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Relevant bibliographies by topics / Speech filtering / Dissertations / Theses

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Dissertations / Theses on the topic 'Speech filtering'

Author: Grafiati

Published: 4 June 2021

Last updated: 4 February 2022

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1

Ledoux, Christelle Michelle. "Robust speech filtering in impulsive noise environments." Thesis, Virginia Tech, 1999. http://hdl.handle.net/10919/46325.

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This thesis presents a new robust filtering technique that suppresses impulsive noise in speech signals. The method makes use of Projection Statistics based on medians to detect segments of speech with impulses. The autoregressive model employed to smooth out the speech signal is identified by means of a robust nonlinear estimator known as the Schweppe-type Huber GM-estimator. Simulation results are presented that demonstrate the effectiveness of the filter. Another contribution of the work is the development of a robust version of the Kalman filter based on the Huber M-estimator. The performa

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2

Ramachandran, Ravi P. "Pitch filtering in adaptive predictive coding of speech." Thesis, McGill University, 1986. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=65345.

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3

Klein, Mark 1977. "Signal subspace speech enhancement with perceptual post-filtering." Thesis, McGill University, 2002. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=33975.

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Speech enhancement blocks form a critical part of voice communications systems. Unfortunately, most enhancement schemes have difficulty eliminating noise from speech without introducing distortion or artefacts. Many of the disturbances originate from poor parameter estimation and interframe fluctuations.<br>This thesis introduces the Enhanced Signal Subspace (ESS) system to mitigate the above problems. Based on a signal subspace framework, ESS has been designed to attenuate disturbances while minimizing audible distortion.<br>Artefacts are reduced by employing an auditory post-filter to smooth

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4

Chan, Dominic Sai Fan. "Speech production modelling based on glottal inverse filtering." Thesis, Imperial College London, 1994. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.307161.

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5

Lewine, Andrew (Andrew P. ). "Speech filtering for improving intelligibility in noisy transients." Thesis, Massachusetts Institute of Technology, 2011. http://hdl.handle.net/1721.1/66433.

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Thesis (M. Eng.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 2011.<br>Cataloged from PDF version of thesis.<br>Includes bibliographical references.<br>Hearing impairment is a problem that affects a large percentage of the population. Cochlear implants allow those with profound or total hearing loss to regain some hearing by stimulating auditory nerve fibers with implanted electrodes, in response to sound picked up by an external microphone. The signal processing chain from microphone input to stimulation output is an important factor in the ove

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6

Dubbin, Gregory. "Applying particle filtering to unsupervised part-of-speech induction." Thesis, University of Oxford, 2014. http://ora.ox.ac.uk/objects/uuid:48caedb6-478f-4bb0-8ca7-975ee7fe5e38.

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Statistical Natural Language Processing (NLP) lies at the intersection of Computational Linguistics and Machine Learning. As linguistic models incorporate more subtle nuances of language and its structure, standard inference techniques can fall behind. One such application is research on the unsupervised induction of part-of-speech tags. It has the potential to improve both our understanding of the plausibility of theories of first language acquisition, and Natural Language Processing applications such as Speech Recognition and Machine Translation. Sequential Monte Carlo (SMC) approaches, i.e.

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7

Papanagiotou, Kyriakos. "Enhancement of body conducted speech from an ear microphone." Thesis, University of Southampton, 2003. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.289914.

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8

Darlington, David J. "The enhancement of noise-corrupted speech by sub-band adaptive filtering." Thesis, University of the West of Scotland, 1998. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.388213.

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9

Hu, Rong. "Enhancement of adaptive de-correlation filtering separation model for robust speech recognition." Diss., Columbia, Mo. : University of Missouri-Columbia, 2007. http://hdl.handle.net/10355/4682.

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Thesis (Ph. D.)--University of Missouri-Columbia, 2007.<br>The entire dissertation/thesis text is included in the research.pdf file; the official abstract appears in the short.pdf file (which also appears in the research.pdf); a non-technical general description, or public abstract, appears in the public.pdf file. Title from title screen of research.pdf file (viewed on September 25, 2007) Vita. Includes bibliographical references.

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10

Mustiere, Frederic. "Particle filtering methods for the enhancement of speech corrupted by additive noise." Thesis, University of Ottawa (Canada), 2006. http://hdl.handle.net/10393/27398.

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In this work, we study the application of particle filtering (PF) algorithms to the problem of speech enhancement. The goal of the thesis is to devise PF algorithms that will enhance speech signals corrupted by additive noise, and to evaluate their performance via comparisons with other existing algorithms based on several quality measures. Speech enhancement, or noise reduction, is an important problem in many applications, such as telephony and telecommunications in general, sound recording, human-coaching interface (where speech recognition is important), etc. Even though many algorithms al

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11

Wang, Yao Electrical Engineering &amp Telecommunications Faculty of Engineering UNSW. "Single channel speech enhancement based on perceptual temporal masking model." Awarded by:University of New South Wales. Electrical Engineering & Telecommunications, 2007. http://handle.unsw.edu.au/1959.4/40454.

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In most speech communication systems, the presence of background noise causes the quality and intelligibility of speech to degrade, especially when the Signal-to-Noise Ratio (SNR) is low. Numerous speech enhancement techniques have been employed successfully in many applications. However, at low signal-to-noise ratios most of these speech enhancement techniques tend to introduce a perceptually annoying residual noise known as "musical noise". The research presented in this thesis aims to minimize this musical noise and maximize the noise reduction ability of speech enhancement algorithms to im

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12

Ma, Ning. "Speech enhancement algorithms using Kalman filtering and masking properties of human auditory systems." Thesis, University of Ottawa (Canada), 2005. http://hdl.handle.net/10393/29229.

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Speech enhancement algorithms have been employed successfully in many areas such as VoIP, automatic speech recognition and speaker verification. Many approaches are presented in the literature. This thesis focuses on enhancing single channel speech degraded by white noise or colored noise. A Kalman filter algorithm combined with the masking properties of human auditory systems is proposed. The threshold computed from the masking properties is used as a constraint in the Kalman filter to theoretically derive a modified Kalman filter. The derivation gives a theoretical foundation for the feasibi

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13

Gransden, I. R. "High speed auditory analysis." Thesis, University of Sheffield, 1995. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.364247.

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14

Gobl, Christer. "The Voice Source in Speech Communication - Production and Perception Experiments Involving Inverse Filtering and Synthesis." Doctoral thesis, KTH, Speech Transmission and Music Acoustics, 2003. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-3665.

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<p>This thesis explores, through a number of production andperception studies, the nature of the voice source signal andhow it varies in spoken communication. Research is alsopresented that deals with the techniques and methodologies foranalysing and synthesising the voice source. The main analytictechnique involves interactive inverse filtering for obtainingthe source signal, which is then parameterised to permit thequantification of source characteristics. The parameterisationis carried by means of model matching, using the four-parameterLF model of differentiated glottal flow.</p><p>The fir

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15

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

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16

Ozbek, Ibrahim Yucel. "Dynamic System Modeling And State Estimation For Speech Signal." Phd thesis, METU, 2010. http://etd.lib.metu.edu.tr/upload/3/12611777/index.pdf.

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This thesis presents an all-inclusive framework on how the current formant tracking and audio (and/or visual)-to-articulatory inversion algorithms can be improved. The possible improvements are summarized as follows: The first part of the thesis investigates the problem of the formant frequency estimation when the number of formants to be estimated fixed or variable respectively. The fixed number of formant tracking method is based on the assumption that the number of formant frequencies is fixed along the speech utterance. The proposed algorithm is based on the combination of a dynamic pro

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17

Abel, Andrew. "Towards an intelligent fuzzy based multimodal two stage speech enhancement system." Thesis, University of Stirling, 2013. http://hdl.handle.net/1893/15989.

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This thesis presents a novel two stage multimodal speech enhancement system, making use of both visual and audio information to filter speech, and explores the extension of this system with the use of fuzzy logic to demonstrate proof of concept for an envisaged autonomous, adaptive, and context aware multimodal system. The design of the proposed cognitively inspired framework is scalable, meaning that it is possible for the techniques used in individual parts of the system to be upgraded and there is scope for the initial framework presented here to be expanded. In the proposed system, the con

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Nallamilli, Sai Chandra Sekhar Reddy, and Nihanth Kandi. "Detection of Human Emotion from Noise Speech." Thesis, Blekinge Tekniska Högskola, Institutionen för tillämpad signalbehandling, 2020. http://urn.kb.se/resolve?urn=urn:nbn:se:bth-19610.

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Detection of a human emotion from human speech is always a challenging task. Factors like intonation, pitch, and loudness of signal vary from different human voice. So, it's important to know the exact pitch, intonation and loudness of a speech for making it a challenging task for detection. Some voices exhibit high background noise which will affect the amplitude or pitch of the signal. So, knowing the detailed properties of a speech to detect emotion is mandatory. Detection of emotion in humans from speech signals is a recent research field. One of the scenarios where this field has been app

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19

Thüne, Philipp [Verfasser], Gerald [Gutachter] Enzner, and Peter [Gutachter] Jax. "Advances in blind multichannel Wiener filtering of noisy speech / Philipp Thüne ; Gutachter: Gerald Enzner, Peter Jax ; Fakultät für Elektrotechnik und Informationstechnik." Bochum : Ruhr-Universität Bochum, 2017. http://d-nb.info/1150509546/34.

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20

Torres, Juan Félix. "Estimation of glottal source features from the spectral envelope of the acoustic speech signal." Diss., Georgia Institute of Technology, 2010. http://hdl.handle.net/1853/34736.

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Speech communication encompasses diverse types of information, including phonetics, affective state, voice quality, and speaker identity. From a speech production standpoint, the acoustic speech signal can be mainly divided into glottal source and vocal tract components, which play distinct roles in rendering the various types of information it contains. Most deployed speech analysis systems, however, do not explicitly represent these two components as distinct entities, as their joint estimation from the acoustic speech signal becomes an ill-defined blind deconvolution problem. Nevertheless,

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21

Wu, Mingyang. "Pitch tracking and speech enhancement in noisy and reverberant environments." Columbus, Ohio : Ohio State University, 2003. http://rave.ohiolink.edu/etdc/view?acc%5Fnum=osu1064341479.

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Thesis (Ph. D.)--Ohio State University, 2003.<br>Title from first page of PDF file. Document formatted into pages; contains xvi, 149 p.; also includes graphics. Includes abstract and vita. Advisor: DeLiang Wang, Dept. of Computer and Information Science. Includes bibliographical references (p. 136-149).

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Roman, Nicoleta. "Auditory-based algorithms for sound segregation in multisource and reverberant environments." Connect to resource, 2005. http://rave.ohiolink.edu/etdc/view?acc%5Fnum=osu1124370749.

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Thesis (Ph. D.)--Ohio State University, 2005.<br>Title from first page of PDF file. Document formatted into pages; contains i-xxii, xx-xxi, 183 p.; also includes graphics. Includes bibliographical references (p. 171-183). Available online via OhioLINK's ETD Center

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23

Tan, Ke. "Convolutional and recurrent neural networks for real-time speech separation in the complex domain." The Ohio State University, 2021. http://rave.ohiolink.edu/etdc/view?acc_num=osu1626983471600193.

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24

Neville, Katrina Lee, and katrina neville@rmit edu au. "Channel Compensation for Speaker Recognition Systems." RMIT University. Electrical and Computer Engineering, 2007. http://adt.lib.rmit.edu.au/adt/public/adt-VIT20080514.093453.

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This thesis attempts to address the problem of how best to remedy different types of channel distortions on speech when that speech is to be used in automatic speaker recognition and verification systems. Automatic speaker recognition is when a person's voice is analysed by a machine and the person's identity is worked out by the comparison of speech features to a known set of speech features. Automatic speaker verification is when a person claims an identity and the machine determines if that claimed identity is correct or whether that person is an impostor. Channel distortion occurs wh

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25

Deivard, Johannes. "How accuracy of estimated glottal flow waveforms affects spoofed speech detection performance." Thesis, Mälardalens högskola, Akademin för innovation, design och teknik, 2020. http://urn.kb.se/resolve?urn=urn:nbn:se:mdh:diva-48414.

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In the domain of automatic speaker verification, one of the challenges is to keep the malevolent people out of the system. One way to do this is to create algorithms that are supposed to detect spoofed speech. There are several types of spoofed speech and several ways to detect them, one of which is to look at the glottal flow waveform (GFW) of a speech signal. This waveform is often estimated using glottal inverse filtering (GIF), since, in order to create the ground truth GFW, special invasive equipment is required. To the author’s knowledge, no research has been done where the correl

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Al-saqaf, Walid. "Breaking digital firewalls : analyzing internet censorship and circumvention in the arab world." Doctoral thesis, Örebro universitet, Institutionen för humaniora, utbildnings- och samhällsvetenskap, 2014. http://urn.kb.se/resolve?urn=urn:nbn:se:oru:diva-34596.

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This dissertation explores the role of Internet censorship and circumvention in the Arab world as well as Arabs’ views on the limits to free speech on the Internet. The project involves the creation of an Internet censorship circumvention tool named Alkasir that allows users to report and access certain types of censored websites. The study covers the Arab world at large with special focus on Egypt, Syria, Tunisia, and Yemen. This work is of interdisciplinary nature and draws on the disciplines of media and communication studies and computer science. It uses a pioneering experimental approach

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Motlagh, Zadeh Lina. "Developing a digits in noise screening test with higher sensitivity to high-frequency hearing loss." University of Cincinnati / OhioLINK, 2019. http://rave.ohiolink.edu/etdc/view?acc_num=ucin1552378973670023.

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28

Houdek, Miroslav. "Rozpoznání emočního stavu člověka z řeči." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2009. http://www.nusl.cz/ntk/nusl-218117.

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This master thesis concerns with emotional states and gender recognition on the basis of speech signal analysis. We used various prosodic and cepstral features for the description of the speech signal. In the text we describe non-invasive methods for glottal pulses estimation. The described features of speech were implemented in MATLAB. For their classification we used the GMM classifier, which uses the Gaussian probability distribution for modeling a feature space. Furthermore, we constructed a system for recognition of emotional states of the speaker and a system for gender recognition from

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Rao, Peddi Srinivas, and Vallabhaneni Sreelatha. "Implementation and Evaluation of Spectral Subtraction with Minimum Statistics using WOLA and FFT Modulated Filter Banks." Thesis, Blekinge Tekniska Högskola, Institutionen för tillämpad signalbehandling, 2014. http://urn.kb.se/resolve?urn=urn:nbn:se:bth-2906.

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In communication system environment speech signal is corrupted due to presence of additive acoustic noise, so with this distortion the effective communication is degraded in terms of the quality and intelligibility of speech. Now present research is going how effectively acoustic noise can be eliminated without affecting the original speech quality, this tends to be our challenging in this current research thesis work. Here this work proposes multi-tiered detection method that is based on time-frequency analysis (i.e. filter banks concept) of the noisy speech signals, by using standard speech

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Matos, Adriano Nogueira. "Extração de características do sinal de voz utilizando análise fatorial verdadeira." Universidade Federal do Amazonas, 2008. http://tede.ufam.edu.br/handle/tede/2959.

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Made available in DSpace on 2015-04-11T14:03:17Z (GMT). No. of bitstreams: 1 DISSERTACAO ADRIANO NOGUEIRA.pdf: 382280 bytes, checksum: fc1f9e0caac3d97ff74a893e97298a71 (MD5) Previous issue date: 2008-12-17<br>Coordenação de Aperfeiçoamento de Pessoal de Nível Superior<br>Digital processing of speech signal is applied in several computer applications, which the major ones are the following: Recognition, synthesis and coding of speech. All these applications require the amount of data in the acoustic signal to be reduced, in order to allow processing by a computer device. The feature extract

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Crespo, Cuaresma Jesus, and Martin Feldkircher. "Spatial Filtering, Model Uncertainty and the Speed of Income Convergence in Europe." Wiley, 2013. http://dx.doi.org/10.1002/jae.2277.

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In this paper we put forward a Bayesian Model Averaging method aimed at performing inference under model uncertainty in the presence of potential spatial autocorrelation. The method uses spatial filtering in order to account for uncertainty in spatial linkages. Our procedure is applied to a dataset of income per capita growth and 50 potential determinants for 255 NUTS-2 European regions. We show that ignoring uncertainty in the type of spatial weight matrix can have an important effect on the estimates of the parameters attached to the model covariates. After integrating out the uncerta

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Olugbenga, Olubodun. "High speed optical phase modulated signaling with offset filtering in a 50 GHz grid." Thesis, Swansea University, 2011. https://cronfa.swan.ac.uk/Record/cronfa42896.

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Hawkins, Mikhel E. "High speed target tracking using Kalman filter and partial window imaging." Thesis, Georgia Institute of Technology, 2002. http://hdl.handle.net/1853/16709.

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34

Sturmel, Nicolas. "Analyse de la qualité vocale appliquée à la parole expressive." Phd thesis, Université Paris Sud - Paris XI, 2011. http://tel.archives-ouvertes.fr/tel-00591638.

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L'analyse des signaux de parole permet de comprendre le fonctionnement de l'appareil vocal, mais aussi de décrire de nouveaux paramètres permettant de qualifier et quantifier la perception de la voix. Dans le cas de la parole expressive, l'intérêt se porte sur des variations importantes de qualité vocales et sur leurs liens avec l'expressivité et l'intention du sujet. Afin de décrire ces liens, il convient de pouvoir estimer les paramètres du modèle de production mais aussi de décomposer le signal vocal en chacune des parties qui contribuent à ce modèle. Le travail réalisé au cours de cette th

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Hollis, Timothy Mowry. "Circuit and Modeling Solutions for High-Speed Chip-to-Chip Communication." BYU ScholarsArchive, 2007. https://scholarsarchive.byu.edu/etd/1067.

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This dissertation presents methods for modeling and mitigating voltage noise and timing jitter across high-speed chip-to-chip interconnects. Channel equalization and associated tuning schemes have been developed to target the distinct characteristics and signal degradation exhibited in the clock and data signals of multi-Gigabit/second digital communication links. Multiple methods for generating realistically degraded signals for the purpose of simulation are also presented and used to verify the proposed equalization and filtering topologies. Specifically, a new technique for modeling high-sp

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Jemâa, Imen. "Suivi de formants par analyse en multirésolution." Thesis, Université de Lorraine, 2013. http://www.theses.fr/2013LORR0026/document.

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Nos travaux de recherches présentés dans ce manuscrit ont pour objectif, l'optimisation des performances des algorithmes de suivi des formants. Pour ce faire, nous avons commencé par l'analyse des différentes techniques existantes utilisées dans le suivi automatique des formants. Cette analyse nous a permis de constater que l'estimation automatique des formants reste délicate malgré l'emploi de diverses techniques complexes. Vue la non disponibilité des bases de données de référence en langue arabe, nous avons élaboré un corpus phonétiquement équilibré en langue arabe tout en élaborant un étiq

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Vatte, Madhu Latha Reddy. "Readout Circuitry for a Logarithmic CMOS Active Pixel Sensor That Facilities High Speed Image Processing." University of Akron / OhioLINK, 2010. http://rave.ohiolink.edu/etdc/view?acc_num=akron1278549382.

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Lai, Ying-Chun. "A Development of a Common-Mode FilterUsing an EBG Structure in High Speed SerialLinks." Thesis, KTH, Elektroteknisk teori och konstruktion, 2012. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-104986.

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As signal speed increases and electronic products become progressively smaller,the risks of electromagnetic radiation and interference are also heightened.Ericsson's SCXB, an Ethernet switch card, experiences exactly this problem,with excessive emission levels probably caused by common-mode noise.In this project, a common-mode lter using the electromagnetic bandgap(EBG) structure has been designed and implemented in the SCXB. Unlikeconventional common-mode lters, the common-mode lter is embedded inthe printed circuit board (PCB) beneath the dierential lines. The eect ofthe common-mode lter is

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Hamlet, Sean Michael. "COMPARING ACOUSTIC GLOTTAL FEATURE EXTRACTION METHODS WITH SIMULTANEOUSLY RECORDED HIGH-SPEED VIDEO FEATURES FOR CLINICALLY OBTAINED DATA." UKnowledge, 2012. http://uknowledge.uky.edu/ece_etds/12.

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Accurate methods for glottal feature extraction include the use of high-speed video imaging (HSVI). There have been previous attempts to extract these features with the acoustic recording. However, none of these methods compare their results with an objective method, such as HSVI. This thesis tests these acoustic methods against a large diverse population of 46 subjects. Two previously studied acoustic methods, as well as one introduced in this thesis, were compared against two video methods, area and displacement for open quotient (OQ) estimation. The area comparison proved to be somewhat amb

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Pokora, C. D. "Spatio-temporal correlations of jets using high-speed particle image velocimetry." Thesis, Loughborough University, 2009. https://dspace.lboro.ac.uk/2134/13185.

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The major source of aircraft noise at take-off is jet noise. If jet noise is not adequately addressed environmental impact concerns will constrain the planned growth of the air transport system. A considerable amount of research worldwide has therefore been aimed at identifying ways to reduce jet noise including development of a predictive tool that can estimate the noise generated by new nozzle designs. Current noise prediction techniques, however, still require the input of empirically calibrated noise source models and their performance is still inadequate. In addition, development of detai

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Chenais, Patrick. "Une carte de traitement et de reconnaissance de la parole : etude de cibles acoustiques." Toulouse 3, 1987. http://www.theses.fr/1987TOU30009.

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Realisation d'une carte de traitement numerique de la parole. L'application visee est la commande d'actionneurs dans un dispositif de controle d'environnement. Les realisations materielle et logicielle sont decrites. Pour diminuer le cout, l'idee de base de l'etude de l'architecture est d''integrer sur le meme module la fonction reconnaissance et le fonction commande. La reconnaissance utilise cinq indices acoustiques. Ils sont calcules en temps reel par le processeur de signal. La segmentation de la parole en unites discretes est abordee selon une strategie de recherche de cibles acoustiques.

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Caliskan, Hakan. "Modeling And Experimental Evaluation Of Variable Speed Pump And Valve Controlled Hydraulic Servo Drives." Master's thesis, METU, 2009. http://etd.lib.metu.edu.tr/upload/3/12611090/index.pdf.

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In this thesis study, a valveless hydraulic servo system controlled by two pumps is investigated and its performance characteristics are compared with a conventional valve controlled system both experimentally and analytically. The two control techniques are applied on the position control of a single rod linear actuator. In the valve controlled system, the flow rate through the actuator is regulated with a servovalve<br>whereas in the pump controlled system, two variable speed pumps driven by servomotors regulate the flow rate according to the needs of the system, thus eliminating the valve l

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Buyukkeles, Umit. "Improved Torque And Speed Control Performance In A Vector-controlled Pwm-vsi Fed Surface-mounted Pmsm Drive With Conventional P-i Controllers." Master's thesis, METU, 2012. http://etd.lib.metu.edu.tr/upload/12614294/index.pdf.

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In this thesis, high performance torque and speed control for a surface-mounted permanent magnet synchronous machine (PMSM) is designed, simulated and implemented. A three-phase two-level pulse width modulation voltage-source inverter (PWM-VSI) with power MOSFETs is used to feed the PMSM. The study has three objectives. The first is to compensate the voltage disturbance caused by nonideal characteristics of the voltage-source inverter (VSI). The second is to decouple the coupled variables in the synchronous reference frame model of the PMSM. The last is to design a load torque estimator in ord

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Olsson, Rickard. "Signal processing and high speed imaging as monitoring tools for pulsed laser welding." Licentiate thesis, Luleå tekniska universitet, Produkt- och produktionsutveckling, 2009. http://urn.kb.se/resolve?urn=urn:nbn:se:ltu:diva-26555.

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In Laser Materials Processing there has always been a need for suitable methods to supervise and monitor the processes on line, to ensure correct production quality or to trigger alarms when failures are detected. Numerous investigations have been made in this field, including experimental and theoretical work. It is common practice in this field to monitor surface temperature, plasma radiation and back-reflected laser light, coaxially with the laser beam. Traditionally, the monitoring systems involved carry out no statistical analysis of the signals received - they merely involve thresholds.

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Bartholomew, David Ray. "Design of a High Speed Mixed Signal CMOS Mutliplying Circuit." Diss., CLICK HERE for online access, 2004. http://contentdm.lib.byu.edu/ETD/image/etd362.pdf.

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Kučera, Jan. "Filtrace paketů ve 100 Gb sítích." Master's thesis, Vysoké učení technické v Brně. Fakulta informačních technologií, 2016. http://www.nusl.cz/ntk/nusl-255423.

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This master's thesis deals with the design and implementation of an algorithm for high-speed network packet filtering. The main goal was to provide hardware architecture, which would support large rule sets and could be used in 100 Gbps networks. The system has been designed with respect to the implementation on an FPGA card and time-space complexity trade-off. Properties of the system have been evaluated using various available rule sets. Due to the highly optimized and deep pipelined architecture it was possible to reach high working frequency (above 220 MHz) together with considerable memor

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Ispir, Mehmet. "Design Of Moving Target Indication Filters With Non-uniform Pulse Repetition Intervals." Master's thesis, METU, 2013. http://etd.lib.metu.edu.tr/upload/12615361/index.pdf.

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Staggering the pulse repetititon intervals is a widely used solution to alleviate the blind speed problem in Moving Target Indication (MTI) radar systems. It is possible to increase the first blind speed on the order of ten folds with the use of non-uniform sampling. Improvement in blind speed results in passband fluctuations that may degregade the detection performance for particular Doppler frequencies. Therefore, it is important to design MTI filters with non-uniform interpulse periods that have minimum passband ripples with sufficient clutter attenuation along with good range and blind vel

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Törnquist, Martin. "Investigation of rotational velocity sensors." Thesis, Linköping University, Department of Electrical Engineering, 2008. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-15904.

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<p><p>To improve the speed measurement of construction equipment, different sensor technologies have been investigated. Many of these sensor technologies are very interesting but to keep the extent of the thesis only two was chosen for testing, magnetic absolute angle sensors using Hall and GMR technology, to investigate if those are a valid replacement for the current measurement system that is using a passive sensor. Tests show that these sensors are capable of speed measurement, but because of noisy angle estimates they need filtering for good speed computation. This filtering introduces a

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Silva, Cristiane Cristina Sousa da. "UM ALGORITMO TIPO RLS BASEADO EM SUPERFÍCIES NÃO QUADRÁTICAS." Universidade Federal do Maranhão, 2013. http://tedebc.ufma.br:8080/jspui/handle/tede/550.

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Made available in DSpace on 2016-08-17T16:54:33Z (GMT). No. of bitstreams: 1 Tese Cristiane Cristina.pdf: 4404224 bytes, checksum: a68e5757bedc2d3d341a5937f100fe1f (MD5) Previous issue date: 2013-07-19<br>Coordenação de Aperfeiçoamento de Pessoal de Nível Superior<br>In adaptive filtering many adaptive filter are based on the mean square error method (MSE). These filters were developed to improve convergence spedd with a lower misadjustment. The least mean square (LMS) and the recursive least square (RLS) algorithms have been the hallmark of adaptive filtering. In this work we develop adap

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Hodaň, David. "Možnosti akcelerace symbolické regrese pomocí kartézského genetického programování." Master's thesis, Vysoké učení technické v Brně. Fakulta informačních technologií, 2019. http://www.nusl.cz/ntk/nusl-403198.

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This thesis is focused on finding procedures that would accelerate symbolic regressions in Cartesian Genetic Programming. It describes Cartesian Genetic Programming and its use in the task of symbolic regression. It deals with the SIMD architecture and the SSE and AVX instruction set. Several optimizations that lead to a significant acceleration of evolution in Cartesian Genetic Programming are presented. A method of a bit-level parallel simulation that uses AVX2 vectors allows to process 256 input combinations of a logic circuit in paralell. Similarly it is possible to use a byte-level parall

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