Methods of Design and Testing of Real Time Systems

Teollusuussovelluksissa vaaditaan nykyisin yhä useammin reaaliaikaista tiedon käsittelyä. Luotettavuus on yksi tärkeimmistä reaaliaikaiseen tiedonkäsittelyyn kykenevän järjestelmän ominaisuuksista. Sen saavuttamiseksi on sekä laitteisto, että ohjelmisto testattava. Tämän työn päätavoitteena on laitteiston testaaminen ja laitteiston testattavuus, koska luotettava laitteistoalusta on perusta tulevaisuuden reaaliaikajärjestelmille. Diplomityössä esitetään digitaaliseen signaalinkäsittelyyn soveltuvan prosessorikortin suunnittelu. Prosessorikortti on tarkoitettu sähkökoneiden ennakoivaa kunnonvalvontaa varten. Uusimmat DFT (Desing for Testability) menetelmät esitellään ja niitä sovelletaan prosessorikortin sunnittelussa yhdessä vanhempien menetelmien kanssa. Kokemukset ja huomiot menetelmien soveltuvuudesta raportoidaan työn lopussa. Työn tavoitteena on kehittää osakomponentti web -pohjaiseen valvontajärjestelmään, jota on kehitetty Sähkötekniikan osastolla Lappeenrannan teknillisellä korkeakoululla.

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.

A nearly optimum linear-phase digital FIR filters design

A new simple method to design linear-phase finite impulse response (FIR) digital filters, based on the steepest-descent optimization method, is presented in this paper. Starting from the specifications of the desired frequency response and a maximum approximation error a nearly optimum digital filter is obtained. Tests have shown that this method is alternative to other traditional ones such as Frequency Sampling and Parks-McClellan, mainly when other than brick wall frequency response is required as a desired frequency response. (C) 2011 Elsevier Inc. All rights reserved.

Fixed-point DSP implementation of nonlinear H(infinity) controller for large gap electromagnetic suspension system

Electromagnetic suspension systems are inherently nonlinear and often face hardware limitation when digitally controlled. The main contributions of this paper are: the design of a nonlinear H(infinity) controller. including dynamic weighting functions, applied to a large gap electromagnetic suspension system and the presentation of a procedure to implement this controller on a fixed-point DSP, through a methodology able to translate a floating-point algorithm into a fixed-point algorithm by using l(infinity) norm minimization due to conversion error. Experimental results are also presented, in which the performance of the nonlinear controller is evaluated specifically in the initial suspension phase. (C) 2009 Elsevier Ltd. All rights reserved.

Dynamic and isometric protocols of knee extension: Effect of fatigue on the EMG signal

The objective of this study was to analyze the electromyographic (EMG) signal behavior of rectus femoris (RF), vastus medialis (VM), vastus lateralis (VL) and biceps femoris (caput longum) (BFCL) from nine women during fatiguing dynamic and isometric knee extensions tests and to determine their EMGFT (Electromyographic Fatigue Threshold). Surface electrodes, biological signal acquisition module, analogical-digital converter board and specific software were used. The RMS (Root Mean Square) values obtained from concentric phase (80 to 30 degrees) of the dynamic knee extension andfrom isometric contraction were correlated with time on each load by linear regression analysis. The respective slopes were correlated with the correspondent load to determine the EMGFT. Force (Kgf) and median frequency - MF (Hz) obtained during MIVC (Maximal Isometric Voluntary Contraction) performed before and after the fatiguing tests were calculated in Matlab environment. The results demonstrated that the endurance time decreases with higher loads the EMG amplitude increase with time and was greater at higher loads, between muscles in dynamic exercise the RF and VL showed higher slopes, and in isometric exercise the VL showed the same behavior The EMGFT values were similar in both exercises; the force values predominantly decreased after fatiguing tests; however the MF only decreased after some loads. The protocols proposed allowed standardizing protocols at least to induce the fatigue process and to determine the EMGFT as an endurance indicative, which may be used to evaluate the effectiveness of rehabilitative or training interventions indicated to reduce muscle weakness and fatigue.

Discrete wavelet transform signal analyzer

This paper addresses the problem of processing biological data, such as cardiac beats in the audio and ultrasonic range, and on calculating wavelet coefficients in real time, with the processor clock running at a frequency of present application-specified integrated circuits and field programmable gate array. The parallel filter architecture for discrete wavelet transform (DWT) has been improved, calculating the wavelet coefficients in real time with hardware reduced up to 60%. The new architecture, which also processes inverse DWT, is implemented with the Radix-2 or the Booth-Wallace constant multipliers. One integrated circuit signal analyzer in the ultrasonic range, including series memory register banks, is presented. © 2007 IEEE.

Método de inversão matricial rápido GSGR para vectoring em sistemas xDSL, com implementação em chip DSP

Este trabalho apresenta um método rápido de inversão de matrizes densas, e uma possível aplicação com métodos de Vectoring, em pré-codificação e cancelamento de crosstalk de sistemas xDSL. A família de tecnologias xDSL utiliza os pares trançados de fios de cobre telefônicos como meio físico para transmitir dados digitais. O crosstalk é a principal causa de degradação de sinais na mais nova geração de sistemas xDSL, o G.fast, e para combatê-lo são utilizadas técnicas de pré-codificação e cancelamento, chamadas de Vectoring. O método proposto, chamado de GSGR, consiste em uma abordagem diferente para o método clássico de Squared Givens Rotations (SGR), adequado a implementações em plataformas embarcadas de processamento digital de sinais. Foram realizados testes comparativos do método GSGR com métodos diretos clássicos de inversão, utilizando uma plataforma digital multicore baseada no chip TI DSP TMS320C6670 e a plataforma de software Matlab. Os resultados dos testes de inversão de matrizes usando dados reais e dados simulados mostraram que o GSGR foi superior em velocidade de execução sem apresentar perdas significativas de acurácia para a aplicação em sistemas xDSL.

Development and simulation of DSP filters for transformer differential protection / by Oskar Reynisson.

Transformer protection is one of the most challenging applications within the power system protective relay field. Transformers with a capacity rating exceeding 10 MVA are usually protected using differential current relays. Transformers are an aging and vulnerable bottleneck in the present power grid; therefore, quick fault detection and corresponding transformer de-energization is the key element in minimizing transformer damage. Present differential current relays are based on digital signal processing (DSP). They combine DSP phasor estimation and protective-logic-based decision making. The limitations of existing DSP-based differential current relays must be identified to determine the best protection options for sensitive and quick fault detection. The development, implementation, and evaluation of a DSP differential current relay is detailed. The overall goal is to make fault detection faster without compromising secure and safe transformer operation. A detailed background on the DSP differential current relay is provided. Then different DSP phasor estimation filters are implemented and evaluated based on their ability to extract desired frequency components from the measured current signal quickly and accurately. The main focus of the phasor estimation evaluation is to identify the difference between using non-recursive and recursive filtering methods. Then the protective logic of the DSP differential current relay is implemented and required settings made in accordance with transformer application. Finally, the DSP differential current relay will be evaluated using available transformer models within the ATP simulation environment. Recursive filtering methods were found to have significant advantage over non-recursive filtering methods when evaluated individually and when applied in the DSP differential relay. Recursive filtering methods can be up to 50% faster than non-recursive methods, but can cause false trip due to overshoot if the only objective is speed. The relay sensitivity is however independent of filtering method and depends on the settings of the relay’s differential characteristics (pickup threshold and percent slope).

Is complex signal processing for bone conduction hearing aids useful?

OBJECTIVES To establish whether complex signal processing is beneficial for users of bone anchored hearing aids. METHODS Review and analysis of two studies from our own group, each comparing a speech processor with basic digital signal processing (either Baha Divino or Baha Intenso) and a processor with complex digital signal processing (either Baha BP100 or Baha BP110 power). The main differences between basic and complex signal processing are the number of audiologist accessible frequency channels and the availability and complexity of the directional multi-microphone noise reduction and loudness compression systems. RESULTS Both studies show a small, statistically non-significant improvement of speech understanding in quiet with the complex digital signal processing. The average improvement for speech in noise is +0.9 dB, if speech and noise are emitted both from the front of the listener. If noise is emitted from the rear and speech from the front of the listener, the advantage of the devices with complex digital signal processing as opposed to those with basic signal processing increases, on average, to +3.2 dB (range +2.3 … +5.1 dB, p ≤ 0.0032). DISCUSSION Complex digital signal processing does indeed improve speech understanding, especially in noise coming from the rear. This finding has been supported by another study, which has been published recently by a different research group. CONCLUSIONS When compared to basic digital signal processing, complex digital signal processing can increase speech understanding of users of bone anchored hearing aids. The benefit is most significant for speech understanding in noise.