Integration of embedded data processing algorithms inside PAMELA devices

PAMELA (Phased Array Monitoring for Enhanced Life Assessment) SHMTM System is an integrated embedded ultrasonic guided waves based system consisting of several electronic devices and one system manager controller. The data collected by all PAMELA devices in the system must be transmitted to the controller, who will be responsible for carrying out the advanced signal processing to obtain SHM maps. PAMELA devices consist of hardware based on a Virtex 5 FPGA with a PowerPC 440 running an embedded Linux distribution. Therefore, PAMELA devices, in addition to the capability of performing tests and transmitting the collected data to the controller, have the capability of perform local data processing or pre-processing (reduction, normalization, pattern recognition, feature extraction, etc.). Local data processing decreases the data traffic over the network and allows CPU load of the external computer to be reduced. Even it is possible that PAMELA devices are running autonomously performing scheduled tests, and only communicates with the controller in case of detection of structural damages or when programmed. Each PAMELA device integrates a software management application (SMA) that allows to the developer downloading his own algorithm code and adding the new data processing algorithm to the device. The development of the SMA is done in a virtual machine with an Ubuntu Linux distribution including all necessary software tools to perform the entire cycle of development. Eclipse IDE (Integrated Development Environment) is used to develop the SMA project and to write the code of each data processing algorithm. This paper presents the developed software architecture and describes the necessary steps to add new data processing algorithms to SMA in order to increase the processing capabilities of PAMELA devices.An example of basic damage index estimation using delay and sum algorithm is provided.

Nuevo algoritmo de compresión multimedia: codificación por saltos logaritmicos

Quizás el Código Morse, inventado en 1838 para su uso en la telegrafía, es uno de los primeros ejemplos de la utilización práctica de la compresión de datos [1], donde las letras más comunes del alfabeto son codificadas con códigos más cortos que las demás. A partir de 1940 y tras el desarrollo de la teoría de la información y la creación de los primeros ordenadores, la compresión de la información ha sido un reto constante y fundamental entre los campos de trabajo de investigadores de todo tipo. Cuanto mayor es nuestra comprensión sobre el significado de la información, mayor es nuestro éxito comprimiéndola. En el caso de la información multimedia, su naturaleza permite la compresión con pérdidas, alcanzando así cotas de compresión imposibles para los algoritmos sin pérdidas. Estos “recientes” algoritmos con pérdidas han estado mayoritariamente basados en transformación de la información al dominio de la frecuencia y en la eliminación de parte de la información en dicho dominio. Transformar al dominio de la frecuencia posee ventajas pero también involucra unos costes computacionales inevitables. Esta tesis presenta un nuevo algoritmo de compresión multimedia llamado “LHE” (Logarithmical Hopping Encoding) que no requiere transformación al dominio de la frecuencia, sino que trabaja en el dominio del espacio. Esto lo convierte en un algoritmo lineal de reducida complejidad computacional. Los resultados del algoritmo son prometedores, superando al estándar JPEG en calidad y velocidad. Para ello el algoritmo utiliza como base la respuesta fisiológica del ojo humano ante el estímulo luminoso. El ojo, al igual que el resto de los sentidos, responde al logaritmo de la señal de acuerdo a la ley de Weber. El algoritmo se compone de varias etapas. Una de ellas es la medición de la “Relevancia Perceptual”, una nueva métrica que nos va a permitir medir la relevancia que tiene la información en la mente del sujeto y en base a la misma, degradar en mayor o menor medida su contenido, a través de lo que he llamado “sub-muestreado elástico”. La etapa de sub-muestreado elástico constituye una nueva técnica sin precedentes en el tratamiento digital de imágenes. Permite tomar más o menos muestras en diferentes áreas de una imagen en función de su relevancia perceptual. En esta tesis se dan los primeros pasos para la elaboración de lo que puede llegar a ser un nuevo formato estándar de compresión multimedia (imagen, video y audio) libre de patentes y de alto rendimiento tanto en velocidad como en calidad. ABSTRACT The Morse code, invented in 1838 for use in telegraphy, is one of the first examples of the practical use of data compression [1], where the most common letters of the alphabet are coded shorter than the rest of codes. From 1940 and after the development of the theory of information and the creation of the first computers, compression of information has been a constant and fundamental challenge among any type of researchers. The greater our understanding of the meaning of information, the greater our success at compressing. In the case of multimedia information, its nature allows lossy compression, reaching impossible compression rates compared with lossless algorithms. These "recent" lossy algorithms have been mainly based on information transformation to frequency domain and elimination of some of the information in that domain. Transforming the frequency domain has advantages but also involves inevitable computational costs. This thesis introduces a new multimedia compression algorithm called "LHE" (logarithmical Hopping Encoding) that does not require transformation to frequency domain, but works in the space domain. This feature makes LHE a linear algorithm of reduced computational complexity. The results of the algorithm are promising, outperforming the JPEG standard in quality and speed. The basis of the algorithm is the physiological response of the human eye to the light stimulus. The eye, like other senses, responds to the logarithm of the signal according with Weber law. The algorithm consists of several stages. One is the measurement of "perceptual relevance," a new metric that will allow us to measure the relevance of information in the subject's mind and based on it; degrade accordingly their contents, through what I have called "elastic downsampling". Elastic downsampling stage is an unprecedented new technique in digital image processing. It lets take more or less samples in different areas of an image based on their perceptual relevance. This thesis introduces the first steps for the development of what may become a new standard multimedia compression format (image, video and audio) free of patents and high performance in both speed and quality.

Integración de RVC-HEVC y OpenHEVC sobre la plataforma MultiDSP C6678

Este proyecto se basa en la integración de funciones optimizadas de OpenHEVC en el códec Reconfigurable Video Coding (RVC) - High Efficiency Video Coding (HEVC). RVC es un framework capaz de generar automáticamente el código que implementa cualquier estándar de video mediante el uso de librerías. Estas librerías contienen la definición de bloques funcionales de los que se componen los distintos estándares de video a implementar. Sin embargo, como desventaja a la facilidad de creación de estándares utilizando este framework, las librerías que utiliza no se encuentran optimizadas. Por ello se pretende que el códec RVC-HEVC sea capaz de realizar llamadas a funciones optimizadas, que para el estudio éstas se encontrarán en la librería OpenHEVC. Por otro lado, estos codificadores de video se pueden encontrar implementados tanto en PCs como en sistemas embebidos. Los Digital Signal Processors (DSPs) son unas plataformas especializadas en el procesamiento digital, teniendo una alta velocidad en el cómputo de operaciones matemáticas. Por ello, para este proyecto se integrará RVC-HEVC con las llamadas a OpenHEVC en una plataforma DSP como la TMS320C6678. Una vez completa la integración se efectuan medidas de eficiencia para ver cómo las llamadas a funciones optimizadas mejoran la velocidad en la decodificación de imágenes. ABSTRACT. This project is based in the integration of optimized functions from OpenHEVC in the RVC-HEVC (Reconfigurable Video Coding- High Efficiency Video Coding) codec. RVC is a framework capable of generating automatically any type of video standard with the use of libraries. Inside these libraries there are the definitions of the functional blocks which make up the different standards, in which for the case of study will be the HEVC standard. Nevertheless, as a downside for the simplicity in producing standards with the RVC tool, these libraries are not optimized. Thus, one of the goals for the project will be to make the RVC-HEVC call optimized functions, in which in this case they will be inside the OpenHEVC library. On the other hand, these video encoders can be implemented both in PCs and embedded systems. The DSPs (Digital Signal Processors) are platforms specialized in digital processing, being able to compute mathematical operations in a short period of time. Consequently, for this project the integration of the RVC-HEVC with calls to the OpenHEVC library will be done in a DSP platform such as a TMS320C6678. Once completed the integration, performance measures will be carried out to evaluate the improvement in the decoding speed obtained when optimized functions are used by the RVC-HEVC.

Prototipado de sistemas multiprocesador en FPGA

Los procesadores tradicionales de un solo núcleo han tenido que enfrentarse a grandes desafíos para poder mejorar su rendimiento y eficiencia energética. Mientras tanto, el rápido avance de las tecnologías de fabricación ha permitido la implementación de varios procesadores en un solo chip, ofreciendo un alto rendimiento y eficiencia energética. Éstos son los llamados procesadores multinúcleo. El objetivo de este proyecto es realizar un sistema multiprocesador para el procesamiento digital de señales de radio. Este sistema multiprocesador puede ser implementado en una tarjeta de prototipado. Para ello se ha utilizado el softcore MB-Lite y el sistema operativo en tiempo real FreeRTOS. ABSTRACT. Traditional single-core processors have faced great challenges to improve their performance and energy efficiency. Meanwhile, rapid advancing fabrication technologies have enabled the implementation of several processors in a single chip, providing high performance and energy efficiency. These are called multi-core processors. The aim of this project is to perform a multiprocessor system for digital radio signal processing. This multiprocessor system can be implemented in a general purpose prototyping card using. To achieve this project, the MB-Lite softcore and the FreeRTOS real time operating system have been used.

Análisis, modelado e igualación no lineal de arrays paramétricos acústicos

La presente Tesis analiza y desarrolla metodología específica que permite la caracterización de sistemas de transmisión acústicos basados en el fenómeno del array paramétrico. Este tipo de estructuras es considerado como uno de los sistemas más representativos de la acústica no lineal con amplias posibilidades tecnológicas. Los arrays paramétricos aprovechan la no linealidad del medio aéreo para obtener en recepción señales en el margen sónico a partir de señales ultrasónicas en emisión. Por desgracia, este procedimiento implica que la señal transmitida y la recibida guardan una relación compleja, que incluye una fuerte ecualización así como una distorsión apreciable por el oyente. Este hecho reduce claramente la posibilidad de obtener sistemas acústicos de gran fidelidad. Hasta ahora, los esfuerzos tecnológicos dirigidos al diseño de sistemas comerciales han tratado de paliar esta falta de fidelidad mediante técnicas de preprocesado fuertemente dependientes de los modelos físicos teóricos. Estos están basados en la ecuación de propagación de onda no lineal. En esta Tesis se propone un nuevo enfoque: la obtención de una representación completa del sistema mediante series de Volterra que permita inferir un sistema de compensación computacionalmente ligero y fiable. La dificultad que entraña la correcta extracción de esta representación obliga a desarrollar una metodología completa de identificación adaptada a este tipo de estructuras. Así, a la hora de aplicar métodos de identificación se hace indispensable la determinación de ciertas características iniciales que favorezcan la parametrización del sistema. En esta Tesis se propone una metodología propia que extrae estas condiciones iniciales. Con estos datos, nos encontramos en disposición de plantear un sistema completo de identificación no lineal basado en señales pseudoaleatorias, que aumenta la fiabilidad de la descripción del sistema, posibilitando tanto la inferencia de la estructura basada en bloques subyacente, como el diseño de mecanismos de compensación adecuados. A su vez, en este escenario concreto en el que intervienen procesos de modulación, factores como el punto de trabajo o las características físicas del transductor, hacen inviables los algoritmos de caracterización habituales. Incluyendo el método de identificación propuesto. Con el fin de eliminar esta problemática se propone una serie de nuevos algoritmos de corrección que permiten la aplicación de la caracterización. Las capacidades de estos nuevos algoritmos se pondrán a prueba sobre un prototipo físico, diseñado a tal efecto. Para ello, se propondrán la metodología y los mecanismos de instrumentación necesarios para llevar a cabo el diseño, la identificación del sistema y su posible corrección, todo ello mediante técnicas de procesado digital previas al sistema de transducción. Los algoritmos se evaluarán en términos de error de modelado a partir de la señal de salida del sistema real frente a la salida sintetizada a partir del modelo estimado. Esta estrategia asegura la posibilidad de aplicar técnicas de compensación ya que éstas son sensibles a errores de estima en módulo y fase. La calidad del sistema final se evaluará en términos de fase, coloración y distorsión no lineal mediante un test propuesto a lo largo de este discurso, como paso previo a una futura evaluación subjetiva. ABSTRACT This Thesis presents a specific methodology for the characterization of acoustic transmission systems based on the parametric array phenomenon. These structures are well-known representatives of the nonlinear acoustics field and display large technological opportunities. Parametric arrays exploit the nonlinear behavior of air to obtain sonic signals at the receptors’side, which were generated within the ultrasonic range. The underlying physical process redunds in a complex relationship between the transmitted and received signals. This includes both a strong equalization and an appreciable distortion for a human listener. High fidelity, acoustic equipment based on this phenomenon is therefore difficult to design. Until recently, efforts devoted to this enterprise have focused in fidelity enhancement based on physically-informed, pre-processing schemes. These derive directly from the nonlinear form of the wave equation. However, online limited enhancement has been achieved. In this Thesis we propose a novel approach: the evaluation of a complete representation of the system through its projection onto the Volterra series, which allows the posterior inference of a computationally light and reliable compensation scheme. The main difficulty in the derivation of such representation strives from the need of a complete identification methodology, suitable for this particular type of structures. As an example, whenever identification techniques are involved, we require preliminary estimates on certain parameters that contribute to the correct parameterization of the system. In this Thesis we propose a methodology to derive such initial values from simple measures. Once these information is made available, a complete identification scheme is required for nonlinear systems based on pseudorandom signals. These contribute to the robustness and fidelity of the resulting model, and facilitate both the inference of the underlying structure, which we subdivide into a simple block-oriented construction, and the design of the corresponding compensation structure. In a scenario such as this where frequency modulations occur, one must control exogenous factors such as devices’ operation point and the physical properties of the transducer. These may conflict with the principia behind the standard identification procedures, as it is the case. With this idea in mind, the Thesis includes a series of novel correction algorithms that facilitate the application of the characterization results onto the system compensation. The proposed algorithms are tested on a prototype that was designed and built for this purpose. The methodology and instrumentation required for its design, the identification of the overall acoustic system and its correction are all based on signal processing techniques, focusing on the system front-end, i.e. prior to transduction. Results are evaluated in terms of input-output modelling error, considering a synthetic construction of the system. This criterion ensures that compensation techniques may actually be introduced, since these are highly sensible to estimation errors both on the envelope and the phase of the signals involved. Finally, the quality of the overall system will be evaluated in terms of phase, spectral color and nonlinear distortion; by means of a test protocol specifically devised for this Thesis, as a prior step for a future, subjective quality evaluation.

Cancelación activa del ruido utilizando el KIT TMS320C5515 EZDSP

El control, o cancelación activa de ruido, consiste en la atenuación del ruido presente en un entorno acústico mediante la emisión de una señal igual y en oposición de fase al ruido que se desea atenuar. La suma de ambas señales en el medio acústico produce una cancelación mutua, de forma que el nivel de ruido resultante es mucho menor al inicial. El funcionamiento de estos sistemas se basa en los principios de comportamiento de los fenómenos ondulatorios descubiertos por Augustin-Jean Fresnel, Christiaan Huygens y Thomas Young entre otros. Desde la década de 1930, se han desarrollado prototipos de sistemas de control activo de ruido, aunque estas primeras ideas eran irrealizables en la práctica o requerían de ajustes manuales cada poco tiempo que hacían inviable su uso. En la década de 1970, el investigador estadounidense Bernard Widrow desarrolla la teoría de procesado adaptativo de señales y el algoritmo de mínimos cuadrados LMS. De este modo, es posible implementar filtros digitales cuya respuesta se adapte de forma dinámica a las condiciones variables del entorno. Con la aparición de los procesadores digitales de señal en la década de 1980 y su evolución posterior, se abre la puerta para el desarrollo de sistemas de cancelación activa de ruido basados en procesado de señal digital adaptativo. Hoy en día, existen sistemas de control activo de ruido implementados en automóviles, aviones, auriculares o racks de equipamiento profesional. El control activo de ruido se basa en el algoritmo fxlms, una versión modificada del algoritmo LMS de filtrado adaptativo que permite compensar la respuesta acústica del entorno. De este modo, se puede filtrar una señal de referencia de ruido de forma dinámica para emitir la señal adecuada que produzca la cancelación. Como el espacio de cancelación acústica está limitado a unas dimensiones de la décima parte de la longitud de onda, sólo es viable la reducción de ruido en baja frecuencia. Generalmente se acepta que el límite está en torno a 500 Hz. En frecuencias medias y altas deben emplearse métodos pasivos de acondicionamiento y aislamiento, que ofrecen muy buenos resultados. Este proyecto tiene como objetivo el desarrollo de un sistema de cancelación activa de ruidos de carácter periódico, empleando para ello electrónica de consumo y un kit de desarrollo DSP basado en un procesador de muy bajo coste. Se han desarrollado una serie de módulos de código para el DSP escritos en lenguaje C, que realizan el procesado de señal adecuado a la referencia de ruido. Esta señal procesada, una vez emitida, produce la cancelación acústica. Empleando el código implementado, se han realizado pruebas que generan la señal de ruido que se desea eliminar dentro del propio DSP. Esta señal se emite mediante un altavoz que simula la fuente de ruido a cancelar, y mediante otro altavoz se emite una versión filtrada de la misma empleando el algoritmo fxlms. Se han realizado pruebas con distintas versiones del algoritmo, y se han obtenido atenuaciones de entre 20 y 35 dB medidas en márgenes de frecuencia estrechos alrededor de la frecuencia del generador, y de entre 8 y 15 dB medidas en banda ancha. ABSTRACT. Active noise control consists on attenuating the noise in an acoustic environment by emitting a signal equal but phase opposed to the undesired noise. The sum of both signals results in mutual cancellation, so that the residual noise is much lower than the original. The operation of these systems is based on the behavior principles of wave phenomena discovered by Augustin-Jean Fresnel, Christiaan Huygens and Thomas Young. Since the 1930’s, active noise control system prototypes have been developed, though these first ideas were practically unrealizable or required manual adjustments very often, therefore they were unusable. In the 1970’s, American researcher Bernard Widrow develops the adaptive signal processing theory and the Least Mean Squares algorithm (LMS). Thereby, implementing digital filters whose response adapts dynamically to the variable environment conditions, becomes possible. With the emergence of digital signal processors in the 1980’s and their later evolution, active noise cancellation systems based on adaptive signal processing are attained. Nowadays active noise control systems have been successfully implemented on automobiles, planes, headphones or racks for professional equipment. Active noise control is based on the fxlms algorithm, which is actually a modified version of the LMS adaptive filtering algorithm that allows compensation for the acoustic response of the environment. Therefore it is possible to dynamically filter a noise reference signal to obtain the appropriate cancelling signal. As the noise cancellation space is limited to approximately one tenth of the wavelength, noise attenuation is only viable for low frequencies. It is commonly accepted the limit of 500 Hz. For mid and high frequencies, conditioning and isolating passive techniques must be used, as they produce very good results. The objective of this project is to develop a noise cancellation system for periodic noise, by using consumer electronics and a DSP development kit based on a very-low-cost processor. Several C coded modules have been developed for the DSP, implementing the appropriate signal processing to the noise reference. This processed signal, once emitted, results in noise cancellation. The developed code has been tested by generating the undesired noise signal in the DSP. This signal is emitted through a speaker simulating the noise source to be removed, and another speaker emits an fxlms filtered version of the same signal. Several versions of the algorithm have been tested, obtaining attenuation levels around 20 – 35 dB measured in a tight bandwidth around the generator frequency, or around 8 – 15 dB measured in broadband.

Information processing in the human brain: magnetoencephalographic approach.

Rapid progress in effective methods to image brain functions has revolutionized neuroscience. It is now possible to study noninvasively in humans neural processes that were previously only accessible in experimental animals and in brain-injured patients. In this endeavor, positron emission tomography has been the leader, but the superconducting quantum interference device-based magnetoencephalography (MEG) is gaining a firm role, too. With the advent of instruments covering the whole scalp, MEG, typically with 5-mm spatial and 1-ms temporal resolution, allows neuroscientists to track cortical functions accurately in time and space. We present five representative examples of recent MEG studies in our laboratory that demonstrate the usefulness of whole-head magnetoencephalography in investigations of spatiotemporal dynamics of cortical signal processing.

Research in speech communication.

Advances in digital speech processing are now supporting application and deployment of a variety of speech technologies for human/machine communication. In fact, new businesses are rapidly forming about these technologies. But these capabilities are of little use unless society can afford them. Happily, explosive advances in microelectronics over the past two decades have assured affordable access to this sophistication as well as to the underlying computing technology. The research challenges in speech processing remain in the traditionally identified areas of recognition, synthesis, and coding. These three areas have typically been addressed individually, often with significant isolation among the efforts. But they are all facets of the same fundamental issue--how to represent and quantify the information in the speech signal. This implies deeper understanding of the physics of speech production, the constraints that the conventions of language impose, and the mechanism for information processing in the auditory system. In ongoing research, therefore, we seek more accurate models of speech generation, better computational formulations of language, and realistic perceptual guides for speech processing--along with ways to coalesce the fundamental issues of recognition, synthesis, and coding. Successful solution will yield the long-sought dictation machine, high-quality synthesis from text, and the ultimate in low bit-rate transmission of speech. It will also open the door to language-translating telephony, where the synthetic foreign translation can be in the voice of the originating talker.

Chemotactic signal integration in bacteria.

Chemotactic signaling in Escherichia coli involves transmission of both negative and positive signals. In order to examine mechanisms of signal processing, behavioral responses to dual inputs have been measured by using photoactivable "caged" compounds, computer video analysis, and chemoreceptor deletion mutants. Signaling from Tar and Tsr, two receptors that sense amino acids and pH, was studied. In a Tar deletion mutant the photoactivated release of protons, a Tsr repellent, and of serine, a Tsr attractant, in separate experiments at pH 7.0 resulted in tumbling (negative) or smooth-swimming (positive) responses in ca. 50 and 140 ms, respectively. Simultaneous photorelease of protons and serine resulted in a single tumbling or smooth-swimming response, depending on the relative amounts of the two effectors. In contrast, in wild-type E. coli, proton release at pH 7.0 resulted in a biphasic response that was attributed to Tsr-mediated tumbling followed by Tar-mediated smooth-swimming. In wild-type E. coli at more alkaline pH values the Tar-mediated signal was stronger than the Tsr signal, resulting in a strong smooth-swimming response preceded by a diminished tumbling response. These observations imply that (i) a single receptor time-averages the binding of different chemotactic ligands generating a single response; (ii) ligand binding to different receptors can result in a nonintegrated response with the tumbling response preceding the smooth-swimming response; (iii) however, chemotactic signals of different intensities derived from different receptors can also result in an apparently integrated response; and (iv) the different chemotactic responses to protons at neutral and alkaline pH may contribute to E. coli migration toward neutrality.

Metodologia para aquisição de sinal elétrico e tratamento de dados em reômetros servo controlados.

A ciência na qual se estuda a deformação de um fluido no qual é aplicada uma tensão de cisalhamento é conhecida como reologia e o equipamento utilizado para a realização dos ensaios é chamado de reômetro. Devido a impraticabilidade de uso de reômetros comerciais, diversos pesquisadores desenvolveram reômetros capazes de analisar suspensões de macropartículas, baseados nos mesmos princípios de funcionamento dos equipamentos já existentes. Em alguns casos, a medição do torque do motor é realizada pela aquisição da tensão, uma vez que esta é proporcional ao torque. Entretanto, para melhor compreensão do resultado e para evitar a possibilidade de conclusões precipitadas, vê-se necessária correta interpretação do sinal elétrico, precisando avaliar qual frequência do sinal é relevante para o ensaio e, também, qual a melhor taxa de amostragem. Além da aquisição, para que o ensaio reológico seja realizado com precisão, é indispensável ótimo controle da taxa ou tensão do motor e uma alternativa é a utilização de um servomotor e um servoconversor. No caso desse ser comercial é essencial saber configurá-lo. Para facilitar o usuário leigo, alguns pesquisadores desenvolveram softwares para controle do equipamento e análise dos dados. Assim, o presente trabalho tem como objetivo propor uma metodologia para compreender o sinal aquisitado de um reômetro servo controlado e desenvolvimento do software de análise para o tratamento dos dados obtidos a partir de ensaios reológicos. Verificou-se a melhor configuração do servocontrolador, a melhor taxa de amostragem, de no mínimo 20 amostras/segundo, e, também, desenvolveu-se um filtro digital passa-baixa do tipo FIR para remover a frequência indesejada. Além disso, foi desenvolvido um software utilizando uma rotina em Matlab e uma interface gráfica do usuário (Graphical User Interface - GUI), para o pós-processamento dos dados para auxiliar o usuário leigo no tratamento e interpretação do resultado, que se mostrou eficaz.

Modelagem de sinais neuronais utilizando filtros lineares de tempo discreto.

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.

Electroencephalogram based Causality Graph Analysis in Behavior Tasks of Parkinson’s Disease Patients

Electroencephalographic (EEG) signals of the human brains represent electrical activities for a number of channels recorded over a the scalp. The main purpose of this thesis is to investigate the interactions and causality of different parts of a brain using EEG signals recorded during a performance subjects of verbal fluency tasks. Subjects who have Parkinson's Disease (PD) have difficulties with mental tasks, such as switching between one behavior task and another. The behavior tasks include phonemic fluency, semantic fluency, category semantic fluency and reading fluency. This method uses verbal generation skills, activating different Broca's areas of the Brodmann's areas (BA44 and BA45). Advanced signal processing techniques are used in order to determine the activated frequency bands in the granger causality for verbal fluency tasks. The graph learning technique for channel strength is used to characterize the complex graph of Granger causality. Also, the support vector machine (SVM) method is used for training a classifier between two subjects with PD and two healthy controls. Neural data from the study was recorded at the Colorado Neurological Institute (CNI). The study reveals significant difference between PD subjects and healthy controls in terms of brain connectivities in the Broca's Area BA44 and BA45 corresponding to EEG electrodes. The results in this thesis also demonstrate the possibility to classify based on the flow of information and causality in the brain of verbal fluency tasks. These methods have the potential to be applied in the future to identify pathological information flow and causality of neurological diseases.

Classificação das amostras do ensaio de Baumann através do processamento digital de imagens.

O presente trabalho apresenta uma alternativa ao processo de classificação do defeito da segregação central em amostras de aço, utilizando as imagens digitais que são geradas durante o ensaio de Baumann. O algoritmo proposto tem como objetivo agregar as técnicas de processamento digital de imagens e o conhecimento dos especialistas sobre o defeito da segregação central, visando a classificação do defeito de referência. O algoritmo implementado inclui a identificação e a segmentação da linha segregada por meio da aplicação da transformada de Hough e limiar adaptativo. Adicionalmente, o algoritmo apresenta uma proposta para o mapeamento dos atributos da segregação central nos diferentes graus de severidade do defeito, em função dos critérios de continuidade e intensidade. O mapeamento foi realizado por meio da análise das características individuais, como comprimento, largura e área, dos elementos segmentados que compõem a linha segregada. A avaliação do desempenho do algoritmo foi realizada em dois momentos específicos, de acordo com sua fase de implementação. Para a realização da avaliação, foram analisadas 255 imagens de amostras reais, oriundas de duas usinas siderúrgicas, distribuídas nos diferentes graus de severidade. Os resultados da primeira fase de implementação mostram que a identificação da linha segregada apresenta acurácia de 93%. As classificações oriundas do mapeamento realizado para as classes de criticidade do defeito, na segunda fase de implementação, apresentam acurácia de 92% para o critério de continuidade e 68% para o critério de intensidade.

TextMess 2.0: las tecnologías del lenguaje humano ante los nuevos retos de la comunicación digital

El objetivo general de este proyecto se centra en el estudio, desarrollo y experimentación de diferentes técnicas y sistemas basados en Tecnologías del Lenguaje Humano (TLH) para el desarrollo de la próxima generación de sistemas de procesamiento inteligente de la información digital (modelado, recuperación, tratamiento, comprensión y descubrimiento) afrontando los actuales retos de la comunicación digital. En este nuevo escenario, los sistemas deben incorporar capacidades de razonamiento que descubrirán la subjetividad de la información en todos sus contextos (espacial, temporal y emocional) analizando las diferentes dimensiones de uso (multilingualidad, multimodalidad y registro).

Direct support and general support maintenance manual : translator, signal data CV-3098/G (NSN 5895-00-237-2472) for satellite communication terminal AN/FSC-78(V).

"June 1978."