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1
Majidi, Mohammad Hassan. "Bayesian estimation of discrete signals with local dependencies." Thesis, Supélec, 2014. http://www.theses.fr/2014SUPL0014/document.
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L'objectif de cette thèse est d'étudier le problème de la détection de données dans le système de communication sans fil, à la fois pour le cas de l'information d'état de canal parfaite et imparfaite au niveau du récepteur. Comme on le sait, la complexité de MLSE est exponentielle en la mémoire de canal et la cardinalité de l'alphabet symbole est rapidement ingérable, ce qui force à recourir à des approches sousoptimales. Par conséquent, en premier lieu, nous proposons une nouvelle égalisation itérative lorsque le canal est inconnu à l'émetteur et parfaitement connu au niveau du récepteur. Ce récepteur est basé sur une approche de continuation, et exploite l'idée d'approcher une fonction originale de coût d'optimisation par une suite de fonctions plus dociles et donc de réduire la complexité de calcul au récepteur.En second lieu, en vue de la détection de données sous un canal dynamique linéaire, lorsque le canal est inconnu au niveau du récepteur, le récepteur doit être en mesure d'effectuer conjointement l'égalisation et l'estimation de canal. De cette manière, on formule une représentation de modèle état-espace combiné du système de communication. Par cette représentation, nous pouvons utiliser le filltre de Kalman comme le meilleur estimateur des paramètres du canal. Le but de cette section est de motiver de façon rigoureuse la mise en place du filltre de Kalman dans l'estimation des sequences de Markov par des canaux dynamiques Gaussien. Par la présente, nous interprétons et explicitons les approximations sous-jacentes dans les approaches heuristiques.Enfin, si nous considérons une approche plus générale pour le canal dynamique non linéaire, nous ne pouvons pas utiliser le filtre de Kalman comme le meilleur estimateur. Ici, nous utilisons des modèles commutation d’espace-état (SSSM) comme modèles espace-état non linéaires. Ce modèle combine le modèle de Markov caché (HMM) et le modèle espace-état linéaire (LSSM). Pour l'estimation de canal et la detection de données, l'approche espérance et maximisation (EM) est utilisée comme approche naturelle. De cette façon, le filtre de Kalman étendu (EKF) et les filtres à particules sont évitésThe aim of this thesis is to study the problem of data detection in wireless communication system, for both case of perfect and imperfect channel state information at the receiver. As well known, the complexity of MLSE being exponential in the channel memory and in the symbol alphabet cardinality is quickly unmanageable and forces to resort to sub-optimal approaches. Therefore, first we propose a new iterative equalizer when the channel is unknown at the transmitter and perfectly known at the receiver. This receiver is based on continuation approach, and exploits the idea of approaching an original optimization cost function by a sequence of more tractable functions and thus reduce the receiver's computational complexity. Second, in order to data detection under linear dynamic channel, when the channel is unknown at the receiver, the receiver must be able to perform joint equalization and channel estimation. In this way, we formulate a combined state-space model representation of the communication system. By this representation, we can use the Kalman filter as the best estimator for the channel parameters. The aim in this section is to motivate rigorously the introduction of the Kalman filter in the estimation of Markov sequences through Gaussian dynamical channels. By this we interpret and make clearer the underlying approximations in the heuristic approaches. Finally, if we consider more general approach for non linear dynamic channel, we can not use the Kalman filter as the best estimator. Here, we use switching state-space model (SSSM) as non linear state-space model. This model combines the hidden Markov model (HMM) and linear state-space model (LSSM). In order to channel estimation and data detection, the expectation and maximization (EM) procedure is used as the natural approach. In this way extended Kalman filter (EKF) and particle filters are avoided
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Sivaramakrishnan, Kamakshi. "Universal schemes for denoising discrete-time continuous-amplitude signals /." May be available electronically:, 2008. http://proquest.umi.com/login?COPT=REJTPTU1MTUmSU5UPTAmVkVSPTI=&clientId=12498.
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Trombetta, Jacob J. "Time-Frequency Representation of Musical Signals Using the Discrete Hermite Transform." University of Akron / OhioLINK, 2011. http://rave.ohiolink.edu/etdc/view?acc_num=akron1304100211.
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Löhr, Andrea. "A noise reduction method based upon statistical analysis for the detection of weak signals in discrete data." [S.l.] : [s.n.], 2003. http://deposit.ddb.de/cgi-bin/dokserv?idn=968817505.
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Slosman, Brian D. "Design of discrete time radio receiver for the demodulation of power-separated co-channel satellite communication signals." Thesis, Monterey, California: Naval Postgraduate School, 2013. http://hdl.handle.net/10945/37719.
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Approved for public release; distribution is unlimitedThis thesis has two purposes: 1) to document the design of a discrete-time radio receiver for the coherent detection of a QPSK signal in the presence of additive white Gaussian noise; and 2) further research into the performance of representative receivers in the successive demodulation of power-separated, co-channel satellite communications signals. Several commercial companies are offering satellite modulators and demodulators that allow frequency reuse over satellite communications links. There are two methods to demodulate these co-channel signals. The first method requires a priori knowledge of one of the two signals linearly superimposed in the satellite downlink. With this knowledge, the known signal is cancelled using subtraction to reveal the unknown co-channel signal. A second method of recovering both signals is possible if adequate power separation of the two signals allows recovery of the strong signal. After recovery of the strong signal, the data can be re-modulated and then cancelled from the composite signal to reveal the weak signal. This method has the advantage of not requiring a priori information which widens the applications for layered modulation techniques to simplex, broadcast, and multi-cast network architectures.
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Raza, Saqlain. "Essays on complementarity : organizational and market changes in agriculture." Thesis, Toulouse, INPT, 2014. http://www.theses.fr/2014INPT0017/document.
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Cette thèse vise à tester pour l’existence d’effet de complémentarités entre différentes activités économiques dans le secteur agricole. Pour cela, nous mobilisons les deux approches que proposent la littérature, à savoir l’approche par la productivité et l’approche par l’adoption. Nous commençons par une revue de la littérature sur l’économie de la complémentarité, en nous focalisant sur ces deux approches de la complémentarité et ses modèles empiriques. Nous proposons ensuite trois analyses empiriques permettant de tester ces modèles. La première explore les déterminants du choix de marque et/ou de signes des qualité par les petites coopératives agricoles françaises, avec un focus particulier sur la coexistence de ces deux signes. La seconde fournit un test direct de complémentarité entre labels et marques en recourant à l’approche par l’adoption. En estimant un probit multinomial, il est en effet possible de séparer l’effet de complémentarité de celui de l’hétérogénéité inobservable. La troisième introduit l’approche par la productivité, en sus de l’approche par l’adoption, pour tester de cet eet de complémentarité dans les systèmes de polyculture élevage adoptés par les petits exploitants de la province du Pendjab au PakistanThe main objective of this thesis is to test for complementarity between different economic activities in agriculture. To do this, we have recourse to the two approaches proposed by the literature, i.e. the productivity approach and the adoption approach. First, we review the economics of complementarity and analyze the different empirical models to test for complementarity. Then, we propose three empirical analyses testing these models. The first examine closely the drivers of the branding and labeling strategies from French small agricultural co-operatives, with a focus on the coexistence of both quality signals. The second directly test for complementarity between branding and labeling using the adoption approach, by estimating a multinomal probit. This allow us to separate what is really due to complementarity and what is caused by unobserved heterogeneity. Third, in addition to adoption approach, we test for complementarity using a productivity approach in the mixed farming systems adopted by smallholder farmers in Punjab, Pakistan
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Sayyah, Jahromi Mohammad Reza Information Technology & Electrical Engineering Australian Defence Force Academy UNSW. "Efficient broadband antenna array processing using the discrete fourier form transform." Awarded by:University of New South Wales - Australian Defence Force Academy. School of Information Technology and Electrical Engineering, 2005. http://handle.unsw.edu.au/1959.4/38690.
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Processing of broadband signals induced on an antenna array using a tapped delay line filter and a set of steering delays has two problems. Firstly one needs to manipulate large matrices to estimate the filter coefficients. Secondly the use of steering delays is not only cumbersome but implementation errors cause loss of system performance. This thesis looks at both of these problems and presents elegant solutions by developing and studying a design method referred to as the DFT method, which does not require steering delays and is computationally less demanding compared to existing methods. Specifically the thesis studies and compares the performance of a time domain element space beamformer using the proposed method and that using an existing method, and develops the DFT method when the processor is implemented in partitioned form. The study presented in the thesis shows that the processors using the DFT method are robust to look direction errors and require less computation than that using the existing method for comparable performance. The thesis further introduces a broadband beamformer design which does not require any steering delays between the sensors and the tapped delay line section as is presently the case. It has the capability of steering the array in an arbitrary direction with a specified frequency response in the look direction while canceling unwanted uncorrelated interferences. The thesis presents and compares the performance of a number of techniques to synthesize an antenna pattern of a broadband array. These techniques are designed to produce isolated point nulls as well as broad sector nulls and to eliminate the need for the steering delays. Two of the pattern synthesis techniques presented in the thesis allow optimization against unwanted interferences in unknown directions. The techniques allow formulation of a beamforming problem such that the processor is not only able to place nulls in specified directions but also able to cancel directional interferences in unknown directions along with a specified frequency response in the look direction over a band of interest. The thesis also presents a set of directional constraints such that one does not need steering delays and an array can be constrained in an arbitrary direction with a specified frequency response. The constraints presented in the thesis are simple to implement. Based on these constraints a pattern synthesis technique for broadband antenna array is also presented.
8
Wirsing, Karlton. "Application of Wavelets to Filtering and Analysis of Self-Similar Signals." Thesis, Virginia Tech, 2014. http://hdl.handle.net/10919/78087.
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Digital Signal Processing has been dominated by the Fourier transform since the Fast Fourier Transform (FFT) was developed in 1965 by Cooley and Tukey. In the 1980's a new transform was developed called the wavelet transform, even though the first wavelet goes back to 1910. With the Fourier transform, all information about localized changes in signal features are spread out across the entire signal space, making local features global in scope. Wavelets are able to retain localized information about the signal by applying a function of a limited duration, also called a wavelet, to the signal.
As with the Fourier transform, the discrete wavelet transform has an inverse transform, which allows us to make changes in a signal in the wavelet domain and then transform it back in the time domain. In this thesis, we have investigated the filtering properties of this technique and analyzed its performance under various settings. Another popular application of wavelet transform is data compression, such as described in the JPEG 2000 standard and compressed digital storage of fingerprints developed by the FBI. Previous work on filtering has focused on the discrete wavelet transform. Here, we extended that method to the stationary wavelet transform and found that it gives a performance boost of as much as 9 dB over that of the discrete wavelet transform. We also found that the SNR of noise filtering decreases as a frequency of the base signal increases up to the Nyquist limit for both the discrete and stationary wavelet transforms.
Besides filtering the signal, the discrete wavelet transform can also be used to estimate the standard deviation of the white noise present in the signal. We extended the developed estimator for the discrete wavelet transform to the stationary wavelet transform. As with filtering, it is found that the quality of the estimate decreases as the frequency of the base signal increases.
Many interesting signals are self-similar, which means that one of their properties is invariant on many different scales. One popular example is strict self-similarity, where an exact copy of a signal is replicated on many scales, but the most common property is statistical self-similarity, where a random segment of a signal is replicated on many different scales. In this work, we investigated wavelet-based methods to detect statistical self-similarities in a signal and their performance on various types of self-similar signals. Specifically, we found that the quality of the estimate depends on the type of the units of the signal being investigated for low Hurst exponent and on the type of edge padding being used for high Hurst exponent.Master of Science
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Mönich, Ullrich Johann [Verfasser], Holger [Akademischer Betreuer] Boche, and Thomas [Akademischer Betreuer] Strohmer. "Reconstruction and Processing of Bandlimited Signals Based on Their Discrete Values / Ullrich Johann Mönich. Gutachter: Holger Boche ; Thomas Strohmer. Betreuer: Holger Boche." München : Universitätsbibliothek der TU München, 2011. http://d-nb.info/1016034962/34.
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Ganesh, Murthy C. N. S. "A Study On Bandpassed Speech From The Point Of Intelligibility." Thesis, Indian Institute of Science, 1989. https://etd.iisc.ac.in/handle/2005/93.
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Speech has been the subject of interest for a very long time. Even with so much advancement in the processing techniques and in the understanding of the source of speech, it is, even today, rather difficult to generate speech in the laboratory in all its aspects. A simple aspect like how the speech can retain its intelligibility even if it is distorted or band passed is not really understood. This thesis deals
with one small feature of speech viz., the intelligibility of speech is retained even when it is bandpassed with a minimum bandwidth of around 1 KHz located any where on the speech
spectrum of 0-4 KHz.
Several experiments have been conducted by the earlier workers by passing speech through various distortors like differentiators, integrators and infinite peak clippers and it is found that the intelligibility is retained to a very large extent in the distorted speech. The integrator and the differentiator remove essentially a certain portion of the spectrum. Therefore, it is thought that the intelligibility of the speech is spread over the entire speech spectrum and that, the intelligibility of speech may not be impaired even when it is bandpassed with a minimum bandwidth and the band may be located any where in the speech spectrum. To test this idea and establish this feature if it exists, preliminary
experiments have been conducted by passing the speech through different filters and it is found that the conjecture seems to be on the right line.
To carry out systematic experiments on this an
experimental set up has been designed and fabricated which consists of a microprocessor controlled speech recording, storing and speech playback system. Also, a personal computer
is coupled to the microprocessor system to enable the storage and processing of the data. Thirty persons drawn from different walks of life like teachers, mechanics and students have been involved for collecting the samples and for
recognition of the information of the processed speech. Even though the sentences like 'This is devices lab' are used to ascertain the effect of bandwidth on the intelligibility, for the purpose of analysis, vowels are used as the speech samples.
The experiments essentially consist of recording words and sentences spoken by the 30 participants and these recorded speech samples are passed through different filters with different bandwidths and central frequencies. The filtered output is played back to the various listeners and
observations regarding the intelligibility of the speech are noted. The listeners do not have any prior information about the content of the speech. It has been found that in almost
all (95%) cases, the messages or words are intelligible for most of the listeners when the band width of the filter is about 1 KHz and this is independent of the location of the pass band in the spectrum of 0-4 KHz. To understand how this feature of speech arises, spectrums of vowels spoken by 30 people have using FFT algorithms on the digitized samples of the speech.
It is felt that there is a cyclic behavior of the spectrum in all the samples. To make sure that the periodicity is present and also to arrive at the periodicity, a moving average procedure is employed to smoothen the spectrum. The smoothened spectrums of all the vowels indeed show a periodicity of about 1 KHz. When the periodicities are analysed the average value of the periodicities has been found to be 1038 Hz with a standard deviation of 19 Hz. In view of this it is thought that the acoustic source
responsible for speech must have generated this periodic spectrum, which might have been modified periodically to imprint the intelligibility. If this is true, one can perhaps easily understand this feature of the speech viz., the intelligibility is retained in a bandpassed speech of bandwidth 1 K H z . the pass band located any where in the speech spectrum of 0-4 KHz. This thesis describing the experiments and the analysis of the speech has been presented in 5 chapters. Chapter 1 deals with the basics of speech and the processing tools used to analyse the speech signal. Chapter 2 presents the literature survey from where the present problem is tracked down. Chapter 3 describes the details of the structure and the fabrication of the experimental setup that has been used. In chapter 4, the detailed account of the way in which the
experiments are conducted and the way in which the speech is analysed is given. In conclusion in chapter 5, the work is summarised and the future work needed to establish the mechanism of speech responsible for the feature of speech described in this thesis is suggested.
11
Ganesh, Murthy C. N. S. "A Study On Bandpassed Speech From The Point Of Intelligibility." Thesis, Indian Institute of Science, 1989. http://hdl.handle.net/2005/93.
Full textAbstract:
Speech has been the subject of interest for a very long time. Even with so much advancement in the processing techniques and in the understanding of the source of speech, it is, even today, rather difficult to generate speech in the laboratory in all its aspects. A simple aspect like how the speech can retain its intelligibility even if it is distorted or band passed is not really understood. This thesis deals
with one small feature of speech viz., the intelligibility of speech is retained even when it is bandpassed with a minimum bandwidth of around 1 KHz located any where on the speech
spectrum of 0-4 KHz.
Several experiments have been conducted by the earlier workers by passing speech through various distortors like differentiators, integrators and infinite peak clippers and it is found that the intelligibility is retained to a very large extent in the distorted speech. The integrator and the differentiator remove essentially a certain portion of the spectrum. Therefore, it is thought that the intelligibility of the speech is spread over the entire speech spectrum and that, the intelligibility of speech may not be impaired even when it is bandpassed with a minimum bandwidth and the band may be located any where in the speech spectrum. To test this idea and establish this feature if it exists, preliminary
experiments have been conducted by passing the speech through different filters and it is found that the conjecture seems to be on the right line.
To carry out systematic experiments on this an
experimental set up has been designed and fabricated which consists of a microprocessor controlled speech recording, storing and speech playback system. Also, a personal computer
is coupled to the microprocessor system to enable the storage and processing of the data. Thirty persons drawn from different walks of life like teachers, mechanics and students have been involved for collecting the samples and for
recognition of the information of the processed speech. Even though the sentences like 'This is devices lab' are used to ascertain the effect of bandwidth on the intelligibility, for the purpose of analysis, vowels are used as the speech samples.
The experiments essentially consist of recording words and sentences spoken by the 30 participants and these recorded speech samples are passed through different filters with different bandwidths and central frequencies. The filtered output is played back to the various listeners and
observations regarding the intelligibility of the speech are noted. The listeners do not have any prior information about the content of the speech. It has been found that in almost
all (95%) cases, the messages or words are intelligible for most of the listeners when the band width of the filter is about 1 KHz and this is independent of the location of the pass band in the spectrum of 0-4 KHz. To understand how this feature of speech arises, spectrums of vowels spoken by 30 people have using FFT algorithms on the digitized samples of the speech.
It is felt that there is a cyclic behavior of the spectrum in all the samples. To make sure that the periodicity is present and also to arrive at the periodicity, a moving average procedure is employed to smoothen the spectrum. The smoothened spectrums of all the vowels indeed show a periodicity of about 1 KHz. When the periodicities are analysed the average value of the periodicities has been found to be 1038 Hz with a standard deviation of 19 Hz. In view of this it is thought that the acoustic source
responsible for speech must have generated this periodic spectrum, which might have been modified periodically to imprint the intelligibility. If this is true, one can perhaps easily understand this feature of the speech viz., the intelligibility is retained in a bandpassed speech of bandwidth 1 K H z . the pass band located any where in the speech spectrum of 0-4 KHz. This thesis describing the experiments and the analysis of the speech has been presented in 5 chapters. Chapter 1 deals with the basics of speech and the processing tools used to analyse the speech signal. Chapter 2 presents the literature survey from where the present problem is tracked down. Chapter 3 describes the details of the structure and the fabrication of the experimental setup that has been used. In chapter 4, the detailed account of the way in which the
experiments are conducted and the way in which the speech is analysed is given. In conclusion in chapter 5, the work is summarised and the future work needed to establish the mechanism of speech responsible for the feature of speech described in this thesis is suggested.
12
Böhr, Frank [Verfasser]. "Model-Based Statistical Testing of Embedded Real-Time Software with Continuous and Discrete Signals in a Concurrent Environment: The Usage Net Approach / Frank Böhr." München : Verlag Dr. Hut, 2012. http://d-nb.info/1021072877/34.
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Fanganiello, Renato Dalto. "Sistema de comunicação utilizando sinais caóticos de tempo discreto em canais de banda limitada." Universidade Presbiteriana Mackenzie, 2010. http://tede.mackenzie.br/jspui/handle/tede/1392.
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Previous issue date: 2010-08-03Universidade Presbiteriana Mackenzie
Many synchronization methods for chaotic systems have been proposed over the past decades seeking for applications in communications. However, they have not satisfactorily performed when submitted to non-ideal communication channels. In this work, a discrete-time master-slave system based on Wu and Chua's synchronization model is considered. It is shown that it is possible to achieve synchronization under bandwidth limitations imposed by the communication channel by applying a digital filter in the feedback loop reponsible for the generation of the chaotic signal that will be transmitted. Furthermore, an analytical demonstration is presented showing that synchronization is not affected by the presence of digital filter introduced in the communication system. Results of computacional simulations relating the filter's parameters, such as its order and cut-off frequency, to maximum Lyapunov expoent of the transmitter system, which determines if the transmitted signal is chaotic or periodic, are also shown.
No decorrer das últimas décadas, muitos métodos de sincronismo para sistemas caóticos têm sido propostos visando aplicações em comunicações. No entanto, suas performances não têm se mostrado satisfatórias quando submetidos a condições não ideais de canais de transmissão. Neste trabalho considera-se um sistema mestre-escravo de tempo discreto baseado no modelo de sincronismo de Wu e Chua. Mostra-se que é possível atingir o sincronismo, mesmo quando há limitação de largura de banda imposta pelo canal de comunicação, por meio da introdução de um filtro digital que limita o espectro de potência na malha de realimentação responsável pela geração do sinal que será transmitido. Ademais, demonstra-se analiticamente que a presença do filtro digital introduzido no sistema de comunicação não interfere no seu sincronismo. Também são mostrados resultados de simulações computacionais que relacionam os parâmetros do filtro digital introduzido no sistema de comunicação, como ordem e frequência de corte, com o maior expoente de Lyapunov do sistema transmissor, que determina se o sinal transmitido será caótico ou periódico.
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Teixeira, Alex Fernandes Rocha. "Identificação de uma coluna de destilação de metanol-água através de modelos paramétricos e redes neurais artificiais." Universidade Federal de Alagoas, 2011. http://www.repositorio.ufal.br/handle/riufal/1195.
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This work presents a black box identification for a continuous methanol-water distillation column setting in open loop and closed loop response. Step changes and Pseudo-Random Binary Signal (PRBS) disturbance were used to excite the plant. The mathematical models candidates to identify were the Artificial Neural Networks (ANN) and the parametric models: ARX(autoregressive with exogenous inputs ), ARMAX (AutoRegressive Moving Average with eXogenous inputs ), OE(Output Error) and the Box-Jenkins (BJ)structure. The closed loop configuration was the R-V. The results showed that for the bottom loop, the best response were given by BJ, OE and RNA for both open and closed loop response. For the top closed loop, the best responses were also given by BJ, OE and RNA while in open loop condition, the RNA was the one that gave satisfactory outcome. It was verified that the pseudo-random binary signal was a good choice of excitation signal in identification for both open loop and closed dynamic systems.Foi realizado neste trabalho identificação caixa preta do processo de destilação Metanol-Água nas configurações malha aberta e malha fechada, utilizando como sinais de perturbação a função degrau e o Sinal Binário Pseudo-Aleatório (PRBS) para excitar a planta. Os modelos matemáticos candidatos a identificação foram as Redes Neurais Artificiais (RNA), e os modelos paramétricos discretos lineares autorregressivo com entradas externas (ARX do inglês AutoRegressive with eXogenous Inputs), autorregressivo com média móvel e entradas exógenas (ARMAX do inglês AutoRegressive Moving Average with eXogenous Inputs), modelo do tipo erro na saída (OE do inglês Output Error) e a estrutura Box-Jenkins (BJ). Com a disposição dos modelos, foram comparados quais dos modelos matemáticos candidatos à identificação melhor representa o processo coluna de destilação metanol-água. Comparou-se qual configuração do processo no ensaio de identificação para geração de dados apresenta mais vantagens, se em malha aberta ou em malha fechada, nas condições e metodologias utilizadas. Constatou-se a funcionalidade do sinal binário pseudo-aleatório como uma boa opção de excitação na identificação em malha aberta e fechada para sistemas dinâmicos.
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Messaoud, Safa. "Translating Discrete Time SIMULINK to SIGNAL." Thesis, Virginia Tech, 2014. http://hdl.handle.net/10919/49299.
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As Cyber Physical Systems (CPS) are getting more complex and safety critical, Model Based
Design (MBD), which consists of building formal models of a system in order to be used in
verification and correct-by-construction code generation, is becoming a promising methodology
for the development of the embedded software of such systems. This design paradigm
significantly reduces the development cost and time while guaranteeing better robustness,
capability and correctness with respect to the original specifications, when compared with
the traditional ad-hoc design methods. SIMULINK has been the most popular tool for embedded
control design in research as well as in industry, for the last decades. As SIMULINK
does not have formal semantics, the application of the model based design methodology and
tools to its models is very limited. In this thesis, we present a semantic translator that
transform discrete time SIMULINK models into SIGNAL programs. The choice of SIGNAL
is motivated by its polychronous formalism that enhances synchronous programming with
asynchronous concurrency, as well as, by the ability of its compiler of generating deterministic
multi thread code. Our translation involves three major steps: clock inference, type
inference and hierarchical top-down translation. We validate the semantic preservation of
our prototype tool by testing it on different SIMULINK models.Master of Science
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Schnell, Thomas. "Legibility optimization of uppercase alphanumeric text for displaying messages in traffic applications." Ohio : Ohio University, 1998. http://www.ohiolink.edu/etd/view.cgi?ohiou1175194520.
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Spinnato, Juliette. "Modèles de covariance pour l'analyse et la classification de signaux électroencéphalogrammes." Thesis, Aix-Marseille, 2015. http://www.theses.fr/2015AIXM4727/document.
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Cette thèse s’inscrit dans le contexte de l’analyse et de la classification de signaux électroencéphalogrammes (EEG) par des méthodes d’analyse discriminante. Ces signaux multi-capteurs qui sont, par nature, très fortement corrélés spatialement et temporellement sont considérés dans le plan temps-fréquence. En particulier, nous nous intéressons à des signaux de type potentiels évoqués qui sont bien représentés dans l’espace des ondelettes. Par la suite, nous considérons donc les signaux représentés par des coefficients multi-échelles et qui ont une structure matricielle électrodes × coefficients. Les signaux EEG sont considérés comme un mélange entre l’activité d’intérêt que l’on souhaite extraire et l’activité spontanée (ou "bruit de fond"), qui est largement prépondérante. La problématique principale est ici de distinguer des signaux issus de différentes conditions expérimentales (classes). Dans le cas binaire, nous nous focalisons sur l’approche probabiliste de l’analyse discriminante et des modèles de mélange gaussien sont considérés, décrivant dans chaque classe les signaux en termes de composantes fixes (moyenne) et aléatoires. Cette dernière, caractérisée par sa matrice de covariance, permet de modéliser différentes sources de variabilité. Essentielle à la mise en oeuvre de l’analyse discriminante, l’estimation de cette matrice (et de son inverse) peut être dégradée dans le cas de grandes dimensions et/ou de faibles échantillons d’apprentissage, cadre applicatif de cette thèse. Nous nous intéressons aux alternatives qui se basent sur la définition de modèle(s) de covariance(s) particulier(s) et qui permettent de réduire le nombre de paramètres à estimerThe present thesis finds itself within the framework of analyzing and classifying electroencephalogram signals (EEG) using discriminant analysis. Those multi-sensor signals which are, by nature, highly correlated spatially and temporally are considered, in this work, in the timefrequency domain. In particular, we focus on low-frequency evoked-related potential-type signals (ERPs) that are well described in the wavelet domain. Thereafter, we will consider signals represented by multi-scale coefficients and that have a matrix structure electrodes × coefficients. Moreover, EEG signals are seen as a mixture between the signal of interest that we want to extract and spontaneous activity (also called "background noise") which is overriding. The main problematic is here to distinguish signals from different experimental conditions (class). In the binary case, we focus on the probabilistic approach of the discriminant analysis and Gaussian mixtures are used, describing in each class the signals in terms of fixed (mean) and random components. The latter, characterized by its covariance matrix, allow to model different variability sources. The estimation of this matrix (and of its inverse) is essential for the implementation of the discriminant analysis and can be deteriorated by high-dimensional data and/or by small learning samples, which is the application framework of this thesis. We are interested in alternatives that are based on specific covariance model(s) and that allow to decrease the number of parameters to estimate
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Perera, W. A. "Design of discrete coefficient digital signal processors." Thesis, University of Cambridge, 1986. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.383313.
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Cheng, Siuling. "Signal reconstruction from discrete-time Wigner distribution." Thesis, Virginia Tech, 1985. http://hdl.handle.net/10919/41550.
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Wigner distribution is considered to be one of the most powerful tools for time-frequency analysis of rumvstationary signals. Wigner distribution is a bilinear signal transformation which provides two dimensional time-frequency characterization of one dimensional signals. Although much work has been done recently in signal analysis and applications using Wigner distribution, not many synthesis methods for Wigner distribution have been reported in the literature.
This thesis is concerned with signal synthesis from discrete-time Wigner distribution and from discrete-time pseudo-Wigner distribution and their applications in noise filtering and signal separation. Various algorithms are developed to reconstruct signals from the modified or specified Wigner distribution and pseudo-Wigner distribution which generally do not have a valid Wigner distributions or valid pseudo-Wigner distribution structures. These algorithms are successfully applied to the noise filtering and signal separation problems.
Master of Science
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Rivet, François. "Contribution à l’étude et à la réalisation d’un frontal radiofréquence analogique en temps discrets pour la radio-logicielle intégrale." Thesis, Bordeaux 1, 2009. http://www.theses.fr/2009BOR13811/document.
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Le concept de Radio Logicielle propose d’intégrer en un seul circuit un émetteur / récepteur RF capable d’émettre et de recevoir n’importe quel signal RF. Cependant, ce concept doit a?ronter des contraintes technologiques dans le cas des terminaux mobiles. La contrainte principale est la consommation de puissance du terminal. En e?et, la conversion analogique numérique qui est la clé de ce système en est aussi le principal verrou technique. Cette thèse présente une architecture de récepteur en rupture avec les architectures classiques a?n de surmonter le problème de la conversion analogique numérique. Il s’agit d’un processeur analogique de traitement du signal dédié à la Radio Logicielle intégrale dans la gamme de fréquence 0 à 5GHz. Sa conception et les mesures d’un prototype sont présentéesMany technological bottlenecks prevent from realizing a Software Radio (SR) mobile terminal. The old way of building radio architectures is over due to the numerous communication standards a single handeld terminal have to address nowadays. This thesis exposes a disruptive SR receiver: a Sampled Analog Signal Processor (SASP) is designed and brought into play to perform downconversion and channel presort. It processes analog voltage samples in order to recover in baseband any RF signal emitted from 0 to 5GHz. An analog Fast Fourier Transform achieves both frequency shifting and ?ltering. A prototype using 65nm CMOS technology from STMicroelectronics is here presented and measured
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Masud, Shahid. "VLSI systems for discrete wavelet transforms." Thesis, Queen's University Belfast, 1999. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.300782.
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Hu, Ta-Hsiang. "Discrete cosine transform implementation in VHDL." Thesis, Monterey, California : Naval Postgraduate School, 1990. http://handle.dtic.mil/100.2/ADA245791.
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Thesis (M.S. in Electrical Engineering)--Naval Postgraduate School, December 1990.Thesis Advisor(s): Lee, Chin-Hwa ; Yang, Chyan. "December 1990." Description based on title screen as viewed on March 29, 2010. DTIC Identifier(s): Fast Fourier Transform, High Level Languages, CHIPS (Electronics), Computerized Simulation, Signal Processing, Theses, Algorithms, Floating Point Operation, VHDL (Vhsic Hardware Description Language). Author(s) subject terms: FFT System, DCT System Implementation. Includes bibliographical references (p. 152). Also available in print.
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NiBouch, M. "Design and FPGA implementations for discrete wavelet transforms." Thesis, Queen's University Belfast, 2002. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.268365.
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Grassin, Stéphane. "Analyse temps-fréquence des signaux discrets." Rennes 1, 1997. http://www.theses.fr/1997REN10165.
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Cette these porte sur l'analyse temps-frequence des signaux et apporte quelques contributions sur la pratique du temps-frequence. Les nouveautes concernent principalement les representations temps-frequence des signaux discrets pour lesquelles nous proposons une formulation exacte que nous justifions par une demonstration basee sur les proprietes de l'impulsion de dirac. Quelques contributions sont apportees egalement a la caracterisation des rtf lorsque les calculs sont tronques pour des raisons d'implementation numerique. Le but de cette these est de realiser des outils d'analyse spectrale d'image avec des methodes du type temps-frequence et de les appliquer sur des images sar de la surface oceanique pour caracteriser des champs de houle et des trains d'ondes internes, qui constituent deux types de signaux non-stationnaires. Des representations du type temps-frequence adaptees aux images avaient deja ete proposees dans la litterature et ce travail de these poursuit l'etude, notamment par les methodes d'extension des noyaux qui peuvent etre envisagees. Les degres de libertes apportes par le doublement dimensionnel du noyau sont a l'origine de nouvelles proprietes specifiques qui n'avaient pas d'equivalent pour les representations temps-frequence des signaux mono-dimensionnels. Nous montrons comment ces proprietes et leurs contraintes associees permettent de reduire l'espace des solutions admissibles pour choisir les noyaux. Une de nos contributions les plus originales a propos de l'analyse spectrale des images et de proposer une nouvelle definition pour etendre la notion de signal analytique aux images. Cette nouvelle definition corrige certains defauts des definitions anterieures et nous indiquons le raisonnement qui y mene ainsi que les regles de calcul. Nous proposons egalement quelques exemples caracteristiques pour montrer les atouts de notre definition. L'application des methodes temps-frequences etendues aux images est finalement presentee sur des images sar dans le but d'en faire une analyse spectrale.
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Story, Mark Allan. "Mutiplierless decimation and commercial postfiltering of a discrete-time signal." Thesis, Massachusetts Institute of Technology, 1997. http://hdl.handle.net/1721.1/42674.
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Thesis (M. Eng.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 1997.Includes bibliographical references (leaf 105).
by Mark Allan Story.
M.Eng.
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Cena, Bernard Maria. "Reconstruction for visualisation of discrete data fields using wavelet signal processing." University of Western Australia. Dept. of Computer Science, 2000. http://theses.library.uwa.edu.au/adt-WU2003.0014.
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The reconstruction of a function and its derivative from a set of measured samples is a fundamental operation in visualisation. Multiresolution techniques, such as wavelet signal processing, are instrumental in improving the performance and algorithm design for data analysis, filtering and processing. This dissertation explores the possibilities of combining traditional multiresolution analysis and processing features of wavelets with the design of appropriate filters for reconstruction of sampled data. On the one hand, a multiresolution system allows data feature detection, analysis and filtering. Wavelets have already been proven successful in these tasks. On the other hand, a choice of discrete filter which converges to a continuous basis function under iteration permits efficient and accurate function representation by providing a “bridge” from the discrete to the continuous. A function representation method capable of both multiresolution analysis and accurate reconstruction of the underlying measured function would make a valuable tool for scientific visualisation. The aim of this dissertation is not to try to outperform existing filters designed specifically for reconstruction of sampled functions. The goal is to design a wavelet filter family which, while retaining properties necessary to preform multiresolution analysis, possesses features to enable the wavelets to be used as efficient and accurate “building blocks” for function representation. The application to visualisation is used as a means of practical demonstration of the results. Wavelet and visualisation filter design is analysed in the first part of this dissertation and a list of wavelet filter design criteria for visualisation is collated. Candidate wavelet filters are constructed based on a parameter space search of the BC-spline family and direct solution of equations describing filter properties. Further, a biorthogonal wavelet filter family is constructed based on point and average interpolating subdivision and using the lifting scheme. The main feature of these filters is their ability to reconstruct arbitrary degree piecewise polynomial functions and their derivatives using measured samples as direct input into a wavelet transform. The lifting scheme provides an intuitive, interval-adapted, time-domain filter and transform construction method. A generalised factorisation for arbitrary primal and dual order point and average interpolating filters is a result of the lifting construction. The proposed visualisation filter family is analysed quantitatively and qualitatively in the final part of the dissertation. Results from wavelet theory are used in the analysis which allow comparisons among wavelet filter families and between wavelets and filters designed specifically for reconstruction for visualisation. Lastly, the performance of the constructed wavelet filters is demonstrated in the visualisation context. One-dimensional signals are used to illustrate reconstruction performance of the wavelet filter family from noiseless and noisy samples in comparison to other wavelet filters and dedicated visualisation filters. The proposed wavelet filters converge to basis functions capable of reproducing functions that can be represented locally by arbitrary order piecewise polynomials. They are interpolating, smooth and provide asymptotically optimal reconstruction in the case when samples are used directly as wavelet coefficients. The reconstruction performance of the proposed wavelet filter family approaches that of continuous spatial domain filters designed specifically for reconstruction for visualisation. This is achieved in addition to retaining multiresolution analysis and processing properties of wavelets.
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Park, Shinwoong. "Reconfigurable Discrete-time Analog FIR filters for Wideband Analog Signal Processing." Diss., Virginia Tech, 2019. http://hdl.handle.net/10919/99794.
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Demand for data communication capacity is rapidly increasing with more and more number of users and higher bandwidth services. As a result, a critical research issue is the implementation of wideband and flexible signal processing in communication and sensing applications. Although software defined radio (SDR) is a possible solution, it may not be practical due to the excessive requirements for analog-to-digital converter (ADCs) and digital filters for wideband signals. In this environment, discrete-time (DT) domain circuits are gaining attention in various architectures such as N-path filters, sampling mixers, and analog FIR/IIR/FFT filters. DT analog signal processing (DT-ASP) ahead of an ADC considerably relaxes the ADC requirements by flexible filtering, offers the potential for higher dynamic range performance, and provides robustness in the presence of digital CMOS scaling.
The primary work presented in this dissertation is the design of wideband analog finite impulse response (AFIR) filters. Analog FIR filters have been used as low pass filters for out-of-band rejection in narrow-band applications. However, this work seeks to develop AFIR filters suitable for wideband applications, extending its possible applications. To achieve these performance goals, capacitive digital to analog converters (CDACs) have been introduced for the first time as wideband analog coefficient multipliers, which has led to high linearity analog multiplication with coefficient selection at the DAC resolution. A prototype 4th order DT FIR filter has been implemented in 32nm SOI CMOS technology and has achieved low-pass, band-pass, and high-pass filter (LPF, BPF and HPF) transfer functions corresponding to the programmed coefficient sets with IIP3>11dBm linearity and less than 2 mW/tap of power consumption. The AFIR filter is also utilized to demonstrate a proof-of-concept FIR-based beamforming. The beamforming network consisting of 4 antenna element inputs followed by AFIR filters was implemented with PCB modules with the previously fabricated AFIR filter chip. Behavioral simulations are used to verify the beamforming function with given coefficient sets. Based on the developed AFIR filter modules, FIR-based beamforming was demonstrated with measurement results matching well with the simulations.
Further work presented is the design and optimization of multi-section CDAC (MS-CDAC) structures. The proposed MS-CDAC approach provides wide range of options to optimize the tradeoff between kT/C noise, linearity versus switching energy, speed and area. When the optimization approach is applied to a proof-of-concept 10-bit CDAC design, the selected MS-CDAC structure reduces total capacitance and switching energy by 97% and 98%, respectively for given linearity and noise limitations. The proposed MS-CDAC structures are applicable in both DT-ASP coefficient multiplier and SAR-ADC applications.PHD
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Chiang, Tony. "Design and Evaluation of a Discrete Wavelet Transform Based Multi-Signal Receiver." Wright State University / OhioLINK, 2006. http://rave.ohiolink.edu/etdc/view?acc_num=wright1152297283.
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Бекус, Ростислав Володимирович, and Rostislav Bekus. "Метод підвищення якості передачі сигналів в бездротових локальних мережах." Master's thesis, Тернопільський національний технічний університет імені Івана Пулюя, 2020. http://elartu.tntu.edu.ua/handle/lib/34148.
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У кваліфікаційній роботі магістра розкрито поняття основ передачі інформації в бездротової середовищі. Проведено дослідження роботи сучасного стандарту WI-Fi 802.11ac. Встановлено, що найактуальнішою проблемою локальний бездротових мереж на сьогоднішній день, є захист переданих даних. Тому розглянуті основні схеми шифрування і модуляції сигналу, проведено порівняння продуктивності актуального стандарту в порівнянні з попереднім. Виявлено перевагу в швидкості передачі інформації і зони покриття при багатоканальної роботі.
Досліджено основні стандарти бездротового зв'язку 802.11. Була розглянута гіпотеза про шкоду здоров'ю від використання Wi-Fi.In the qualification work of the master the concept of bases of information transfer in the wireless environment is opened. A study of the modern standard WI-Fi 802.11ac. It is established that the most pressing problem of local wireless networks today is the protection of transmitted data. Therefore, the main schemes of signal encryption and modulation are considered, the performance of the current standard is compared with the previous one. The advantage in the speed of information transmission and coverage area during multichannel operation is revealed. The basic standards of 802.11 wireless communication are investigated. The hypothesis of harm from health from the use of Wi-Fi was considered.
ВСТУП 8 РОЗДІ 1. АНАЛІТИЧНА ЧАСТИНА 10 1.1 Основи передачі ніформації в безпровідних технологіях 10 1.2 Класифікація безпровдіних технологій 14 1.2.1 Бездротові персональні мережі 15 1.2.2 Бездротові локальні мережі 16 1.2.3 Бездротові мережі масштабу міста 17 1.2.4 Бездротові глобальні мережі топологія 18 1.3 Порівняльний аналіз найбільш актуальних стандартів безпровідного зв'язку 18 1.4 Висновки до розділу 1 19 РОЗДІЛ 2. ОСНОВНА ЧАСТИНА 20 2.1. Основні терміни та елементи мережі 20 2.2. Актуальні стандарти безпровідних локальних мереж Wi-Fi 20 2.3. Актуальні стандарти бездротових мереж 21 2.3.1 IEEE 802.11g 22 2.3.2 IEEE 802.11n 23 2.3.3 IEEE 802.11ac 24 2.4 Об'єднання технологій безпеки Wi-Fi 26 2.4.1 Історія розвитку 26 2.4.2 Механізм аутентифікації WPA2 29 2.4.3 Механізм шифрування WPA2 35 2.4.4 Wardriving 37 2.4.5 Сніфери 38 2.4.6 Сніфери 39 2.5 Технології Wi-Fi які пливають на здоров'я людини 40 2.6. Висновки до розділу 2 41 РОЗДІЛ 3. НАУКОВО-ДОСЛІДНА ЧАСТИНА 42 7 3.1. Порівняння продуктивності 802.11 n і 802.11 ac 42 3.2 Висновки до розділу 3 45 РОЗДІЛ 4. СПЕЦІАЛЬНА ЧАСТИНА 46 4.1. Область застосування програмного забезпечення Microsoft Office Visio 46 4.2. Загальні принципи програми Microsoft Office Visio 48 4.3. Висновки до розділу 4 51 РОЗДІЛ 5. ОХОРОНА ПРАЦІ ТА БЕЗПЕКА В НАДЗВИЧАЙНИХ СИТУАЦІЯХ 52 5.1. Охорона праці 52 5.2. Безпека в надзвичайних ситуаціях 54 5.3. Висновки до розділу 5 56 ЗАГАЛЬНІ ВИСНОВКИ 57 ПЕРЕЛІК ПОСИЛАНЬ 58 Додаток А. Копія тези конференції 60
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Sanchez, Fabrício Lopes. "Análise cepstral baseada em diferentes famílias transformada wavelet." Universidade de São Paulo, 2008. http://www.teses.usp.br/teses/disponiveis/82/82131/tde-01092010-113906/.
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Este trabalho apresenta um estudo comparativo entre diferentes famílias de transformada Wavelet aplicadas à análise cepstral de sinais digitais de fala humana, com o objetivo específico de determinar o período de pitch dos mesmos e, ao final, propõe um algoritmo diferencial para realizar tal operação, levando-se em consideração aspectos importantes do ponto de vista computacional, tais como: desempenho, complexidade do algoritmo, plataforma utilizada, dentre outros. São apresentados também, os resultados obtidos através da implementação da nova técnica (baseada na transformada wavelet) em comparação com a abordagem tradicional (baseada na transformada de Fourier). A implementação da técnica foi testada em linguagem C++ padrão ANSI sob as plataformas Windows XP Professional SP3, Windows Vista Business SP1, Mac OSX Leopard e Linux Mandriva 10.This work presents a comparative study between different family of wavelets applied on cepstral analysis of the digital speech human signal with specific objective for determining of pitch period of the same and in the end, proposes an differential algorithm to make such a difference operation take into consideration important aspects of computational point of view, such as: performance, algorithm complexity, used platform, among others. They are also present, the results obtained through of the technique implementation compared with the traditional approach. The technique implementation was tested in C++ language standard ANSI under the platform Windows XP Professional SP3 Edition, Windows Vista Business SP1, MacOSX Leopard and Linux Mandriva 10.
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Lo, King Chuen. "Theory and realization of novel algorithms for random sampling in digital signal processing." Thesis, Durham University, 1996. http://etheses.dur.ac.uk/5239/.
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Random sampling is a technique which overcomes the alias problem in regular sampling. The randomization, however, destroys the symmetry property of the transform kernel of the discrete Fourier transform. Hence, when transforming a randomly sampled sequence to its frequency spectrum, the Fast Fourier transform cannot be applied and the computational complexity is N(^2). The objectives of this research project are (1) To devise sampling methods for random sampling such that computation may be reduced while the anti-alias property of random sampling is maintained : Two methods of inserting limited regularities into the randomized sampling grids are proposed. They are parallel additive random sampling and hybrid additive random sampling, both of which can save at least 75% of the multiplications required. The algorithms also lend themselves to the implementation by a multiprocessor system, which will further enhance the speed of the evaluation. (2) To study the auto-correlation sequence of a randomly sampled sequence as an alternative means to confirm its anti-alias property : The anti-alias property of the two proposed methods can be confirmed by using convolution in the frequency domain. However, the same conclusion is also reached by analysing in the spatial domain the auto-correlation of such sample sequences. A technique to evaluate the auto-correlation sequence of a randomly sampled sequence with a regular step size is proposed. The technique may also serve as an algorithm to convert a randomly sampled sequence to a regularly spaced sequence having a desired Nyquist frequency. (3) To provide a rapid spectral estimation using a coarse kernel : The approximate method proposed by Mason in 1980, which trades the accuracy for the speed of the computation, is introduced for making random sampling more attractive. (4) To suggest possible applications for random and pseudo-random sampling : To fully exploit its advantages, random sampling has been adopted in measurement Random sampling is a technique which overcomes the alias problem in regular sampling. The randomization, however, destroys the symmetry property of the transform kernel of the discrete Fourier transform. Hence, when transforming a randomly sampled sequence to its frequency spectrum, the Fast Fourier transform cannot be applied and the computational complexity is N"^. The objectives of this research project are (1) To devise sampling methods for random sampling such that computation may be reduced while the anti-alias property of random sampling is maintained : Two methods of inserting limited regularities into the randomized sampling grids are proposed. They are parallel additive random sampling and hybrid additive random sampling, both of which can save at least 75% , of the multiplications required. The algorithms also lend themselves to the implementation by a multiprocessor system, which will further enhance the speed of the evaluation. (2) To study the auto-correlation sequence of a randomly sampled sequence as an alternative means to confirm its anti-alias property : The anti-alias property of the two proposed methods can be confirmed by using convolution in the frequency domain. However, the same conclusion is also reached by analysing in the spatial domain the auto-correlation of such sample sequences. A technique to evaluate the auto-correlation sequence of a randomly sampled sequence with a regular step size is proposed. The technique may also serve as an algorithm to convert a randomly sampled sequence to a regularly spaced sequence having a desired Nyquist frequency. (3) To provide a rapid spectral estimation using a coarse kernel : The approximate method proposed by Mason in 1980, which trades the accuracy for the speed of the computation, is introduced for making random sampling more attractive. (4) To suggest possible applications for random and pseudo-random sampling : To fully exploit its advantages, random sampling has been adopted in measurement instruments where computing a spectrum is either minimal or not required. Such applications in instrumentation are easily found in the literature. In this thesis, two applications in digital signal processing are introduced. (5) To suggest an inverse transformation for random sampling so as to complete a two-way process and to broaden its scope of application. Apart from the above, a case study of realizing in a transputer network the prime factor algorithm with regular sampling is given in Chapter 2 and a rough estimation of the signal-to-noise ratio for a spectrum obtained from random sampling is found in Chapter 3. Although random sampling is alias-free, problems in computational complexity and noise prevent it from being adopted widely in engineering applications. In the conclusions, the criteria for adopting random sampling are put forward and the directions for its development are discussed.
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Martucci, Stephen A. "Symmetric convolution and the discrete sine and cosine transforms : principles and applications." Diss., Georgia Institute of Technology, 1993. http://hdl.handle.net/1853/15038.
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Pavlišta, Libor. "Programové vybavení pro práci s mikroskopickým obrazem." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2012. http://www.nusl.cz/ntk/nusl-219633.
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This project informs about the basic image processing on grayscale and binary images. The project deals with analyzing the data obtained. Acquaints with terms such as morphology and image segmentation. Acquired knowledge is applied to create a program to complete the processing of microscopic images in LabView programming environment with subsequent analysis and plotting of results into image. For a better idea supporting data about operation in progress are continuous images.
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Shoubaki, Ehab Hamed. "UNIFIED LARGE AND SMALL SIGNAL DISCRETE-SPACE MODELING FOR PWM CONVERTERS IN CCM." Master's thesis, University of Central Florida, 2005. http://digital.library.ucf.edu/cdm/ref/collection/ETD/id/3961.
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In this Thesis a Unified Discrete State-Space Model for power converters in CCM is presented. Two main approaches to arriving at the discrete model are used. The first approach involves an impulse function approximation of the duty cycle modulations of the converter switches , and this approach results in a small signal discrete model. The Second approach is direct and does not involve any approximation of the modulations , this approach yields both a large signal nonlinear discrete model and a linear small signal model. Harmonic analysis of the converter states at steady-state is done for steady-state waveform acquisition , which increases the accuracy of the model especially for finding the control to inductor current frequency response. Finally the Discrete model is verified for the Half-Bridge DC/DC topology for its three main control schemes (Asymmetric , Symmetric , DCS). A GUI platform in MATLAB is presented as a wrapper that utilizes the models and analysis presented in this thesis.M.S.E.E.
Department of Electrical and Computer Engineering
Engineering and Computer Science
Electrical Engineering
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Forbes, Jason. "Discrete signal processing techniques for power converters : multi-carrier modulation and efficient filtering techniques." Thesis, University of British Columbia, 2013. http://hdl.handle.net/2429/45446.
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Digital control has become ubiquitous in the field of power electronics due to the ease of implementation, reusability, and flexibility. Practical engineers have been hesitant to use digital control rather than the more traditional analog control methods due to the unfamiliar theory, relatively complicated implementation and various challenges associated with digital quantization. This thesis presents discrete signal processing theory to solve issues in digitally controlled power converters including reference generation and filtering.
First, this thesis presents advancements made in the field of digital control of dc-ac and ac-dc power converters. First, a multi-carrier PWM strategy is proposed for the accurate and computationally inexpensive generation of sinusoidal signals. This method aims to reduce the cost of implementing a sine-wave generator by reducing both memory and computational requirements. The technique, backed by theoretical and experimental evidence, is simple to implement, and does not rely on any specialized hardware. The method was simulated and experimentally implemented in a voltage-controlled PWM inverter and can be extended to any application involving the digital generation of periodic signals.
The second advancement described in this thesis is the use of simple digital filters to improve the response time of single-phase active rectifiers. Under traditional analog control strategies, the bandwidth of an active rectifier is unduly restricted in order to reduce any unwanted harmonic distortion. This work investigates digital filters as a proposed means to improve the bandwidth, and thereby create a faster, more efficient ac-dc power converter. Finally, a moving average filter is proposed, due to its simple implementation and minor computational burden, as an efficient means to expand the bandwidth. Since moving average filters are well known and widely understood in industry, this proposed filter is an attractive solution for practicing engineers.
The theory developed in this thesis is verified through simulations and experiments.
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Murali, Swetha. "Design of Assistive Human-Machine-Interface control signal classifiers using the Discrete Cosine Transform." Available to subscribers only, 2008. http://proquest.umi.com/pqdweb?did=1674094141&sid=8&Fmt=2&clientId=1509&RQT=309&VName=PQD.
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Logette, Patrice. "Etude et réalisation d'un processeur acousto-optique numérique de traitement des signaux." Valenciennes, 1997. https://ged.uphf.fr/nuxeo/site/esupversions/bbfd31df-2499-46a6-843d-52f346b1db41.
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Un système hybride acousto-optique/numérique axe principalement sur le filtrage F. I. R. A été développé au laboratoire dans le cadre d'une thèse antérieure. L'objet du présent travail est, d'une part, d'améliorer le système existant et, d'autre part, de tester les aptitudes du système ainsi modifié à effectuer d'autres types de calcul. Nous commençons par un résumé des travaux relatifs à l'ancien système, afin de bien positionner le problème. Nous exposons ensuite la conception du nouveau système. Une première partie décrit les modifications des circuits électroniques, avec l'utilisation de circuits de logique programmable de marque Altera. Une seconde partie est dédiée à l'aspect commande. On y détaille le programme de pilotage du système, la création et l'utilisation de modules indépendants pour chaque type de calcul, ainsi que les utilitaires associés (simulation, génération d'algorithmes). Nous terminons par une présentation de quelques exemples de calculs (FIR, IIR, DFT, DCT, corrélation) et évaluons les performances de notre système pour chacun de ces types d’opérations. Le bilan est assez satisfaisant dans l'ensemble, bien que l'apport des circuits Altera ne se soit pas révélé à la hauteur de nos espérances. Le filtrage IIR est le moins performant et nécessiterait la recherche d'autres algorithmes. Cependant, pour être réellement opérationnel, il faudrait améliorer la partie acousto-optique, ou à moyen terme, passer au tout numérique. Nous pourrions, alors, disposer d'un système simple et pratique pour simuler, tester ou valider, sur maquette, des algorithmes ou des sous-systèmes développés au laboratoire dans divers domaines du traitement de signal.
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Giacometti, Romain. "Détection et localisation des signaux radar (systèmes passifs ou discrets)." Thesis, Brest, 2017. http://www.theses.fr/2017BRES0083.
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L’objectif de cette thèse est de développer de nouvelles solutions pour détecter et localiser des sources électromagnétiques radar au niveau d'une unique station de réception en exploitant les signaux directs et indirects reçus. Dans le cadre de notre étude, nous avons dans un premier temps développé une modélisation du signal reçu au niveau d'un récepteur en tenant compte des caractéristiques des émetteurs et de la zone environnante. L'évaluation de cette modélisation a été effectuée en s'appuyant sur un cas particulier de détection et de localisation des réflecteurs. Ce dernier, traité dans la littérature, repose sur l’exploitation des trajets multiples. Ces derniers peuvent être également utilisés pour localiser des sources d’émission. Néanmoins, la plupart des méthodes existantes se basent sur des réflexions dites spéculaires. Les techniques employant les réflexions non spéculaires sur un réflecteur quelconque pour localiser des sources d'émission dans un environnement inconnu font l'objet de peu de publications dans la littérature ouverte. La méthode de localisation que nous proposons a l'avantage de n'employer qu'un récepteur fixe mesurant seulement deux types de grandeurs : les angles d'arrivée (AOA) et les différences de temps d'arrivée (TDOA). En pratique, un problème d'affectation doit être résolu avant de procéder à la localisation des émetteurs et des réflecteurs. Le problème consiste à affecter chaque paire de mesures TDOAAOA à un réflecteur donné, en supposant que chaque paire a déjà été affectée à un émetteur.La méthode que nous avons développée a été testée et évaluée, d'une part grâce à des données simulées et d'autre part en utilisant des mesures réellesThe purpose of this work is to develop new methods for the detection and the location of radar sources. The developed approach exploits the direct and indirect signals received at the receiving point. In our study, we first develop a model of these signals that takes into account the characteristics of the transmitters and the reflectors. We evaluate this model by simulating a particular case of reflectors detection and location, defined in the literature. Our goal is to use the multipaths to locate emission sources. Most existing methods are based on specular reflections. Methods based on non-specular reflections, to locate emission sources in an unknown environment, are rarely studied in the literature. In our study, we propose a new location method that uses a fixed receiver measuring the Angle of Arrival (AOA) and Time Difference of Arrival (TDOA). In practice, an assignment problem must be solved before locating the emitters and reflectors. The problem is to assign each pair of TDOA-AOA measurements to a given reflector, assuming that each pair has already been assigned to a transmitter. The method developed has been tested and evaluated by using simulated data and real measurements
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Crouzy, Serge. "Méthodes d'analyse des signaux de Patch-Clamp à temps discret." Grenoble INPG, 1989. http://www.theses.fr/1989INPG0005.
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L'objectif consiste a appliquer a des signaux echantillonnes des methodes connues d'analyse des signaux de patch-clamp a temps continu et de developper de nouvelles analyses sur des signaux tres bruites
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Arichi, Maiko. "Direct Mathematical Method for Real-time Ischemic Detection from Electrocardiograms Using the Discrete Hermite Transform." University of Akron / OhioLINK, 2011. http://rave.ohiolink.edu/etdc/view?acc_num=akron1313529081.
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Bang, Jeong Hwan. "Adaptive radar detection in the presence of textured and discrete interference." Diss., Georgia Institute of Technology, 2013. http://hdl.handle.net/1853/49109.
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Under a number of practical operating scenarios, traditional moving target indicator (MTI) systems inadequately suppress ground clutter in airborne radar systems. Due to the moving platform, the clutter gains a nonzero relative velocity and spreads the power across Doppler frequencies. This obfuscates slow-moving targets of interest near the "direct current" component of the spectrum. In response, space-time adaptive processing (STAP) techniques have been developed that simultaneously operate in the space and time dimensions for effective clutter cancellation. STAP algorithms commonly operate under the assumption of homogeneous clutter, where the returns are described by complex, white Gaussian distributions. Empirical evidence shows that this assumption is invalid for many radar systems of interest, including high-resolution radar and radars operating at low grazing angles. We are interested in these heterogeneous cases, i.e., cases when the Gaussian model no longer suffices.
Hence, the development of reliable STAP algorithms for real systems depends on the accuracy of the heterogeneous clutter models. The clutter of interest in this work includes heterogeneous texture clutter and point clutter. We have developed a cell-based clutter model (CCM) that provides simple, yet faithful means to simulate clutter scenarios for algorithm testing. The scene generated by the CMM can be tuned with two parameters, essentially describing the spikiness of the clutter scene. In one extreme, the texture resembles point clutter, generating strong returns from localized range-azimuth bins. On the other hand, our model can also simulate a flat, homogeneous environment. We prove the importance of model-based STAP techniques, namely knowledge-aided parametric covariance estimation (KAPE), in filtering a gamut of heterogeneous texture scenes. We demonstrate that the efficacy of KAPE does not diminish in the presence of typical spiky clutter.
Computational complexities and susceptibility to modeling errors prohibit the use of KAPE in real systems. The computational complexity is a major concern, as the standard KAPE algorithm requires the inversion of an MNxMN matrix for each range bin, where M and N are the number of array elements and the number of pulses of the radar system, respectively. We developed a Gram Schmidt (GS) KAPE method that circumvents the need of a direct inversion and reduces the number of required power estimates. Another unavoidable concern is the performance degradations arising from uncalibrated array errors. This problem is exacerbated in KAPE, as it is a model-based technique; mismatched element amplitudes and phase errors amount to a modeling mismatch. We have developed the power-ridge aligning (PRA) calibration technique, a novel iterative gradient descent algorithm that outperforms current methods. We demonstrate the vast improvements attained using a combination of GS KAPE and PRA over the standard KAPE algorithm under various clutter scenarios in the presence of array errors.
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Krasňanský, Milan. "Radar Signal Processing for Radio Altimeter." Master's thesis, Vysoké učení technické v Brně. Fakulta informačních technologií, 2017. http://www.nusl.cz/ntk/nusl-363816.
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Táto diplomová práca sa zaoberá návrhom a implementáciou algoritmu pre spracovaniu signálu z radaru využívajúceho frekvenčne modulovanú kontinuálnu vlnu. Cieľom je implementácia algoritmu, ktorý by bol dostatočne rýchly (výpočet v reálnom čase na cieľovej platforme) a dostatočne presný pre použitie v rádiovýškomere v ľahkom lietadle so zameraním na použitie počas pristávacieho manévru. Hlavnou metódou spracovania signálu, použitou v implementácii, je Diskrétna Fourierova transformácia. Vytvorený algoritmus bol otestovaný na reálnych letových dátach a pre pristávací manéver dosiahol uspokojivé výsledky.
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Smecher, Graeme. "Discrete-time crossing-point estimation for switching power converters." Thesis, McGill University, 2008. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=115995.
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In a number of electrical engineering problems, so-called "crossing points" -- the instants at which two continuous-time signals cross each other -- are of interest. Often, particularly in applications using a Digital Signal Processor (DSP), only periodic samples along with a partial statistical characterization of the signals are available. In this situation, we are faced with the following problem: Given limited information about these signals, how can we efficiently and accurately estimate their crossing points?For example, an audio amplifier typically receives its input from a digital source decoded into regular samples (e.g. from MP3, DVD, or CD audio), or obtained from a continuous-time signal using an analog-to-digital converter (ADC). In a switching amplifier based on Pulse-Width Modulation (PWM) or Click Modulation (CM), a signal derived from the sampled audio is compared against a deterministic reference waveform; the crossing points of these signals control a switching power stage. Crossing-point estimates must be accurate in order to preserve audio quality. They must also be simple to calculate, in order to minimize processing requirements and delays.
We consider estimating the crossing points of a known function and a Gaussian random process, given uniformly-spaced, noisy samples of the random process for which the second-order statistics are assumed to be known. We derive the Maximum A-Posteriori (MAP) estimator, along with a Minimum Mean-Squared Error (MMSE) estimator which we show to be a computationally efficient approximation to the MAP estimator.
We also derive the Cramer-Rao bound (CRB) on estimator variance for the problem, which allows practical estimators to be evaluated against a best-case performance limit. We investigate several comparison estimators chosen from the literature. The structure of the MMSE estimator and comparison estimators is shown to be very similar, making the difference in computational expense between each technique largely dependent on the cost of evaluating various (generally non-linear) functions.
Simulations for both Pulse-Width and Click Modulation scenarios show the MMSE estimator performs very near to the Cramer-Rao bound and outperforms the alternative estimators selected from the literature.
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Požár, Pavel. "Měření zkreslení signálu EKG." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2012. http://www.nusl.cz/ntk/nusl-219531.
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This master's thesis deals with electrocardiogram and its distortion, which causes errors, due to which a doctor can make a wrong diagnosis. In the theoretical part electrocardiogram is briefly described, further theory of wavelet transform, introduction to the signal compression and some information about ECG delineation. Methods used for the signal evaluation and therefore determination of the distortion size are described in the main theoretical part. This thesis contains known methods and also a method of own design based on the ECG delineation. Some of the described methods are checked in the practical part. These methods are tested for two different compression algorithms, which cause various distortion. In the final part distortion was tested on the influence on setting the wrong diagnosis.
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Kallapur, Abhijit Aerospace Civil & Mechanical Engineering Australian Defence Force Academy UNSW. "A discrete-time robust extended kalman filter for estimation of nonlinear uncertain systems." Publisher:University of New South Wales - Australian Defence Force Academy. Information Technology & Electrical Engineering, 2009. http://handle.unsw.edu.au/1959.4/44095.
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This thesis provides a novel approach to the problem of state estimation for discrete-time nonlinear systems in the presence of large model uncertainties. Though classical nonlinear Kalman filters such as the extended Kalman filter (EKF) can handle uncertainties by increasing the value of noise covariances, this is only applicable to systems with small uncertainties. To this end, a discretetime robust extended Kalman filter (REKF) is formulated and applied to examples from the fields of aerospace engineering and signal processing with an emphasis on attitude estimation for small unmanned aerial vehicles (UAVs) and image processing under the influence of atmospheric turbulence. The robust filter is an approximate set-valued state estimator where the Riccati and filter equations are obtained as an approximate solution to a reverse-time optimal control problem defining the set-valued state estimator. The advantages of the REKF over the classical EKF are investigated for examples from the fields aerospace engineering and signal processing where large model uncertainties are introduced. In the case of small UAVs, an alternative attitude estimation algorithm based on the REKF is proposed in the event of gyroscopic failure and the inability of the vehicle to carry redundant sensors due to limited payload capabilities. In the case of image reconstruction under atmospheric turbulence, a robust pixel-wandering (random shifts) scheme is proposed to aid the process of image reconstruction. Also, problems pertaining to platform vibration analysis for aerospace vehicles and a frequency demodulation process in the presence of channel-induced uncertainties is also discussed.
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Aval, Jean-Christophe. "Combinatoire autour du groupe symétrique." Habilitation à diriger des recherches, Université Sciences et Technologies - Bordeaux I, 2013. http://tel.archives-ouvertes.fr/tel-00978093.
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Cette HDR présente mes travaux récents en combinatoire (énumérative et algébrique) autour du groupe symétrique, et répartis sur trois axes principaux : les co-quasi-invariants polynomiaux, les matrices à signes alternants et les tableaux boisés.
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Palmieri, Igor. "Modelagem de sinais neuronais utilizando filtros lineares de tempo discreto." Universidade de São Paulo, 2015. http://www.teses.usp.br/teses/disponiveis/3/3139/tde-11072016-162720/.
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A aquisição experimental de sinais neuronais é um dos principais avanços da neurociência. Por meio de observações da corrente e do potencial elétricos em uma região cerebral, é possível entender os processos fisiológicos envolvidos na geração do potencial de ação, e produzir modelos matemáticos capazes de simular o comportamento de uma célula neuronal. Uma prática comum nesse tipo de experimento é obter leituras a partir de um arranjo de eletrodos posicionado em um meio compartilhado por diversos neurônios, o que resulta em uma mistura de sinais neuronais em uma mesma série temporal. Este trabalho propõe um modelo linear de tempo discreto para o sinal produzido durante o disparo do neurônio. Os coeficientes desse modelo são calculados utilizando-se amostras reais dos sinais neuronais obtidas in vivo. O processo de modelagem concebido emprega técnicas de identificação de sistemas e processamento de sinais, e é dissociado de considerações sobre o funcionamento biofísico da célula, fornecendo uma alternativa de baixa complexidade para a modelagem do disparo neuronal. Além disso, a representação por meio de sistemas lineares permite idealizar um sistema inverso, cuja função é recuperar o sinal original de cada neurônio ativo em uma mistura extracelular. Nesse contexto, são discutidas algumas soluções baseadas em filtros adaptativos para a simulação do sistema inverso, introduzindo uma nova abordagem para o problema de separação de spikes neuronais.The experimental acquisition of neuronal signals is a major advance in neuroscience. Through observations of electric current and potential in a brain region, it is possible to understand the physiological processes involved in the action potential generation, and create mathematical models capable of simulating the behavior of the neuronal cell. A common practice in this kind of experiment is to obtain readings from an array of electrodes positioned in a medium shared by several neurons, which results in a mixture of neuronal signals in the same time series. This work proposes a discrete-time linear model of the neuronal signal during the firing of the cell. The coefficients of this model are estimated using real samples of the neuronal signals obtained in vivo. The conceived modeling process employs system identification and signal processing concepts, and is dissociated from any considerations about the biophysical function of the neuronal cell, providing a low-complexity alternative to model the neuronal spike. In addition, the use of a linear representation allows the idealization of an inverse system, whose main purpose is to recover the original signal of each active neuron in a given extracellular mixture. In this context, some solutions based on adaptive filters are discussed for the inverse model simulation, introducing a new approach to the problem of neuronal spike separation.
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Mio, Karine. "Etude de filtres et de processus non linéaires discrets." Grenoble INPG, 1996. http://www.theses.fr/1996INPG0118.
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Ce rapport concerne l'application de modeles non lineaires a des problemes classiques de traitement du signal: identification de filtres et modelisation de processus. Pour etudier ces modeles, nous introduisons d'abord les outils appropries, principalement les statistiques d'ordre superieur a deux et les algorithmes d'identification de parametres. Nous nous consacrons ensuite a l'etude de differents filtres non lineaires: filtres de hammerstein, de volterra, narmax, bilineaires et reseaux de neurones. Ces filtres sont compares dans le cas de la soustraction de bruit. Cette application necessite des hypotheses specifiques qui sont detaillees pour chaque modele. Ces travaux sont valides sur des signaux sonar reels. Enfin, nous nous interessons a la modelisation de processus non lineaires. Le principe est de caracteriser une serie temporelle comme le resultat du filtrage non lineaire d'un bruit blanc fictif. Nous nous sommes plus particulierement interesses aux processus bilineaires
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Rocha, Ryan D. "A Frequency-Domain Method for Active Acoustic Cancellation of Known Audio Sources." DigitalCommons@CalPoly, 2014. https://digitalcommons.calpoly.edu/theses/1240.
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Active noise control (ANC) is a real-time process in which a system measures an external, unwanted sound source and produces a canceling waveform. The cancellation is due to destructive interference by a perfect copy of the received signal phase-shifted by 180 degrees. Existing active noise control systems process the incoming and outgoing audio on a sample-by-sample basis, requiring a high-speed digital signal processor (DSP) and analog-to-digital converters (ADCs) with strict timing requirements on the order of tens of microseconds. These timing requirements determine the maximum sample rate and bit size as well as the maximum attenuation that the system can achieve. In traditional noise cancellation systems, the general assumption is that all unwanted sound is indeterminate. However, there are many instances in which an unwanted sound source is predictable, such as in the case of a song. This thesis presents a method for active acoustic cancellation of a known audio signal using the frequency characteristics of the known audio signal compared to that of a sampled, filtered excerpt of the same known audio signal.
In this procedure, we must first correctly locate the sample index for which a measured audio excerpt begins via the cross-correlation function. Next, we obtain the frequency characteristics of both the known source (WAVE file of the song) and the measured unwanted audio by taking the Fast Fourier Transform (FFT) of each signal, and calculate the effective environmental transfer function (degradation function) by taking the ratio of the two complex frequency-domain results. Finally, we attempt to recreate the environmental audio from the known data and produce an inverted, synchronized, and amplitude-matched signal to cancel the audio via destructive interference. Throughout the process, we employ many signal conditioning methods such as FIR filtering, median filtering, windowing, and deconvolution. We illustrate this frequency-domain method in Native Instruments’ LabVIEW running on the Windows operating system, and discuss its reliability, areas for improvement, and potential future applications in mobile technologies. We show that under ideal conditions (unwanted sound is a known white noise source, and microphone, loudspeaker, and environmental filter frequency responses are all perfectly flat), we can achieve a theoretical maximum attenuation of approximately 300 dB. If we replace the white noise source with an actual song and the environmental filter with a low-order linear filter, then we can achieve maximum attenuation in the range of 50-70 dB. However, in a real-world environment, with additional noise and imperfect microphones, speakers, synchronization, and amplitude-matching, we can expect to see attenuation values in the range of 10-20 dB.
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Rowley, Alexander. "Signal processing methods for cerebral autoregulation." Thesis, University of Oxford, 2008. http://ora.ox.ac.uk/objects/uuid:3d85ab53-9c9b-4b50-98f2-2e67848e5da4.
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Cerebral autoregulation describes the clinically observed phenomenon that cerebral blood flow remains relatively constant in healthy human subjects despite large systemic changes in blood pressure, dissolved blood gas concentrations, heart rate and other systemic variables. Cerebral autoregulation is known to be impaired post ischaemic stroke, after severe head injury, in patients suffering from autonomic dysfunction and under the action of various drugs. Cerebral auto-regulation is a dynamic, multivariate phenomenon. Sensitive techniques are required to monitor cerebral auto-regulation in a clinical setting. This thesis presents 4 related signal processing studies of cerebral autoregulation. The first study shows how consideration of changes in blood gas concentrations simultaneously with changes in blood pressure can improve the accuracy of an existing frequency domain technique for monitoring cerebral autoregulation from spontaneous fluctuations in blood pressure and a transcranial doppler measure of cerebral blood flow velocity. The second study shows how the continuous wavelet transform can be used to investigate coupling between blood pressure and near infrared spectroscopy measures of cerebral haemodynamics in patients with autonomic failure. This introduces time information into the frequency based assessment, however neglects the contribution of blood gas concentrations. The third study shows how this limitation can be resolved by introducing a new time-varying multivariate system identification algorithm based around the dual tree undecimated wavelet transform. All frequency and time-frequency domain methods of monitoring cerebral autoregulation assume linear coupling between the variables under consideration. The fourth study therefore considers nonlinear techniques of monitoring cerebral autoregulation, and illustrates some of the difficulties inherent in this form of analysis. The general approach taken in this thesis is to formulate a simple system model; usually in the form of an ODE or a stochastic process. The form of the model is adapted to encapsulate a hypothesis about features of cerebral autoregulation, particularly those features that may be difficult to recover using existing methods of analysis. The performance of the proposed method of analysis is then evaluated under these conditions. After this testing, the techniques are then applied to data provided by the Laboratory of Human Cerebrovascular Physiology in Alberta, Canada, and the National Hospital for Neurology and Neurosurgery in London, UK.