mikrofoner, vilket kräver ett avancerat reglersystem. I doktorsavhandlingen ''Active Control of Propeller-Induced Noise in Aircraft'' behandlas olika metoder för att reducera kabinbuller härrörande från propellrarna. Här presenteras olika strukturer på reglersystem samt beräkningsalgoritmer för att ställa in systemet. För stora system där många högtalare och mikrofoner används, samt flera frekvenser skall dämpas, är det viktigt att systemet inte behöver för stor beräkningskapacitet för att generera motljudet. Metoderna som behandlas ger en effektiv dämpning till låg beräkningskostnad. Delar av materialet som presenteras i avhandlingen har ingått i ett EU-projekt med inriktning mot bullerundertryckning i propellerflygplan. I projektet har flera europeiska flygplanstillverkare deltagit. Avhandlingen behandlar även aktiv bullerdämpning i headset, som används av helikopterpiloter. I denna tillämpning har aktiv bullerdämpning används för att öka taluppfattbarheten.
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Cheng, Yong-Sheng, and 鄭詠聲. "Application of Feedback Active Noise Control on the Car Interior Noise." Thesis, 2014. http://ndltd.ncl.edu.tw/handle/k975uq.
Full textAbstract:
碩士國立臺北科技大學
車輛工程系所
102
The interior noise of car is usually reduced by passive noise reduction method, which uses sound-absorbing or sound-isolation materials to reducenoise.However, this technique is ineffective on low-frequency noise.Therefore as a result emerges the active noise control method (ANC), which is successful to reduce low-frequency noise.Thisresearchstudies the application of active noise control theory to reduce car cabin noise. The implementation of ANC uses the digital signal processor TMS320C6713DSK. This study uses feedback control system and the secondary-path is modeled by FIR (Finite Impulse Filter) filter. The influencesof filter length in theFXLMS,Leaky FXLMS and FXNLMS algorithm are discussed. In order to compare the noise reduction and stability of FXLMS,LeakyFXLMSandFXNLMS algorithm, experimentsare conducted in single-tone noise,dual-tone noise and frequency-band noise conditions, bychangingnoisesound level and experimental time.
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Lee, Bo-Long, and 李柏榮. "Study of Low Noise Amplifier with Source Inductance Feedback." Thesis, 1998. http://ndltd.ncl.edu.tw/handle/30587058219729267197.
Full textAbstract:
碩士國立交通大學
電信工程研究所
86
This thesis discusses the technique of source inductive feedback (SIF) in low noise amplifier design to achieve both input and noise matching. The behaviors of device with and without SIF are investigated by noisy two-port analysis. In the mean time, a simpli-fied noise model is proposed to verify the possibility for input/noise match. The variations of and with SIF are also examined. A full small signal model containing noise sources is presented to figure out the deteriorate factors for noise figure. The limitation for using SIF is also indicated. Finally, a C-band amplifier is designed to confirm our investigation.
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Kuo, Po-Wei, and 郭柏偉. "Design of Feedback Low Noise Amplifiers and Distributed Amplifiers." Thesis, 2000. http://ndltd.ncl.edu.tw/handle/05667231046987366815.
Full textAbstract:
碩士國立臺灣大學
電信工程學研究所
88
The low noise amplifiers (LNA’s) are essential components in communication system front end in wireless local area networks (LAN’s), satellite links, and radiometric sensors. It dominates the noise figure and input voltage standing-wave ratio (VSWR) of the overall system because the first block signal fed from the antenna is the LNA. On the other hand, ultra-wideband amplifiers are widely used as baseband amplifiers in high-speed lightwave systems and gain blocks in microwave/millimeter-wave communication and sensor systems or in wideband instruments. The distributed amplifier (DA) approach provides the largest gain-bandwidth products, together with a lower input and output reflection for broad band applications. This thesis includes the design methodology and implementation of both microwave and millimeter wave frequency LNAs and DAs. The first part is focused in the noise behavior of two-port network using the equivalent noise resistance Rn (or equivalent noise conductance gn). The gn circle is developed to analyze the parallel-feedback amplifier. The design and performance of two K-band and V-band MMIC single-ended low noise amplifiers are also presented. The associated device modeling is also discussed. Pseudomorphic high electron mobility transistor (PHEMT) small signal equivalent circuit parameters and noise model parameters are obtained from curve fitting of measured parameters. The two-stage K-band LNA has achieved 2.5-dB noise figure with associated gain of 17.7 dB at 24 GHz. The two-stage V-band LNA has achieved a small signal gain of 8.7 dB at 50 GHz. The second part includes the design of broadband amplifiers using distributed amplifier approach. Three MMIC DAs are presented. A 1-10 GHz HBT-based DA has a simulated gain of 10 dB. The capacitive-division HEMT-based DAs achieved a simulated 8-dB gain with 60-GHz bandwidth and 6-dB gain with 80-GHz bandwidth, respectively for microstrip-line and grounded-CPW design. Due to the turn-around time, the DA chips will be available later.
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Daud, Muhyin, and 李明言. "Phase-Noise Analysis and Improvement of Microwave Feedback Oscillators." Thesis, 2013. http://ndltd.ncl.edu.tw/handle/46940951885937007638.
Full textAbstract:
碩士國立臺灣科技大學
電子工程系
101
In this thesis, two types of frequency selective elements, the harmonic suppressed and high complex quality-factor elements, are respectively employed to design and implement the low phase-noise, namely low jitter, oscillators. It is experimentally proved that both proposed design approaches effectively contributes to the oscillator performance. The phase-noise improvement due to high Qsc can be up to 20 dB at 1MHz offset frequency. Meanwhile, the contribution of the phase-noise improvement due to harmonic suppression can be only up to 10 dB. In addition, the mathematical analysis, impulse sensitivity function (ISF), is applied to reveal these improvements. Since the c1 coefficient of the ISF represents noise around the oscillation frequency, it can be reduced by employing high Qsc frequency selective element. In addition, the c2 and c3 coefficients of the ISF represent the noise around the 2nd and 3rd harmonic frequencies, respectively. One can adopt the harmonic suppressed element to lower these two coefficients. The ISF RMS value of the proposed oscillator using the stepped impedance resonator (SIR)-based trisection filter is lower than the one using the conventional elements with lower Qsc. The adopted SIR-based trisection filter has a high Qsc value, 131.04, and about 30 dB 2nd and 3rd harmonic suppressions. The proposed oscillator has a phase-noise of -148.0 dBc/Hz at 1MHz offset frequency with a figure-of-merit (FOM) of -203.0 dBc/Hz. In addition, a very low phase-noise oscillator can be achieved using the magnetic-coupling trisection filter. Due to its high Qsc value, 150, the proposed configuration has phase-noise of -150.8 dBc/Hz at 1MHz offset frequency with the figure-of-merit (FOM) of -205.2 dBc/Hz.
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Fan, Chiang-Yi, and 范姜毅. "Noise Reduction and feedback cancellation algorithm for hearing aids." Thesis, 2014. http://ndltd.ncl.edu.tw/handle/22852971460435615358.
Full textAbstract:
博士國立交通大學
電子工程學系 電子研究所
103
This dissertation proposes algorithms for digital hearing aids (HAs). The algorithms include voice activity detection (VAD), noise reduction (NR), feedback cancellation (FC), and dynamic range compression (DRC). VAD is used to indicate the speech periods and noise periods to assist NR, DRC, and FC processing. NR is used to reduce the noise for the speech intelligence and listening comfort. FC is used to cancel the feedback sound for the stability in high-gain HAs. DRC is adopted to match the residual range of the hearing loss patients. These functions are very important for digital HAs. Thus, this dissertation aims to propose high performance algorithms with low complexity for these functions. This dissertation presents a low computational complexity hardware-oriented neuromorphic pitch based noise reduction algorithm for monosyllable HA applications. The proposed NR design consists of a pitch-based voice activity detection for speech detection and a neuromorphic noise reduction for speech enhancement. The pitch-based VAD is developed on ANSI S1.11 based filter bank architecture and employs the characteristics of monosyllable and nonlinear energy operator to improve the VAD accuracy. The neuromorphic noise reduction reduces the background noise by using the characteristics of the human hearing system and the clues of speech. Simulations show the proposed algorithm can provide about 80\% VAD accuracy and 4dB signal-to-noise ratio (SNR) improvement at 0dB SNR, which can satisfy the requirement of the mild hearing loss patients. To meet the requirement of the moderate or moderately severe hearing loss patients and to eliminate the contradiction between NR and DRC. An onset based noise reduction (ONR) with two dynamic range compression (T-DRC) is proposed for HA systems. The ONR is proposed to achieve higher SNR and perception evaluation of speech quality (PESQ), compared with the neuromorphic pitch based noise reduction algorithm. The ONR is implemented with a 10ms quasi-ANSI S1.11 1/3 octave based filter bank. The ONR uses two noise reduction gain curves for different levels of speech energy and six gain levels for different onset energy levels. To further improve the ONR performance, thresholds of gain levels of middle frequency subbands are refined to have a smoother threshold between low frequency subbands and high frequency subbands. When a series concatenation of ONR and dynamic range compression is used, the SNR and PESQ enhancement obtained from the ONR can be degraded. Thus, the T-DRC uses one DRC with normal compression for speech periods and another DRC with higher compression for noise periods based on the applied gain calculated from the ONR. Compared to commonly used methods for HAs, the ONR can achieve higher SNR and comparable PESQ by using only 60\% to 65\% multiplication operations. Also, simulation results show that the ONR with T-DRC can achieve better SNR and PESQ enhancement compared to the ONR without T-DRC. For binaural HAs applications, the dissertation proposes a fusion of two VAD algorithms, namely the pitch-based VAD (PBVAD) and the binaural cross-correlation based VAD (BCRVAD) with the aim of increasing the overall VAD accuracy obtained with different noise types. The proposed algorithm has low complexity and thus is suitable for practical binaural HAs applications. Furthermore, monosyllable speech applications are targeted and so specific speech characteristics can be exploited. The pitch-based VAD algorithm can achieve high accuracy in white and car noise by incorporating known properties of the human hearing system and monosyllable speech characteristics. On the other hand, the computationally more expensive binaural cross-correlation based VAD can achieve excellent accuracy in babble and factory noise by exploiting the spatial cues and monosyllable speech characteristics. With the aim of achieving high accuracy in different noise types and low computational complexity, a babble noise detector is introduced to activate the binaural cross-correlation based VAD algorithm only during babble noise and factory noise periods. The resulting fusion VAD algorithm achieves about 90\% VAD accuracy for white and car noise and about 81\% for babble and factory noise. It also shows that noise reduction performance for SNR improvement is in general proportional to VAD performance, so the fusion VAD can lead to higher SNR improvement. Comparisons with previous methods for binaural HAs are carried out to show that the proposed algorithm achieves superior VAD accuracy in all noise types. For feedback cancellation algorithm, this dissertation proposes a novel algorithm and architecture for the adaptive feedback cancellation (AFC) based on the pitch and the formant information for HA applications. The proposed method, named as Pitch based Formant Estimation (PFE-AFC), has significantly low complexity compared to Prediction Error Method AFC (PEM-AFC). The proposed PFE-AFC consists of a forward and a backward path processing. The forward path processing includes a low complexity pitch based formant estimator for decorrelation filter coefficients update and a pitch-based VAD for speech detection, which facilitates the feedback cancellation filter in the backward path to reduce feedback component and maintain speech quality. From system point of view, the PFE-AFC has low complexity overhead since it is easy to share computation resource with other components in the HA system, such as VAD and NR. In addition, the PFE-AFC is suitable for hardware implementation owing to its regular structure. Complexity evaluations show that the PFE-AFC has four orders lower complexity than the PEM-AFC. Simulation results show that the PFE-AFC and the PEM-AFC can achieve similar PESQ and added stable gain. Moreover, the proposed PFE-AFC can outperform the conventional AFC in PESQ and added stable gain.
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Chou, Mu-Xuan, and 周睦軒. "Adapt feedback noise cancellation to hearing aid by NLMS algorithm." Thesis, 2009. http://ndltd.ncl.edu.tw/handle/83f62t.
Full textAbstract:
碩士國立陽明大學
醫學工程研究所
97
Feedback is one of the most frequent complaints of hearing aid users. Feedback and the occlusion effect pose great challenges for hearing aid design and usage. However, conventional solutions for those two problems often face the dilemma. In this research will discuss some advanced signal processing strategies of the feedback cancellation and the occlusion effect reduction. The strategies currently used to reduce feedback (i.e.,adaptive feedback reduction algorithms using adaptive gain reduction, notch filtering, and phase cancellation strategies). The adaptive gain reduction will limit the allowable maximum gain, and notch filtering may cause some signal loss, there are different defects will occurred. But the phase cancellation strategy not only can limit the allowable maximum gain, but also keep the signal’s completeness. This research main purpose is to use the phase cancellation strategy by NLMS(Normalized Least Mean Squares) algorithm to cancel the feedback noise by digital processor chip (GA-3280) in hearing aid. According to the experiment in our design when feedback cancellation starting. In standard sound booth would increase 10 dB SPL of allowable maximum gain and can keep the signal’s completeness.
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Lin, Jiun-Hung, and 林俊宏. "Effects of Feedback Adaptive Active Noise Cancellation on Speech Intelligibility." Thesis, 2006. http://ndltd.ncl.edu.tw/handle/26792052667526881863.
Full textAbstract:
博士國立陽明大學
醫學工程研究所
94
In the noise environment of the modern society, everybody may be influenced by noise. Noise except that will exert an influence to physiology and psychology, and impact on hearing whether a lot of people are puzzle deeply even more. Especially as for working under the louder noise environments, it influences greater and more far-reaching. Many researches have already proposed that workers may lose its function of sense of hearing in his working environment .Therefore, there have lots of workers must wearing the hearing protector to avoid the injury to its sense of hearing of various kinds of industrial noises in its environment. Generally, the hearing protector use two techniques: passive and active method. Passive cancellation methods include insulation and absorptive treatments to isolate the impact on ears of people of the noise. Although such treatments may be a solution for hearing protection, but these techniques work best at middle and high frequencies. The active noise protector could be used on the noise of low frequency. The active method employs the superposition principle cancels the unwanted noise signal by an antinoise signal with equal amplitude and contrary phase. To the person who wears the hearing aids, the noise not only influence the sense of hearing to distinguish, meanwhile, because the high ambient noise would be amplified to an even higher level by hearing aids function, and then cause more serious injury to ears. They may wear the hearing protector in order to prevent the noise affections, but this may result in the wearer failing to hear speech messages or warning sounds. So, it is an extremely urgent and essential demand for the worker or hearing impairer exposure on a high-intensity industrial noise how the effective environmental noise can be canceled without influence other meaningful sounds. Although many papers have already developed the noise cancellation technology to the hearing aids at present, but relatively fall short the method to solve industry noise. In this study, we consider the hearing impair person would face the possible industrial noise in his working environment, and developed a headset equipped with a feedback adaptive active noise cancellation (FbAANC) methods, then reduce the transformer, fans, power-station, compressor industrial wide-band noise effectively. Result has shown that the proposed FbAANC headset can achieved the noise cancellation of 40dB to 60dB, when the noise with the frequency in 63Hz to 1250Hz and was also effective against wideband industrial noise with a maximal dominant noise spectrum power reduction of 37.9 dB, 23.8, 8.2, and 9.5 dB separately for four testing industrial noise. The study also evaluates the effects of FbAANC headset on speech intelligibility on a disyllabic Mandarin word discrimination test (WDT) platform. We expect someone under the noisy environment in SNR equal 0, –5, –10, -15, -20 and –25 dB, the noise cancellation technique could either improve the speech discrimination effectively. Finally, the SNR below -10dB, the mean WDT score with FbAANC headset was superior to without FbAANC headset 13 to 32% through 30 subjects of normal hearing threshold for evaluation. These results suggest that the FbAANC headset would be useful for hearing protection in workplaces with high levels industrial noise.
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Lin, Shi-Wei, and 林士為. "Ultra-Wideband Cascode Low Noise Amplifier with Resistive-Feedback technique." Thesis, 2010. http://ndltd.ncl.edu.tw/handle/45909805532141329419.
Full textAbstract:
碩士雲林科技大學
電子與資訊工程研究所
98
This thesis has a novel feedback topology and load circuit that allows the designer to obtain high voltage gain and matching in the wide frequency range from 3.1 GHz to 10.6 GHz on low noise amplifier. The chip was fabricated using TSMC commercial 0.18-μm 1P6M CMOS technology. The efficiency of the circuit was demonstrated by measurement. In first chip, An ultra-wideband (UWB) low noise amplifier (LNA) utilizes an High Pass filter as input matching network and shunt resistive-feedback technique is proposed. The second stage adopts two cascode Stage topology. The implemented LNA achieves a maximum power gain of 13 dB, The input return loss and output return loss are S11 of -6 dB and S22 of -10.5 dB. The measured IIP3 is about -4 dBm, and the noise figure (NF) of 5.5-7.1 dB was obtained in the frequency band of 3.1-10.6 GHz. It consumes a power dissipation of 20mW under a 1.5V power supply. In second chip, we utilize the High Pass filter to design a UWB-wideband low noise amplifier. The minimum noise figure is 4.3 dB and maximum gain is 11.5dB from 3 to 10.6GHz while drawing 17.6 mW from a 1.8V supply voltage. The input and output return loss are S11 of -5 dB and S22 of -10 dB., isolation better than -27dB, respectively. The input third-order intercept point IIP3 is -8dBm.
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Kitching, John E. "Quantum noise reduction in semiconductor lasers using dispersive optical feedback." Thesis, 1995. https://thesis.library.caltech.edu/4115/1/Kitching_je_1995.pdf.
Full textAbstract:
This thesis describes the phase and amplitude noise properties of semiconductor lasers subjected to weak, dispersive optical feedback. In the first half, experiments demonstrating reductions in the laser frequency noise power spectrum and spectral linewidth by several orders of magnitude are presented. Weak optical feedback is applied to the laser by an external cavity containing an atomic vapor. The presence of the vapor adds to the dispersion of the cavity and simultaneously locks the laser to a fixed frequency reference. The role of 1/f frequency noise in limiting the effectiveness of this linewidth reduction technique is investigated and 1/f noise is found to be the dominant contribution to the linewidth under strong optical feedback conditions.
An electronic feedback scheme utilizing FM sideband locking is then implemented alongside the optical feedback, and an additional reduction in the low frequency 1/f frequency noise power spectrum by over two orders of magnitude is obtained. With both systems operating simultaneously, the spectral linewidth is narrowed from its free-running value of about 20 MHz to 1.4 kHz. Excellent absolute frequency stability is also achieved.
In the second half, the effects of optical feedback on the quantum mechanical amplitude noise properties of the laser are examined. A fully quantum mechanical theory of amplitude and phase noise for a semiconductor laser with weak optical feedback is developed, and the nature and limits of the noise reduction using this technique are established. Particular attention is given to the feedback-induced enhancement of the amplitude squeezing which can be obtained in a pump-suppressed semiconductor laser: an improvement in the squeezing by 3 dB is predicted under moderate pumping. Somewhat larger noise reductions are expected when the laser is operating closer to threshold. Measurements performed on a laser biased close to threshold are then described and a reduction in the low frequency amplitude noise power spectrum by 7 dB is obtained.
An experimental investigation of the effects of optical feedback on the amplitude squeezing in a semiconductor laser is then discussed. The low frequency squeezing in a room temperature device is increased from 3% below the standard quantum limit (SQL) under free-running conditions to 19% below the SQL with optimal feedback. The experimental results are found to agree poorly with the single-mode model and a multi-mode model including the effects of asymmetrical cross-mode non-linear gain is developed to explain the discrepancy. Finally, further experimental investigation into the generation of amplitude squeezed light is presented using a commercial laser with no intentional external modifications. Squeezing as large as 29% below the SQL is measured using a balanced homodyne detector with the laser operating near room temperature, corresponding to 41% below the SQL at the output facet of the laser.
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Rojas, Norman Alejandro Jose. "Feedback control over signal to noise ratio constrained communication channels." Thesis, 2006. http://hdl.handle.net/1959.13/24913.
Full textAbstract:
The present thesis addresses the problem of stabilisability of a linear time invariant (LTI) output feedback control loop in the presence of a communication link. The communication link itself can be either located between the controller and the plant or between the plant and the controller. The communication link is assumed to be an additive coloured Gaussian noise channel with (or without) bandwidth limitation (memory) in the continuous-(or discrete-)time domain. The requirement for stabilisability of the feedback loop is then characterised as a lower bound on the channel signal to noise ratio (SNR). This lower bound is tight and it will depend on the channel model, plant and channel model NMP zeros, plant time delay and plant unstable poles. Performance requirements are also investigated, by loop shaping in the continuous-time domain, whilst a linear quadratic Gaussian (LQG) control approach is suggested for the discrete-time domain.PhD Doctorate
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Rojas, Norman Alejandro Jose. "Feedback Control over Signal to Noise Ratio Constrained Communication Channels." 2006. http://hdl.handle.net/1959.13/24913.
Full textAbstract:
The present thesis addresses the problem of stabilisability of a linear time invariant (LTI) output feedback control loop in the presence of a communication link. The communication link itself can be either located between the controller and the plant or between the plant and the controller. The communication link is assumed to be an additive coloured Gaussian noise channel with (or without) bandwidth limitation (memory) in the continuous-(or discrete-)time domain. The requirement for stabilisability of the feedback loop is then characterised as a lower bound on the channel signal to noise ratio (SNR). This lower bound is tight and it will depend on the channel model, plant and channel model NMP zeros, plant time delay and plant unstable poles. Performance requirements are also investigated, by loop shaping in the continuous-time domain, whilst a linear quadratic Gaussian (LQG) control approach is suggested for the discrete-time domain.PhD Doctorate
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Wei, Cheng-Wen, and 魏誠文. "Low Power Noise and Feedback Reduction for Digital CIC Hearing Aids." Thesis, 2013. http://ndltd.ncl.edu.tw/handle/3nkcch.
Full textAbstract:
博士國立交通大學
電子研究所
101
With the advanced digital technology and signal processing, digital hearing aids have more potential to provide good performance to improve user usage experience. However, these sophisticated signal processing algorithms are still hard to be integrated due to the limitation of battery size and capacity, which demands efficient low power algorithm, architecture and circuit design. Thus, this dissertation proposes low power designs for two fundamental blocks of hearings aids: noise reduction (NR) and feedback cancellation (FC). The proposed NR designs are based on perceptual decomposition for efficient processing. The first NR design adopts a mixed frequency decomposition in conjunction with an efficient spectral subtraction and VAD (voice activity detection) for ultra low power noise suppression. The design can achieve about 4dB SNR improvement in low SNR environment and only consumes 0.65μW at 1.0V operation using 0.18μm process. However, this design adopts a simple scheme for NR, thus not providing good perceptual performance. To solve this problem, the second NR proposes an efficient multiband spectral subtraction design by using sample based processing, data preprocessing scheme and other sophisticated strategies to meet low power and low latency requirement. This design can achieve robust sound quality improvement in terms of SNR, PESQ and composite measure with 83.7μW at 0.6V operation with 90nm HVT (high VT) standard cell library. The performance of the second design is limited by the accuracy of entropy VAD in low SNR and nonstationary environment. To solve this problem, the third design proposes an efficient pitch based VAD for robust voice detection to assist noise suppression. This VAD has an efficient structure and is robust even in nonstationary environment. Based on this VAD, the noise suppression can provide 4dB SNR improvement with 55.52μW at 0.5V operation with 0.65μm high VT process. The pitch based processing is further applied to FC design which uses pitch results to estimate speech formant to enhance the robustness and the sound quality of adaptive feedback cancellation (AFC). The proposed AFC design can achieve similar added stable gain (ASG) and PESQ but with five orders complexity reduction compared to conventional designs. Based on the pitch based information, this dissertation also proposes an efficient pitch based processor for further system development.