Model-Based Troubleshooting of Digital Systems

This thesis describes a methodology, a representation, and an implemented program for troubleshooting digital circuit boards at roughly the level of expertise one might expect in a human novice. Existing methods for model-based troubleshooting have not scaled up to deal with complex circuits, in part because traditional circuit models do not explicitly represent aspects of the device that troubleshooters would consider important. For complex devices the model of the target device should be constructed with the goal of troubleshooting explicitly in mind. Given that methodology, the principal contributions of the thesis are ways of representing complex circuits to help make troubleshooting feasible. Temporally coarse behavior descriptions are a particularly powerful simplification. Instantiating this idea for the circuit domain produces a vocabulary for describing digital signals. The vocabulary has a level of temporal detail sufficient to make useful predictions abut the response of the circuit while it remains coarse enough to make those predictions computationally tractable. Other contributions are principles for using these representations. Although not embodied in a program, these principles are sufficiently concrete that models can be constructed manually from existing circuit descriptions such as schematics, part specifications, and state diagrams. One such principle is that if there are components with particularly likely failure modes or failure modes in which their behavior is drastically simplified, this knowledge should be incorporated into the model. Further contributions include the solution of technical problems resulting from the use of explicit temporal representations and design descriptions with tangled hierarchies.

Digital filtering of oscillations intrinsic to transmission line modeling based on lumped parameters

A correction procedure based on digital signal processing theory is proposed to smooth the numeric oscillations in electromagnetic transient simulation results from transmission line modeling based on an equivalent representation by lumped parameters. The proposed improvement to this well-known line representation is carried out with an Finite Impulse Response (FIR) digital filter used to exclude the high-frequency components associated with the spurious numeric oscillations. To prove the efficacy of this correction method, a well-established frequency-dependent line representation using state equations is modeled with an FIR filter included in the model. The results obtained from the state-space model with and without the FIR filtering are compared with the results simulated by a line model based on distributed parameters and inverse transforms. Finally, the line model integrated with the FIR filtering is also tested and validated based on simulations that include nonlinear and time-variable elements. © 2012 Elsevier Ltd. All rights reserved.

Synchronizing chaotic optically-programmable digital circuits

In this paper an approach to the synchronization of chaotic circuits has been reported. It is based on an optically programmable logic cell and the signals involved are fully digital. It is based on the reception of the same input signal on sender and receiver and from this approach, with a posterior correlation between both outputs, an identical chaotic output is obtained in both systems. No conversion from analog to digital signals is needed. The model here presented is based on a computer simulation.

Time evolution of frequency components In a chaotic digital signal

The type of signals obtained has conditioned chaos analysis tools. Almost in every case, they have analogue characteristics. But in certain cases, a chaotic digital signal is obtained and theses signals need a different approach than conventional analogue ones. The main objective of this paper will be to present some possible approaches to the study of this signals and how information about their characteristics may be obtained in the more straightforward possible way. We have obtained digital chaotic signals from an Optical Logic Cell with some feedback between output and one of the possible control gates. This chaos has been reported in several papers and its characteristics have been employed as a possible method to secure communications and as a way to encryption. In both cases, the influence of some perturbation in the transmission medium gave problems both for the synchronization of chaotic generators at emitter and receiver and for the recovering of information data. A proposed way to analyze the presence of some perturbation is to study the noise contents of transmitted signal and to implement a way to eliminate it. In our present case, the digital signal will be converted to a multilevel one by grouping bits in packets of 8 bits and applying conventional methods of time-frequency analysis to them. The results give information about the change in signals characteristics and hence some information about the noise or perturbations present. Equivalent representations to the phase and to the Feigenbaum diagrams for digital signals are employed in this case.

Speech Analysis using Modern Techniques of Nonlinear Dynamics

Medical fields requires fast, simple and noninvasive methods of diagnostic techniques. Several methods are available and possible because of the growth of technology that provides the necessary means of collecting and processing signals. The present thesis details the work done in the field of voice signals. New methods of analysis have been developed to understand the complexity of voice signals, such as nonlinear dynamics aiming at the exploration of voice signals dynamic nature. The purpose of this thesis is to characterize complexities of pathological voice from healthy signals and to differentiate stuttering signals from healthy signals. Efficiency of various acoustic as well as non linear time series methods are analysed. Three groups of samples are used, one from healthy individuals, subjects with vocal pathologies and stuttering subjects. Individual vowels/ and a continuous speech data for the utterance of the sentence "iruvarum changatimaranu" the meaning in English is "Both are good friends" from Malayalam language are recorded using a microphone . The recorded audio are converted to digital signals and are subjected to analysis.Acoustic perturbation methods like fundamental frequency (FO), jitter, shimmer, Zero Crossing Rate(ZCR) were carried out and non linear measures like maximum lyapunov exponent(Lamda max), correlation dimension (D2), Kolmogorov exponent(K2), and a new measure of entropy viz., Permutation entropy (PE) are evaluated for all three groups of the subjects. Permutation Entropy is a nonlinear complexity measure which can efficiently distinguish regular and complex nature of any signal and extract information about the change in dynamics of the process by indicating sudden change in its value. The results shows that nonlinear dynamical methods seem to be a suitable technique for voice signal analysis, due to the chaotic component of the human voice. Permutation entropy is well suited due to its sensitivity to uncertainties, since the pathologies are characterized by an increase in the signal complexity and unpredictability. Pathological groups have higher entropy values compared to the normal group. The stuttering signals have lower entropy values compared to the normal signals.PE is effective in charaterising the level of improvement after two weeks of speech therapy in the case of stuttering subjects. PE is also effective in characterizing the dynamical difference between healthy and pathological subjects. This suggests that PE can improve and complement the recent voice analysis methods available for clinicians. The work establishes the application of the simple, inexpensive and fast algorithm of PE for diagnosis in vocal disorders and stuttering subjects.

Uma ferramenta para Análise Multiresolução de dados não regularmente amostrados

Digital signal processing (DSP) aims to extract specific information from digital signals. Digital signals are, by definition, physical quantities represented by a sequence of discrete values and from these sequences it is possible to extract and analyze the desired information. The unevenly sampled data can not be properly analyzed using standard techniques of digital signal processing. This work aimed to adapt a technique of DSP, the multiresolution analysis, to analyze unevenly smapled data, to aid the studies in the CoRoT laboratory at UFRN. The process is based on re-indexing the wavelet transform to handle unevenly sampled data properly. The was efective presenting satisfactory results

MS2SV: Tradução de modelos em Simulink para VHDL-AMS

This paper presents a methodology and a tool for projects involving analogue and digital signals. A sub-systems group was developed to translation a Matlab/Simulink model in the correspondent structural model described in VHDL-AMS. The developed translation tool, named of MS(2)SV, can reads a file containing a Simulink model translating it in the correspondent VHDL-AMS structural code. The tool also creates the directories structure and necessary files to simulate the model translated in System Vision environment. Three models of D/A converters available commercially that use R-2R ladder network were studied. This work considers some of challenges set by the electronic industry for the further development of simulation methodologies and tools in the field of mixed-signal technology. Although the objective of the studies has been the D/A converter, the developed methodology has potentiality to be extended to consider control systems and mechatronic systems.

Desenvolvimento de experiências de acionamento e controle do movimento angular de motores de indução trifásico

This paper aims to show practical and effectiveexperiencesfor lessons Industrial Automation Laboratory taught inundergraduate degreein ElectricalEngineering from the University Júlio MesquitaFilho - UNESP, Guaratinguetá. Experiments carriedsimulatecontrol and drive systems of electric three phase induction motors (MIT)widely usedinindustries. The experiments simulate a manufacturing environment where there isa need to control the activation and continuous operation ofelectricmotors. Seven experimentsthat simulatethe firing of electrical motors through a controlsystem, a driver along with asimulator loads coupled to the electric motor was developed. Experiments usinga Programmable Logic Controller (PLC) as acontroller,an inverter frequencyasdriver, and MagneticBrake, as simulatorengine loads . The experiments were divided accordingto the speed reference signal used fordrivingand operating the electric motor: digital and analog. The first five experiments performing the drive control and operation of the electric motor via digital signals. The sixth and seventh experiments using an analog signal as a reference speed for the electric motor

Avaliação de parâmetros farmacêuticos em comprimidos matriciais empregando a técnica de Biosusceptometria AC

Oral administration is widely accepted route for drug delivery and solid dosage forms are commonly employed. The variation of absorption profiles along the human gastrointestinal tract (GIT) and the ability to target drugs by adequate dosage forms to distinct sites is the challenge in the pharmaceutical development of solid dosage forms. AC Biosusceptometry (ACB) is a technique that deserves consideration due to its features, accuracy of results and versatility. The purpose of this work was to evaluate, by employing the AC Biosusceptometer, the rate of swelling of systems matrices consisting of hydrophilic polymer (hydroxypropyl methyl cellulose) and magnetic material. Matrices tablets were evaluated in vitro to a more detailed analysis of kinetics of swelling, in addition to the study and application of mathematical models to correlate the magnetic area variation and the water uptake. All the procedures for qualitative and quantitative analysis of digital signals as well as the magnetic images processing were performed in MatLab® (Mathworks Inc.). ACB technique proved to be useful towards estimating the swelling properties of hydrophilic matrices in vitro, showing a promising capacity for further analyses involving dissolution test and in vivo studies, supporting their innovative potential pharmaceutical applications

Inserção e extração de marca d'água em imagens digitais usando a transformada wavelet

Pós-graduação em Engenharia Elétrica - FEIS

Sinais digitais de um durômetro portátil para a estimativa de propriedades de madeiras para dormentes

Pós-graduação em Agronomia (Energia na Agricultura) - FCA

Low Correlated Sequences & Architectures and Algorithms for Analog-to-Information Converters: Theory, Design, Implementation and Applications.

Most electronic systems can be described in a very simplified way as an assemblage of analog and digital components put all together in order to perform a certain function. Nowadays, there is an increasing tendency to reduce the analog components, and to replace them by operations performed in the digital domain. This tendency has led to the emergence of new electronic systems that are more flexible, cheaper and robust. However, no matter the amount of digital process implemented, there will be always an analog part to be sorted out and thus, the step of converting digital signals into analog signals and vice versa cannot be avoided. This conversion can be more or less complex depending on the characteristics of the signals. Thus, even if it is desirable to replace functions carried out by analog components by digital processes, it is equally important to do so in a way that simplifies the conversion from digital to analog signals and vice versa. In the present thesis, we have study strategies based on increasing the amount of processing in the digital domain in such a way that the implementation of analog hardware stages can be simplified. To this aim, we have proposed the use of very low quantized signals, i.e. 1-bit, for the acquisition and for the generation of particular classes of signals.

Correlación entre señales de audio multi-microfónicas

Este proyecto pretende mostrar los desfases existentes entre señales de audio obtenidas de la misma fuente en distintos puntos distanciados entre sí. Para ello nos basamos en el análisis de la correlación de las señales de audio multi-microfónicas, para determinar los retrasos entre dichas señales. Durante las de tres partes diferentes que conforman este proyecto, explicaremos el dónde, cómo y por qué se produce este efecto en este tipo de señales. En la primera se presentan algunos de los conceptos teóricos necesarios para entender el desarrollo posterior, tales como la coherencia y correlación entre señales, los retardos de fase y la importancia del micro-tiempo. Además se explican diversas técnicas microfónicas que se utilizarán en la tercera parte. A lo largo de la segunda, se presenta el software desarrollado para determinar y corregir el retraso entre las señales que se deseen analizar. Para ello se ha escogido la herramienta de programación Matlab, ya que ha sido la más utilizada en la mayoría de las asignaturas que componen la titulación y por ello se posee el suficiente dominio de la misma. Además de presentar el propio software, al final de esta parte hay un manual de usuario del mismo, en el que se explica el manejo para posibles usos futuros por parte de otras personas interesadas. En la última parte se demuestra en varios casos reales, el estudio de la alineación de tomas multi-microfónicas en las cuales se produce en efecto que se intenta detectar y corregir. Aquí se realizan tres estudios de dicho fenómeno. En el primero se emplean señales digitales internas, concretamente ruido blanco, retrasando algunas muestras dichas señales unas de otras, para luego analizarlas con el software desarrollado y comprobar la eficacia del mismo. En el segundo se analizan la señales de audio obtenidas en el estudio de grabación de varios grupos de música moderna, mostrando los resultados del empleo del software en algunas de ellas, tales como las tomas de batería, bajo y guitarra. En el tercero se analizan las señales de audio obtenidas fuera del estudio de grabación, en donde no se dispone de las supuestas condiciones ideales que se tienen en el entorno que rodea a un estudio de grabación (acústicamente hablando). Se utilizan algunas de las técnicas microfónicas explicadas en el último apartado de la parte dedicada a los conceptos teóricos, para la grabación de una orquesta sinfónica, para luego analizar el efecto buscado mediante nuestro software, presentando los resultados obtenidos. De igual manera se realiza en el estudio con una agrupación coral de cuatro voces dentro de una Iglesia. ABSTRACT This project aims to show delays between audio signals obtained from the same source at diferent points spaced apart. To do this we rely on the analysis of the correlation of multi-microphonic audio signals, to determine the delay between these signals. During three diferent parts that make up this project, we will explain where, how and why this effect occurs in this type of signals. At the first part we present some of the theoretical concepts necessary to understand the subsequent development, such as coherence and correlation between signals, phase delays and the importance of micro-time. Also explains several microphone techniques to be used in the third part. During the second, it presents the software developed to determine and correct the delay between the signals that are desired to analyze. For this we have chosen the programming software Matlab , as it has been the most used in the majority of the subjects in the degree and therefore has suficient command of it. Besides presenting the software at the end of this part there is a user manual of it , which explains the handling for future use by other interested people. The last part is shown in several real cases, the study of aligning multi- microphonic sockets in which it is produced in effect trying to detect and correct. This includes three studies of this phenomenon. In the first internal digital signals are used, basically white noise, delaying some samples the signals from each other, then with software developed analyzing and verifying its efectiveness. In the second analyzes the audio signals obtained in the recording studio several contemporary bands, showing the results of using the software in some of them, such as the taking of drums, bass and guitar. In the third analyzes audio signals obtained outside the recording studio, where there are no ideal conditions alleged to have on the environment surrounding a recording studio (acoustically speaking). We use some of the microphone techniques explained in the last paragraph of the section on theoretical concepts, for the recording of a symphony orchestra, and then analyze the effect sought by our software, presenting the results. Similarly, in the study performed with a four-voice choir in a church.