Avaliação e implementação de métodos de estimação de tempo de atraso de sinais de ultra-som

Coordenação de Aperfeiçoamento de Pessoal de Nível Superior (CAPES)

Controle da posição da perna de pessoas hígidas utilizando um controlador PID

Coordenação de Aperfeiçoamento de Pessoal de Nível Superior (CAPES)

Estudo de controladores aplicados a inversores para UPS com operação ilhada

Coordenação de Aperfeiçoamento de Pessoal de Nível Superior (CAPES)

An integrated environment for filter design

Analog filters and direct digital filters are implemented using digital signal processing techniques. Specifically, Butterworth, Elliptic, and Chebyshev filters are implemented using the Motorola 56001 Digital Signal Processor by the integration of three software packages: MATLAB, C++, and Motorola's Application Development System. The integrated environment allows the novice user to design a filter automatically by specifying the filter order and critical frequencies, while permitting more experienced designers to take advantage of MATLAB's advanced design capabilities. This project bridges the gap between the theoretical results produced by MATLAB and the practicalities of implementing digital filters using the Motorola 56001 Digital Signal Processor. While these results are specific to the Motorola 56001 they may be extended to other digital signal processors. MATLAB handles the filter calculations, a C++ routine handles the conversion to assembly code, and the Motorola software compiles and transmits the code to the processor

Caracterización y optimización de un sistema de sonido profesional

El presente proyecto tiene como objeto caracterizar y optimizar un equipo de sonido profesional, entendiendo por “caracterizar” el determinar los atributos particulares de cada uno de los componentes integrados en el sistema, y entendiendo por “optimizar” el hallar la mejor manera de obtener una respuesta plana para todo el rango de frecuencias, libre de distorsión, y en la mayor área posible. El sistema de sonido utilizado pertenece a un grupo musical de directo, por lo que se instala y se configura en cada concierto en función de las características del recinto, sea cerrado o al aire libre. Con independencia de estas particularidades, el sistema completo se divide en dos formaciones, L y R (lado izquierdo y lado derecho del escenario), por lo que cada formación se compone de un procesador digital de la señal, cuatro etapas de amplificación, un sistema line array de ocho unidades, y un conjunto de ocho altavoces de subgraves. Para llevar a cabo el objetivo planteado, se ha dividido el proyecto en las fases que a continuación se describen. En primer lugar, se han realizado, en la cámara anecoica de la EUITT, las medidas que permiten obtener las características de cada uno de los elementos que componen el sistema. Estas medidas se han almacenado en formato ASCII. En segundo lugar, se ha diseñado una interfaz gráfica que permite, utilizando las medidas almacenadas, caracterizar tanto la respuesta individual de cada elemento de la cadena del sistema de sonido como la respuesta combinada de una unidad line array y una unidad de subgraves. La interfaz es interactiva, y tiene además la capacidad de entregar automáticamente los valores de configuración que permiten la optimización del conjunto. Esto es, obtener alineamiento en el rango de frecuencias compartido por ambas unidades. Las medidas realizadas en la cámara anecoica se han utilizado igualmente para modelar el sistema line array al completo y poder realizar simulaciones en campo libre utilizando programas de predicción acústica. Se ha experimentado con los valores de configuración que permiten el alineamiento de los elementos individuales y obtenidos a través de la interfaz desarrollada, para comprobar la validez de los mismos con la formación line array y subgraves al completo. Por otro lado, se han analizado los métodos de optimización de sistemas propuestos por profesionales reconocidos del medio con el objetivo de aplicarlos en un evento real. En la preparación y montaje del evento, se han aplicado los valores de configuración proporcionados por la interfaz, y se ha comprobado la validez de los mismos realizando medidas in situ según los criterios propuestos en los métodos de optimización estudiados. ABSTRACT. This project aims to characterize and optimize a professional sound system. Characterize must be understood as determining the particular attributes of each component integrated in the system; optimize must be understood as finding the best way to get a flat response for all the frequency range, distortion free, in the largest possible area. The sound system under test belongs to a live musical group, so it is setup and configured on each concert depending on the characteristics of the enclosure, whether it’s indoor or outdoor. Apart from these features, the whole system is divided into two clusters, L and R (left and right side of the stage), so that each one is provided with a digital signal processor, four amplification stages, an eight-units line array system, and a set of eight subwoofers . To accomplish the stated objective, the project has been divided into the steps described below. To begin with, measures have been realized in the anechoic chamber of EUITT, which make possible obtaining the characteristics of each of the elements of the system. These measures have been stored in ASCII format. Then, a graphical interface has been designed that allow, using the stored measurements and from graphics, to characterize both the individual response of each element of the string sound system and the combined response of the several elements. The interface is interactive, and also has the ability to automatically deliver the configuration settings that allow the whole optimization. That means to get alignment in the frequency range shared by a line array unit and a subwoofer unit. The measurements made in the anechoic chamber have also been used to model the complete line array system and to perform free-field simulations using acoustical prediction programs. Simulations have been done with the configuration settings that allow the individual elements alignment (provided by the graphical interface developed), in order to check their validity with the full line array and subwoofer systems. On the other hand, analysis about the optimization methods, proposed by renowned professionals of the field, has been made in order to apply them in a real concert. In the setup and assembly of the event, configuration settings provided by the interface have been applied. Their validity has been proved by making measures on-site according to the criteria set in the studied optimization methods.

Diseño e implementación de un controlador de velocidad PID multivariable para un robot con configuración de triciclo

La robótica móvil constituye un área de desarrollo y explotación de interés creciente. Existen ejemplos de robótica móvil de relevancia destacada en el ámbito industrial y se estima un fuerte crecimiento en el terreno de la robótica de servicios. En la arquitectura software de todos los robots móviles suelen aparecer con frecuencia componentes que tienen asignadas competencias de gobierno, navegación, percepción, etcétera, todos ellos de importancia destacada. Sin embargo, existe un elemento, difícilmente prescindible en este tipo de robots, el cual se encarga del control de velocidad del dispositivo en sus desplazamientos. En el presente proyecto se propone desarrollar un controlador PID basado en el modelo y otro no basado en el modelo. Dichos controladores deberán operar en un robot con configuración de triciclo disponible en el Departamento de Sistemas Informáticos y deberán por tanto ser programados en lenguaje C para ejecutar en el procesador digital de señal destinado para esa actividad en el mencionado robot (dsPIC33FJ128MC802). ABSTRACT Mobile robotics constitutes an area of development and exploitation of increasing interest. There are examples of mobile robotics of outstanding importance in industry and strong growth is expected in the field of service robotics. In the software architecture of all mobile robots usually appear components which have assigned competences of government, navigation, perceptionetc., all of them of major importance. However, there is an essential element in this type of robots, which takes care of the speed control. The present project aims to develop a model-based and other non-model-based PID controller. These controllers must operate in a robot with tricycle settings, available from the Department of Computing Systems, and should therefore be programmed in C language to run on the digital signal processor dedicated to that activity in the robot (dsPIC33FJ128MC802).

Efficiency and power density improvement of grid-connected hybrid renewable energy systems utilizing high frequency-based power converters

High efficiency of power converters placed between renewable energy sources and the utility grid is required to maximize the utilization of these sources. Power quality is another aspect that requires large passive elements (inductors, capacitors) to be placed between these sources and the grid. The main objective is to develop higher-level high frequency-based power converter system (HFPCS) that optimizes the use of hybrid renewable power injected into the power grid. The HFPCS provides high efficiency, reduced size of passive components, higher levels of power density realization, lower harmonic distortion, higher reliability, and lower cost. The dynamic modeling for each part in this system is developed, simulated and tested. The steady-state performance of the grid-connected hybrid power system with battery storage is analyzed. Various types of simulations were performed and a number of algorithms were developed and tested to verify the effectiveness of the power conversion topologies. A modified hysteresis-control strategy for the rectifier and the battery charging/discharging system was developed and implemented. A voltage oriented control (VOC) scheme was developed to control the energy injected into the grid. The developed HFPCS was compared experimentally with other currently available power converters. The developed HFPCS was employed inside a microgrid system infrastructure, connecting it to the power grid to verify its power transfer capabilities and grid connectivity. Grid connectivity tests verified these power transfer capabilities of the developed converter in addition to its ability of serving the load in a shared manner. In order to investigate the performance of the developed system, an experimental setup for the HF-based hybrid generation system was constructed. We designed a board containing a digital signal processor chip on which the developed control system was embedded. The board was fabricated and experimentally tested. The system's high precision requirements were verified. Each component of the system was built and tested separately, and then the whole system was connected and tested. The simulation and experimental results confirm the effectiveness of the developed converter system for grid-connected hybrid renewable energy systems as well as for hybrid electric vehicles and other industrial applications.

Special issue on design and programming of signal processors for multimedia communication:Special issue on design and programming of signal processors for multimedia communication

The technical challenges in the design and programming of signal processors for multimedia communication are discussed. The development of terminal equipment to meet such demand presents a significant technical challenge, considering that it is highly desirable that the equipment be cost effective, power efficient, versatile, and extensible for future upgrades. The main challenges in the design and programming of signal processors for multimedia communication are, general-purpose signal processor design, application-specific signal processor design, operating systems and programming support and application programming. The size of FFT is programmable so that it can be used for various OFDM-based communication systems, such as digital audio broadcasting (DAB), digital video broadcasting-terrestrial (DVB-T) and digital video broadcasting-handheld (DVB-H). The clustered architecture design and distributed ping-pong register files in the PAC DSP raise new challenges of code generation.

Advanced educational parallel DSP system based on TMS320C25 processors

This paper describes the design, application, and evaluation of a user friendly, flexible, scalable and inexpensive Advanced Educational Parallel (AdEPar) digital signal processing (DSP) system based on TMS320C25 digital processors to implement DSP algorithms. This system will be used in the DSP laboratory by graduate students to work on advanced topics such as developing parallel DSP algorithms. The graduating senior students who have gained some experience in DSP can also use the system. The DSP laboratory has proved to be a useful tool in the hands of the instructor to teach the mathematically oriented topics of DSP that are often difficult for students to grasp. The DSP laboratory with assigned projects has greatly improved the ability of the students to understand such complex topics as the fast Fourier transform algorithm, linear and circular convolution, the theory and design of infinite impulse response (IIR) and finite impulse response (FIR) filters. The user friendly PC software support of the AdEPar system makes it easy to develop DSP programs for students. This paper gives the architecture of the AdEPar DSP system. The communication between processors and the PC-DSP processor communication are explained. The parallel debugger kernels and the restrictions of the system are described. The programming in the AdEPar is explained, and two benchmarks (parallel FFT and DES) are presented to show the system performance.

Rapid Implementation and Optimisation of DSP Systems on SoPC Heterogeneous Platforms

The emergence of programmable logic devices as processing platforms for digital signal processing applications poses challenges concerning rapid implementation and high level optimization of algorithms on these platforms. This paper describes Abhainn, a rapid implementation methodology and toolsuite for translating an algorithmic expression of the system to a working implementation on a heterogeneous multiprocessor/field programmable gate array platform, or a standalone system on programmable chip solution. Two particular focuses for Abhainn are the automated but configurable realisation of inter-processor communuication fabrics, and the establishment of novel dedicated hardware component design methodologies allowing algorithm level transformation for system optimization. This paper outlines the approaches employed in both these particular instances.

Nonlinear behavioral modeling of mixed signal RF-digital I/O devices

Esta tese investiga a caracterização (e modelação) de dispositivos que realizam o interface entre os domínios digital e analógico, tal como os buffers de saída dos circuitos integrados (CI). Os terminais sem fios da atualidade estão a ser desenvolvidos tendo em vista o conceito de rádio-definido-por-software introduzido por Mitola. Idealmente esta arquitetura tira partido de poderosos processadores e estende a operação dos blocos digitais o mais próximo possível da antena. Neste sentido, não é de estranhar que haja uma crescente preocupação, no seio da comunidade científica, relativamente à caracterização dos blocos que fazem o interface entre os domínios analógico e digital, sendo os conversores digital-analógico e analógico-digital dois bons exemplos destes circuitos. Dentro dos circuitos digitais de alta velocidade, tais como as memórias Flash, um papel semelhante é desempenhado pelos buffers de saída. Estes realizam o interface entre o domínio digital (núcleo lógico) e o domínio analógico (encapsulamento dos CI e parasitas associados às linhas de transmissão), determinando a integridade do sinal transmitido. Por forma a acelerar a análise de integridade do sinal, aquando do projeto de um CI, é fundamental ter modelos que são simultaneamente eficientes (em termos computacionais) e precisos. Tipicamente a extração/validação dos modelos para buffers de saída é feita usando dados obtidos da simulação de um modelo detalhado (ao nível do transístor) ou a partir de resultados experimentais. A última abordagem não envolve problemas de propriedade intelectual; contudo é raramente mencionada na literatura referente à caracterização de buffers de saída. Neste sentido, esta tese de Doutoramento foca-se no desenvolvimento de uma nova configuração de medição para a caracterização e modelação de buffers de saída de alta velocidade, com a natural extensão aos dispositivos amplificadores comutados RF-CMOS. Tendo por base um procedimento experimental bem definido, um modelo estado-da-arte é extraído e validado. A configuração de medição desenvolvida aborda não apenas a integridade dos sinais de saída mas também do barramento de alimentação. Por forma a determinar a sensibilidade das quantias estimadas (tensão e corrente) aos erros presentes nas diversas variáveis associadas ao procedimento experimental, uma análise de incerteza é também apresentada.

Narrowband signal detection techniques in shallow ocean by acoustic vector sensor array

This paper presents the formulation and performance analysis of four techniques for detection of a narrowband acoustic source in a shallow range-independent ocean using an acoustic vector sensor (AVS) array. The array signal vector is not known due to the unknown location of the source. Hence all detectors are based on a generalized likelihood ratio test (GLRT) which involves estimation of the array signal vector. One non-parametric and three parametric (model-based) signal estimators are presented. It is shown that there is a strong correlation between the detector performance and the mean-square signal estimation error. Theoretical expressions for probability of false alarm and probability of detection are derived for all the detectors, and the theoretical predictions are compared with simulation results. It is shown that the detection performance of an AVS array with a certain number of sensors is equal to or slightly better than that of a conventional acoustic pressure sensor array with thrice as many sensors.

An Unbiased Risk Estimator for Multiplicative Noise - Application to 1-D Signal Denoising

The effect of multiplicative noise on a signal when compared with that of additive noise is very large. In this paper, we address the problem of suppressing multiplicative noise in one-dimensional signals. To deal with signals that are corrupted with multiplicative noise, we propose a denoising algorithm based on minimization of an unbiased estimator (MURE) of meansquare error (MSE). We derive an expression for an unbiased estimate of the MSE. The proposed denoising is carried out in wavelet domain (soft thresholding) by considering time-domain MURE. The parameters of thresholding function are obtained by minimizing the unbiased estimator MURE. We show that the parameters for optimal MURE are very close to the optimal parameters considering the oracle MSE. Experiments show that the SNR improvement for the proposed denoising algorithm is competitive with a state-of-the-art method.