Дисертації з теми "Noisy feedback"
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Quintero, Florez Victor. "Noisy channel-output feedback in the interference channel." Thesis, Lyon, 2017. http://www.theses.fr/2017LYSEI128/document.
Повний текст джерелаIn 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
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.
Повний текст джерелаCette 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.
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.
Повний текст джерелаTukhlina, Natalia. "Feedback control of complex oscillatory systems." Phd thesis, Universität Potsdam, 2008. http://opus.kobv.de/ubp/volltexte/2008/1854/.
Повний текст джерелаIn 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.
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.
Повний текст джерелаThesis 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.
Duke, Cole Victor. "Analog Feedback Control of Broadband Fan Noise." BYU ScholarsArchive, 2012. https://scholarsarchive.byu.edu/etd/3646.
Повний текст джерелаBoglione, Luciano. "Low noise microwave feedback amplifier design with simultaneous signal and noise matching." Thesis, University of Leeds, 1998. http://etheses.whiterose.ac.uk/900/.
Повний текст джерелаPawełczyk, 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.
Повний текст джерелаLongtin, 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.
Повний текст джерелаRaja, 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.
Повний текст джерелаCreasy, Miles Austin. "Adaptive Collocated Feedback for Noise Absorption in Acoustic Enclosures." Thesis, Virginia Tech, 2006. http://hdl.handle.net/10919/45209.
Повний текст джерелаMaster of Science
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.
Повний текст джерелаKlein, 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.
Повний текст джерелаIncludes 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.
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.
Повний текст джерелаAmoêdo, 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.
Повний текст джерелаIn 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.
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.
Повний текст джерелаKawamura, 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.
Повний текст джерелаXiong, 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.
Повний текст джерелаWeststrate, 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.
Повний текст джерелаThesis (PhD(Eng))--University of Pretoria, 2011.
Electrical, Electronic and Computer Engineering
unrestricted
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.
Повний текст джерелаIncludes bibliographical references (p. 151-156).
by Kambiz C. Zangi.
Ph.D.
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.
Повний текст джерелаMaster of Science
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.
Повний текст джерелаSagers, 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.
Повний текст джерелаForsgren, 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.
Повний текст джерелаZylich, 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.
Повний текст джерелаWerner, 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.
Повний текст джерелаMunyai, Pandelani Reuben Mulalo. "On the improvement of phase noise in wideband frequency synthesizers." Diss., University of Pretoria, 2017. http://hdl.handle.net/2263/63003.
Повний текст джерелаThesis (MEng)--University of Pretoria, 2017.
Electrical, Electronic and Computer Engineering
MEng
Unrestricted
Nguyen, Lan K. "Dynamical modelling of feedback gene regulatory networks." Diss., Lincoln University, 2009. http://hdl.handle.net/10182/1340.
Повний текст джерелаAdiseno. "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.
Повний текст джерелаIn 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.
Zhou, Limin. "ASSESSING AND MITIGATING AIRBORNE NOISE FROM POWER GENERATION EQUIPMENT." UKnowledge, 2013. http://uknowledge.uky.edu/me_etds/22.
Повний текст джерелаSiviero, 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.
Повний текст джерелаDissertação (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
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.
Повний текст джерелаTokatli, 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.
Повний текст джерелаcomposition of tones, drill noise and propeller plane cabin noise. The results reveal that adaptive system has better overall performance.
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.
Повний текст джерелаFrank, 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.
Повний текст джерелаWickert, 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.
Повний текст джерелаMany 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.
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.
Повний текст джерелаNeurons 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.
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.
Повний текст джерелаBuller 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.
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.
Повний текст джерела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.
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.
Повний текст джерела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.
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.
Повний текст джерела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.
Daud, Muhyin, and 李明言. "Phase-Noise Analysis and Improvement of Microwave Feedback Oscillators." Thesis, 2013. http://ndltd.ncl.edu.tw/handle/46940951885937007638.
Повний текст джерела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.
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.
Повний текст джерела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.
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.
Повний текст джерела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.
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.
Повний текст джерела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.
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.
Повний текст джерела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.
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.
Повний текст джерелаRojas, Norman Alejandro Jose. "Feedback control over signal to noise ratio constrained communication channels." Thesis, 2006. http://hdl.handle.net/1959.13/24913.
Повний текст джерелаPhD Doctorate
Rojas, Norman Alejandro Jose. "Feedback Control over Signal to Noise Ratio Constrained Communication Channels." 2006. http://hdl.handle.net/1959.13/24913.
Повний текст джерелаPhD Doctorate
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.
Повний текст джерела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.