947 resultados para Advanced signal processing
Resumo:
El objetivo principal del presente proyecto es proporcionar al ingeniero de telecomunicaciones una visión general de las técnicas que se utilizan en el modelado del sistema auditivo. El modelado del sistema auditivo se realiza con los siguientes objetivos: a) Interpretar medidas directas, b)unificar el entendimiento de diferentes fenómenos, c) guiar estrategias de amplificación para suplir pérdidas auditivas y d) tener predicciones experimentalmente comprobables de comportamientos, con diferentes niveles de complejidad. En este trabajo se tratarán y explicarán brevemente las diferentes técnicas utilizadas para modelar las partes del sistema auditivo, desde las analogías electroacústicas, modelos biofísicos, binaurales, hasta la implementación de filtros auditivos mediante procesado de señal. Podemos concluir que el modelado mediante analogías electroacústicas permite una rápida implementación y entendimiento, pero tiene ciertas limitaciones. Las simulaciones mediante análisis numéricos son precisas y de gran utilidad tanto para del oído medio como para el interno. El procesado de señal es el procedimiento más completo y utilizado ya que permite modelar oído externo y medio además de permitir la implementación de filtros cocleares muy precisos y coherentes con la realidad incluyéndolos en modelos perceptivos. ABSTRACT. The main aim of the Project is to provide the Telecommunications Engineer an overview about the approaches for modelling the auditory system. The auditory system modelling is done for the next objectives: a) Interpret direct measures, b) Understand different phenomena c) get strategies of amplification for hearing impaired people and d) Obtain testable predictions experimentally about some behaviors with different complexity levels. Inside this document, several approaches about modeling of the auditory system parts will be explained: analog circuits, biophysics models, binaural models, and auditory filters made through signal processing. In conclusion, analog circuits are made quickly and they are easier to understand but they have many limitations. Simulations through numerical analysis are accurate and useful in middle and inner ear models. Signal processing is the more versatile approach because it lets to make a model of external and middle ear and then it allows to make complex auditory filters. Perceptive models can be made entirely through this method.
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We report conditions on a switching signal that guarantee that solutions of a switched linear systems converge asymptotically to zero. These conditions are apply to continuous, discrete-time and hybrid switched linear systems, both those having stable subsystems and mixtures of stable and unstable subsystems.
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Stochastic model updating must be considered for quantifying uncertainties inherently existing in real-world engineering structures. By this means the statistical properties,instead of deterministic values, of structural parameters can be sought indicating the parameter variability. However, the implementation of stochastic model updating is much more complicated than that of deterministic methods particularly in the aspects of theoretical complexity and low computational efficiency. This study attempts to propose a simple and cost-efficient method by decomposing a stochastic updating process into a series of deterministic ones with the aid of response surface models and Monte Carlo simulation. The response surface models are used as surrogates for original FE models in the interest of programming simplification, fast response computation and easy inverse optimization. Monte Carlo simulation is adopted for generating samples from the assumed or measured probability distributions of responses. Each sample corresponds to an individual deterministic inverse process predicting the deterministic values of parameters. Then the parameter means and variances can be statistically estimated based on all the parameter predictions by running all the samples. Meanwhile, the analysis of variance approach is employed for the evaluation of parameter variability significance. The proposed method has been demonstrated firstly on a numerical beam and then a set of nominally identical steel plates tested in the laboratory. It is found that compared with the existing stochastic model updating methods, the proposed method presents similar accuracy while its primary merits consist in its simple implementation and cost efficiency in response computation and inverse optimization.
Resumo:
Monte Carlo techniques, which require the generation of samples from some target density, are often the only alternative for performing Bayesian inference. Two classic sampling techniques to draw independent samples are the ratio of uniforms (RoU) and rejection sampling (RS). An efficient sampling algorithm is proposed combining the RoU and polar RS (i.e. RS inside a sector of a circle using polar coordinates). Its efficiency is shown in drawing samples from truncated Cauchy and Gaussian random variables, which have many important applications in signal processing and communications. RESUMEN. Método eficiente para generar algunas variables aleatorias de uso común en procesado de señal y comunicaciones (por ejemplo, Gaussianas o Cauchy truncadas) mediante la combinación de dos técnicas: "ratio of uniforms" y "rejection sampling".
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The constant development of digital systems in radio communications demands the adaptation of the current receiving equipment to the new technologies. In this context, a new Software Defined Radio based receiver is being implemented with the aim of carrying out different experiments to analyze the propagation of signals through the atmosphere from a satellite beacon. The receiver selected for this task is the PERSEUS SDR from the Italian company Microtelecom s.r.l. It is a software defined VLF-LF-MF-HF receiver based on an outstanding direct sampling digital architecture which features a 14 bit 80 MSamples/s analog-to-digital converter, a high-performance FPGA-based digital down-converter and a high-speed 480 Mbit/s USB2.0 PC interface. The main goal is to implement the related software and adapt the new receiver to the current working environment. In this paper, SDR technology guidelines are given and PERSEUS receiver digital signal processing is presented with the most remarkable results.
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Atrial fibrillation (AF) is a common heart disorder. One of the most prominent hypothesis about its initiation and maintenance considers multiple uncoordinated activation foci inside the atrium. However, the implicit assumption behind all the signal processing techniques used for AF, such as dominant frequency and organization analysis, is the existence of a single regular component in the observed signals. In this paper we take into account the existence of multiple foci, performing a spectral analysis to detect their number and frequencies. In order to obtain a cleaner signal on which the spectral analysis can be performed, we introduce sparsity-aware learning techniques to infer the spike trains corresponding to the activations. The good performance of the proposed algorithm is demonstrated both on synthetic and real data. RESUMEN. Algoritmo basado en técnicas de regresión dispersa para la extracción de las señales cardiacas en pacientes con fibrilación atrial (AF).
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Signal processing in any living being is much more complex than the one performed in artificial systems. Cortex architecture, although only partly known, gives some useful ideas to be employed in sensing technology. To analyze some of these structures is the objective of this paper. Among the points to be analyzed are the parallel transfer of information, the similarity of the different systems and the massive amount of data analyzed by physical techniques. As an example of these concepts, the possibility to transmit images in a parallel way will be reported.
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Computing the modal parameters of structural systems often requires processing data from multiple non-simultaneously recorded setups of sensors. These setups share some sensors in common, the so-called reference sensors, which are fixed for all measurements, while the other sensors change their position from one setup to the next. One possibility is to process the setups separately resulting in different modal parameter estimates for each setup. Then, the reference sensors are used to merge or glue the different parts of the mode shapes to obtain global mode shapes, while the natural frequencies and damping ratios are usually averaged. In this paper we present a new state space model that processes all setups at once. The result is that the global mode shapes are obtained automatically, and only a value for the natural frequency and damping ratio of each mode is estimated. We also investigate the estimation of this model using maximum likelihood and the Expectation Maximization algorithm, and apply this technique to simulated and measured data corresponding to different structures.
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This paper presents a description of our system for the Albayzin 2012 LRE competition. One of the main characteristics of this evaluation was the reduced number of available files for training the system, especially for the empty condition where no training data set was provided but only a development set. In addition, the whole database was created from online videos and around one third of the training data was labeled as noisy files. Our primary system was the fusion of three different i-vector based systems: one acoustic system based on MFCCs, a phonotactic system using trigrams of phone-posteriorgram counts, and another acoustic system based on RPLPs that improved robustness against noise. A contrastive system that included new features based on the glottal source was also presented. Official and postevaluation results for all the conditions using the proposed metrics for the evaluation and the Cavg metric are presented in the paper.
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The study of the Vertical-Cavity Semiconductor Optical Amplifiers (VCSOAs) for optical signal processing applications is increasing his interest. Due to their particular structure, the VCSOAs present some advantages when compared to their edge-emitting counterparts including low manufacturing costs, high coupling efficiency to optical fibers and the ease to fabricate 2-D arrays of this kind of devices. As a consequence, all-optical logic gates based on VCSOAs may be very promising devices for their use in optical computing and optical switching in communications. Moreover, since all the boolean logic functions can be implemented by combining NAND logic gates, the development of a Vertical-Cavity NAND gate would be of particular interest. In this paper, the characteristics of the dispersive optical bistability appearing on a VCSOA operated in reflection are studied. A progressive increment of the number of layers compounding the top Distributed Bragg Reflector (DBR) of the VCSOA results on a change on the shape of the appearing bistability from an S-shape to a clockwise bistable loop. This resulting clockwise bistability has high on-off contrast ratio and input power requirements one order of magnitude lower than those needed for edge-emitting devices. Based on these results, an all-optical vertical-cavity NAND gate with high on-off contrast ratio and an input power for operation of only 10|i\V will be reported in this paper.
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Non-parametric belief propagation (NBP) is a well-known message passing method for cooperative localization in wireless networks. However, due to the over-counting problem in the networks with loops, NBP’s convergence is not guaranteed, and its estimates are typically less accurate. One solution for this problem is non-parametric generalized belief propagation based on junction tree. However, this method is intractable in large-scale networks due to the high-complexity of the junction tree formation, and the high-dimensionality of the particles. Therefore, in this article, we propose the non-parametric generalized belief propagation based on pseudo-junction tree (NGBP-PJT). The main difference comparing with the standard method is the formation of pseudo-junction tree, which represents the approximated junction tree based on thin graph. In addition, in order to decrease the number of high-dimensional particles, we use more informative importance density function, and reduce the dimensionality of the messages. As by-product, we also propose NBP based on thin graph (NBP-TG), a cheaper variant of NBP, which runs on the same graph as NGBP-PJT. According to our simulation and experimental results, NGBP-PJT method outperforms NBP and NBP-TG in terms of accuracy, computational, and communication cost in reasonably sized networks.
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We extend in this paper some previous results concerning the differential-algebraic index of hybrid models of electrical and electronic circuits. Specifically, we present a comprehensive index characterization which holds without passivity requirements, in contrast to previous approaches, and which applies to nonlinear circuits composed of uncoupled, one-port devices. The index conditions, which are stated in terms of the forest structure of certain digraph minors, do not depend on the specific tree chosen in the formulation of the hybrid equations. Additionally, we show how to include memristors in hybrid circuit models; in this direction, we extend the index analysis to circuits including active memristors, which have been recently used in the design of nonlinear oscillators and chaotic circuits. We also discuss the extension of these results to circuits with controlled sources, making our framework of interest in the analysis of circuits with transistors, amplifiers, and other multiterminal devices.
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We apply diffusion strategies to propose a cooperative reinforcement learning algorithm, in which agents in a network communicate with their neighbors to improve predictions about their environment. The algorithm is suitable to learn off-policy even in large state spaces. We provide a mean-square-error performance analysis under constant step-sizes. The gain of cooperation in the form of more stability and less bias and variance in the prediction error, is illustrated in the context of a classical model. We show that the improvement in performance is especially significant when the behavior policy of the agents is different from the target policy under evaluation.
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This paper contributes with a unified formulation that merges previ- ous analysis on the prediction of the performance ( value function ) of certain sequence of actions ( policy ) when an agent operates a Markov decision process with large state-space. When the states are represented by features and the value function is linearly approxi- mated, our analysis reveals a new relationship between two common cost functions used to obtain the optimal approximation. In addition, this analysis allows us to propose an efficient adaptive algorithm that provides an unbiased linear estimate. The performance of the pro- posed algorithm is illustrated by simulation, showing competitive results when compared with the state-of-the-art solutions.
Resumo:
The aim of automatic pathological voice detection systems is to serve as tools, to medical specialists, for a more objective, less invasive and improved diagnosis of diseases. In this respect, the gold standard for those system include the usage of a optimized representation of the spectral envelope, either based on cepstral coefficients from the mel-scaled Fourier spectral envelope (Mel-Frequency Cepstral Coefficients) or from an all-pole estimation (Linear Prediction Coding Cepstral Coefficients) forcharacterization, and Gaussian Mixture Models for posterior classification. However, the study of recently proposed GMM-based classifiers as well as Nuisance mitigation techniques, such as those employed in speaker recognition, has not been widely considered inpathology detection labours. The present work aims at testing whether or not the employment of such speaker recognition tools might contribute to improve system performance in pathology detection systems, specifically in the automatic detection of Obstructive Sleep Apnea. The testing procedure employs an Obstructive Sleep Apnea database, in conjunction with GMM-based classifiers looking for a better performance. The results show that an improved performance might be obtained by using such approach.