Técnicas de procesado digital de señal aplicadas a acelerómetros

En este proyecto se estudian y analizan las diferentes técnicas de procesado digital de señal aplicadas a acelerómetros. Se hace uso de una tarjeta de prototipado, basada en DSP, para realizar las diferentes pruebas. El proyecto se basa, principalmente, en realizar filtrado digital en señales provenientes de un acelerómetro en concreto, el 1201F, cuyo campo de aplicación es básicamente la automoción. Una vez estudiadas la teoría de procesado y las características de los filtros, diseñamos una aplicación basándonos sobre todo en el entorno en el que se desarrollaría una aplicación de este tipo. A lo largo del diseño, se explican las diferentes fases: diseño por ordenador (Matlab), diseño de los filtros en el DSP (C), pruebas sobre el DSP sin el acelerómetro, calibración del acelerómetro, pruebas finales sobre el acelerómetro... Las herramientas utilizadas son: la plataforma Kit de evaluación 21-161N de Analog Devices (equipado con el entorno de desarrollo Visual DSP 4.5++), el acelerómetro 1201F, el sistema de calibración de acelerómetros CS-18-LF de Spektra y los programas software MATLAB 7.5 y CoolEditPRO 2.0. Se realizan únicamente filtros IIR de 2º orden, de todos los tipos (Butterworth, Chebyshev I y II y Elípticos). Realizamos filtros de banda estrecha, paso-banda y banda eliminada, de varios tipos, dentro del fondo de escala que permite el acelerómetro. Una vez realizadas todas las pruebas, tanto simulaciones como físicas, se seleccionan los filtros que presentan un mejor funcionamiento y se analizan para obtener conclusiones. Como se dispone de un entorno adecuado para ello, se combinan los filtros entre sí de varias maneras, para obtener filtros de mayor orden (estructura paralelo). De esta forma, a partir de filtros paso-banda, podemos obtener otras configuraciones que nos darán mayor flexibilidad. El objetivo de este proyecto no se basa sólo en obtener buenos resultados en el filtrado, sino también de aprovechar las facilidades del entorno y las herramientas de las que disponemos para realizar el diseño más eficiente posible. In this project, we study and analize digital signal processing in order to design an accelerometer-based application. We use a hardware card of evaluation, based on DSP, to make different tests. This project is based in design digital filters for an automotion application. The accelerometer type is 1201F. First, we study digital processing theory and main parameters of real filters, to make a design based on the application environment. Along the application, we comment all the different steps: computer design (Matlab), filter design on the DSP (C language), simulation test on the DSP without the accelerometer, accelerometer calibration, final tests on the accelerometer... Hardware and software tools used are: Kit of Evaluation 21-161-N, based on DSP, of Analog Devices (equiped with software development tool Visual DSP 4.5++), 1201-F accelerometer, CS-18-LF calibration system of SPEKTRA and software tools MATLAB 7.5 and CoolEditPRO 2.0. We only perform 2nd orden IIR filters, all-type : Butterworth, Chebyshev I and II and Ellyptics. We perform bandpass and stopband filters, with very narrow band, taking advantage of the accelerometer's full scale. Once all the evidence, both simulations and physical, are finished, filters having better performance and analyzed and selected to draw conclusions. As there is a suitable environment for it, the filters are combined together in different ways to obtain higher order filters (parallel structure). Thus, from band-pass filters, we can obtain many configurations that will give us greater flexibility. The purpose of this project is not only based on good results in filtering, but also to exploit the facilities of the environment and the available tools to make the most efficient design possible.

Explointing FPGA block memories for protected cryptographic implementations

Modern Field Programmable Gate Arrays (FPGAs) are power packed with features to facilitate designers. Availability of features like huge block memory (BRAM), Digital Signal Processing (DSP) cores, embedded CPU makes the design strategy of FPGAs quite different from ASICs. FPGA are also widely used in security-critical application where protection against known attacks is of prime importance. We focus ourselves on physical attacks which target physical implementations. To design countermeasures against such attacks, the strategy for FPGA designers should also be different from that in ASIC. The available features should be exploited to design compact and strong countermeasures. In this paper, we propose methods to exploit the BRAMs in FPGAs for designing compact countermeasures. BRAM can be used to optimize intrinsic countermeasures like masking and dual-rail logic, which otherwise have significant overhead (at least 2X). The optimizations are applied on a real AES-128 co-processor and tested for area overhead and resistance on Xilinx Virtex-5 chips. The presented masking countermeasure has an overhead of only 16% when applied on AES. Moreover Dual-rail Precharge Logic (DPL) countermeasure has been optimized to pack the whole sequential part in the BRAM, hence enhancing the security. Proper robustness evaluations are conducted to analyze the optimization for area and security.

Digital Coherent Receivers and Advanced Optical Modulation Formats in 100 and 200 Gb/s DWDM Systems

Recent advances in coherent optical receivers is reviewed. Digital-Signal-Processing (DSP) based phase and polarization management techniques make coherent detection robust and feasible. With coherent detection, the complex field of the received optical signal is fully recovered, allowing compensation of linear and nonlinear optical impairments including chromatic dispersion (CD) and polarization-mode dispersion (PMD) using digital filters. Coherent detection and advanced optical modulation formats have become a key ingredient to the design of modern dense wavelength-division multiplexed (DWDM) optical broadband networks. In this paper, firstly we present the different subsystems of a digital coherent optical receiver, and secondly, we will compare the performance of some multi-level and multi-dimensional modulation formats in some physical impairments and in high spectral-efficiency (SE) and high-capacity DWDM transmissions, simulating the DSP with Matlab and the optical network performance with OptiSystem software.

Diseño y análisis de unidades didácticas para Procesado Digital de la Señal.

Este Proyecto Fin de Carrera pretende desarrollar una serie de unidades didácticas orientadas a mejorar el aprendizaje de la teoría de procesado digital de señales a través de la aplicación práctica. Con tal fin, se han diseñado una serie de prácticas que permitan al alumno alcanzar un apropiado nivel de conocimiento de la asignatura, la adquisición de competencias y alcanzar los resultados de aprendizaje previstos. Para desarrollar el proyecto primero se ha realizado una selección apropiada de los contenidos de la teoría de procesado digital de señales en relación con los resultados de aprendizaje esperados, seguidamente se han diseñado y validado unas prácticas basadas en un entorno de trabajo basado en MATLAB y DSP, y por último se ha redactado un manual de laboratorio que combina una parte teórica con su práctica correspondiente. El objetivo perseguido con la realización de estas prácticas es alcanzar un equilibrio teórico/práctico que permita sacar el máximo rendimiento de la asignatura desde el laboratorio, trabajando principalmente con el IDE Code Composer Studio junto con un kit de desarrollo basado en un DSP. ABSTRACT. This dissertation intends to develop some lessons oriented to improve about the digital signal processing theory. In order to get this objective some practices have been developed to allow to the students to achieve an appropriate level of knowledge of the subject, acquire skills and achieve the intended learning outcomes. To develop the project firstly it has been made an appropriate selection of the contents of the digital signal processing theory related with the expected results. After that, five practices based in a work environment based on Matlab and DSP have been designed and validated, and finally a laboratory manual has been drafted that combines the theoretical part with its corresponding practice. The objective with the implementation of these practices is to achieve a theoretical / practical balance to get the highest performance to the subject from the laboratory working mainly with the Code Composer Studio IDE together a development kit based on DSP.

Caracterización de un descodificador HEVC ejecutándose en un DSP

HEVC es el nuevo estándar de codificación de vídeo que está siendo desarrollado conjuntamente por las organizaciones ITU-T Video Coding Experts Group (VCEG) e ISO/IEC Moving Picture Experts Group (MPEG). Su objetivo principal es mejorar la compresión de vídeo, en relación a los actuales estándares. Es común hoy en día, debido a su flexibilidad para aplicaciones de bajo consumo, diseñar sistemas de descodificación de vídeo basados en un procesador digital de señal (DSP). En la mayoría de las veces, los diseños parten de un código creado para ser ejecutado en un ordenador personal y posteriormente se optimizan para tecnología DSP. El objetivo principal de este proyecto es caracterizar el rendimiento de un sistema basado en DSP que ejecute el código de un descodificador de video HEVC. ABSTRACT. HEVC is a new video coding standard which is being developed by both ITU-T Video Coding Experts Group (VCEG) and ISO/IEC Moving Picture Experts Group (MPEG). Its main goal is to improve video compression, compared with the actual standards. It is common practice, because of the flexibility in low power applications, to design video decoding systems using digital signal processors (DSP). Most of the time, these designs start with a code suitable to be executed in personal computers and then it is optimized forDSP technology. The main goal in this final degree project is to characterize the performance of a DSP based system executing an HEVC video decoder.

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.

Mode-division-multiplexed 3x112-Gb/s DP-QPSK transmission over 80 km few-mode fiber with inline MM-EDFA and blind DSP

We show transmission of a 3x112-Gb/s DP-QPSK mode-division-multiplexed signal up to 80km, with and without multi-mode EDFA, using blind 6x6 MIMO digital signal processing. We show that the OSNR-penalty induced by mode-mixing in the multi-mode EDFA is negligible.

Signal processing for molecular and cellular biological physics:an emerging field

Recent advances in our ability to watch the molecular and cellular processes of life in action-such as atomic force microscopy, optical tweezers and Forster fluorescence resonance energy transfer-raise challenges for digital signal processing (DSP) of the resulting experimental data. This article explores the unique properties of such biophysical time series that set them apart from other signals, such as the prevalence of abrupt jumps and steps, multi-modal distributions and autocorrelated noise. It exposes the problems with classical linear DSP algorithms applied to this kind of data, and describes new nonlinear and non-Gaussian algorithms that are able to extract information that is of direct relevance to biological physicists. It is argued that these new methods applied in this context typify the nascent field of biophysical DSP. Practical experimental examples are supplied.

Phase noise influence in coherent optical DnPSK systems with DSP based dispersion compensation

We present a comparative study of the influence of dispersion induced phase noise for n-level PSK systems. From the analysis, we conclude that the phase noise influence for classical homodyne/heterodyne PSK systems is entirely determined by the modulation complexity (expressed in terms of constellation diagram) and the analogue demodulation format. On the other hand, the use of digital signal processing (DSP) in homodyne/intradyne systems renders a fiber length dependence originating from the generation of equalization enhanced phase noise. For future high capacity systems, high constellations must be used in order to lower the symbol rate to practically manageable speeds, and this fact puts severe requirements to the signal and local oscillator (LO) linewidths. Our results for the bit-error-rate (BER) floor caused by the phase noise influence in the case of QPSK, 16PSK and 64PSK systems outline tolerance limitations for the LO performance: 5 MHz linewidth (at 3-dB level) for 100 Gbit/s QPSK; 1 MHz for 400 Gbit/s QPSK; 0.1 MHz for 400 Gbit/s 16PSK and 1 Tbit/s 64PSK systems. This defines design constrains for the phase noise impact in distributed-feed-back (DFB) or distributed-Bragg-reflector (DBR) semiconductor lasers, that would allow moving the system capacity from 100 Gbit/s system capacity to 400 Gbit/s in 3 years (1 Tbit/s in 5 years). It is imperative at the same time to increase the analogue to digital conversion (ADC) speed such that the single quadrature symbol rate goes from today's 25 GS/s to 100 GS/s (using two samples per symbol). © 2014 by Walter de Gruyter Berlin/Boston.

Mode-division-multiplexed 3x112-Gb/s DP-QPSK transmission over 80 km few-mode fiber with inline MM-EDFA and blind DSP

We show transmission of a 3x112-Gb/s DP-QPSK mode-division-multiplexed signal up to 80km, with and without multi-mode EDFA, using blind 6x6 MIMO digital signal processing. We show that the OSNR-penalty induced by mode-mixing in the multi-mode EDFA is negligible.

Progress toward 100-G digital coherent pluggables using InP-based photonics

A new generation of high-capacity WDM systems with extremely robust performance has been enabled by coherent transmission and digital signal processing. To facilitate widespread deployment of this technology, particularly in the metro space, new photonic components and subsystems are being developed to support cost-effective, compact, and scalable transceivers. We briefly review the recent progress in InP-based photonic components, and report numerical simulation results of an InP-based transceiver comprising a dual-polarization I/Q modulator and a commercial DSP ASIC. Predicted performance penalties due to the nonlinear response, lower bandwidth, and finite extinction ratio of these transceivers are less than 1 and 2 dB for 100-G PM-QPSK and 200-G PM-16QAM, respectively. Using the well-established Gaussian-Noise model, estimated system reach of 100-G PM-QPSK is greater than 600 km for typical ROADM-based metro-regional systems with internode losses up to 20 dB. © 1983-2012 IEEE.

Comparison of DSP-based nonlinear equalizers for intra-channel nonlinearity compensation in coherent optical OFDM

A novel versatile digital signal processing (DSP)-based equalizer using support vector machine regression (SVR) is proposed for 16-quadrature amplitude modulated (16-QAM) coherent optical orthogonal frequency-division multiplexing (CO-OFDM) and experimentally compared to traditional DSP-based deterministic fiber-induced nonlinearity equalizers (NLEs), namely the full-field digital back-propagation (DBP) and the inverse Volterra series transfer function-based NLE (V-NLE). For a 40 Gb/s 16-QAM CO-OFDM at 2000 km, SVR-NLE extends the optimum launched optical power (LOP) by 4 dB compared to V-NLE by means of reduction of fiber nonlinearity. In comparison to full-field DBP at a LOP of 6 dBm, SVR-NLE outperforms by ∼1 dB in Q-factor. In addition, SVR-NLE is the most computational efficient DSP-NLE.

Parametric signal regeneration

We present an ultra-high bandwidth all-optical digital signal regeneration device concept utilising non-degenerate parametric interaction in a one-dimensional waveguide. Performance is analysed in terms of re-amplification, re-timing, and re-shaping (including centre frequency correction) of time domain multiplexed signals. Bandwidths of 10-100 THz are achievable. (C) 2001 Published by Elsevier Science B.V.

Conversor estático de potência tolerante a falhas

O trabalho apresentado nesta dissertação refere-se à concepção, projecto e realização experimental de um conversor estático de potência tolerante a falhas. Foram analisados trabalhos de investigação sobre modos de falha de conversores electrónicos de potência, topologias de conversores tolerantes a falhas, métodos de detecção de falhas, entre outros. Com vista à concepção de uma solução, foram nomeados e analisados os principais modos de falhas para três soluções propostas de conversores com topologias tolerantes a falhas onde existem elementos redundantes em modo de espera. Foram analisados os vários aspectos de natureza técnica dos circuitos de potência e guiamento de sinais onde se salientam a necessidade de tempos mortos entre os sinais de disparo de IGBT do mesmo ramo, o isolamento galvânico entre os vários andares de disparo, a necessidade de minimizar as auto-induções entre o condensador DC e os braços do conversor de potência. Com vista a melhorar a fiabilidade e segurança de funcionamento do conversor estático de potência tolerante a falhas, foi concebido um circuito electrónico permitindo a aceleração da actuação normal de contactores e outro circuito responsável pelo encaminhamento e inibição dos sinais de disparo. Para a aplicação do conversor estático de potência tolerante a falhas desenvolvido num accionamento com um motor de corrente contínua, foi implementado um algoritmo de controlo numa placa de processamento digital de sinais (DSP), sendo a supervisão e actuação do sistema realizados em tempo-real, para a detecção de falhas e actuação de contactores e controlo de corrente e velocidade do motor utilizando uma estratégia de comando PWM. Foram realizados ensaios que, mediante uma detecção adequada de falhas, realiza a comutação entre blocos de conversores de potência. São apresentados e discutidos resultados experimentais, obtidos usando o protótipo laboratorial.

Projeto e implementação de um pré-regulador de fator de potência com controlo digital

Hoje em dia as fontes de alimentação possuem correção do fator de potência, devido às diversas normas regulamentares existentes, que introduziram grandes restrições no que respeita à distorção harmónica (THD) e fator de potência (FP). Este trabalho trata da análise, desenvolvimento e implementação de um Pré-Regulador de fator de potência com controlo digital. O controlo digital de conversores com recurso a processamento digital de sinal tem vindo a ser ao longo dos últimos anos, objeto de investigação e desenvolvimento, estando constantemente a surgirem modificações nas topologias existentes. Esta dissertação tem como objetivo estudar e implementar um Pré-Regulador Retificador Boost e o respetivo controlo digital. O controlo do conversor é feito através da técnica dos valores médios instantâneos da corrente de entrada, desenvolvido através da linguagem de descrição de hardware VHDL (VHSIC HDL – Very High Speed Integrated Circuit Hardware Description Language) e implementado num dispositivo FPGA (Field Programmable Gate Array) Spartan-3E. Neste trabalho são apresentadas análises matemáticas, para a obtenção das funções de transferência pertinentes ao projeto dos controladores. Para efetuar este controlo é necessário adquirir os sinais da corrente de entrada, tensão de entrada e tensão de saída. O sinal resultante do módulo de controlo é um sinal de PWM com valor de fator de ciclo variável ao longo do tempo. O projeto é simulado e validado através da plataforma MatLab/Simulink e PSIM, onde são apresentados resultados para o regime permanente e para transitórios da carga e da tensão de alimentação. Finalmente, o Pré-Regulador Retificador Boost controlado de forma digital é implementado em laboratório. Os resultados experimentais são apresentados para validar a metodologia e o projeto desenvolvidos.