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.

Some applications of distributed signal processing

In this work we review some earlier distributed algorithms developed by the authors and collaborators, which are based on two different approaches, namely, distributed moment estimation and distributed stochastic approximations. We show applications of these algorithms on image compression, linear classification and stochastic optimal control. In all cases, the benefit of cooperation is clear: even when the nodes have access to small portions of the data, by exchanging their estimates, they achieve the same performance as that of a centralized architecture, which would gather all the data from all the nodes.

Area-Efficient Linear Regression Architecture for Real-Time Signal Processing on FPGAs

Linear regression is a technique widely used in digital signal processing. It consists on finding the linear function that better fits a given set of samples. This paper proposes different hardware architectures for the implementation of the linear regression method on FPGAs, specially targeting area restrictive systems. It saves area at the cost of constraining the lengths of the input signal to some fixed values. We have implemented the proposed scheme in an Automatic Modulation Classifier, meeting the hard real-time constraints this kind of systems have.

Optical signal processing with electrooptic modulators and dispersion

A review of the main techniques that have been proposed for temporal processing of optical pulses that are the counterpart of the well-known spatial arrangements will be presented. They are translated to the temporal domain via the space-time duality and implemented with electrooptical phase and amplitude modulators and dispersive devices. We will introduce new variations of the conventional approaches and we will focus on their application to optical communications systems

Piezoelectric sensing and non-parametric statistical signal processing for health monitoring of hysteretic dampers used in seismic-resistant structures

The paper proposes a new application of non-parametric statistical processing of signals recorded from vibration tests for damage detection and evaluation on I-section steel segments. The steel segments investigated constitute the energy dissipating part of a new type of hysteretic damper that is used for passive control of buildings and civil engineering structures subjected to earthquake-type dynamic loadings. Two I-section steel segments with different levels of damage were instrumented with piezoceramic sensors and subjected to controlled white noise random vibrations. The signals recorded during the tests were processed using two non-parametric methods (the power spectral density method and the frequency response function method) that had never previously been applied to hysteretic dampers. The appropriateness of these methods for quantifying the level of damage on the I-shape steel segments is validated experimentally. Based on the results of the random vibrations, the paper proposes a new index that predicts the level of damage and the proximity of failure of the hysteretic damper

Banco de medidas automático para linealización de amplificadores

This work is focused on building and configuring a measurement test bench for non linear High Power Amplifiers, more precisely those ones based on the Envelope Elimination and Restoration. At first sight the test bench is composed of several arbitrary waveform generators, an oscilloscope, a vector signal generator and a spectrum analyzer all of them controlled remotely. The test bench works automatically, that is why several software control programs have been developed in order to control all this equipment. The control programs have been developed in Matlab/Octave Scripting language and at last chance in a more low level language as C. The signal processing algorithms, taking into account that the time alignment one is the most important, have been developed in Matlab/Octave Scripting too. An improvement of 10dB in the ACPR(Adjacent Channel Power Ratio) has been obtained just by applying the time alignment algorithm developed in this work

Unidad de efectos de audio en Simulink

La tecnología moderna de computación ha permitido cambiar radicalmente la investigación tecnológica en todos los ámbitos. El proceso general utilizado previamente consistía en el desarrollo de prototipos analógicos, creando múltiples versiones del mismo hasta llegar al resultado adecuado. Este es un proceso costoso a nivel económico y de carga de trabajo. Es por ello por lo que el proceso de investigación actual aprovecha las nuevas tecnologías para lograr el objetivo final mediante la simulación. Gracias al desarrollo de software para la simulación de distintas áreas se ha incrementado el ritmo de crecimiento de los avances tecnológicos y reducido el coste de los proyectos en investigación y desarrollo. La simulación, por tanto, permite desarrollar previamente prototipos simulados con un coste mucho menor para así lograr un producto final, el cual será llevado a cabo en su ámbito correspondiente. Este proceso no sólo se aplica en el caso de productos con circuitería, si bien es utilizado también en productos programados. Muchos de los programas actuales trabajan con algoritmos concretos cuyo funcionamiento debe ser comprobado previamente, para después centrarse en la codificación del mismo. Es en este punto donde se encuentra el objetivo de este proyecto, simular algoritmos de procesado digital de la señal antes de la codificación del programa final. Los sistemas de audio están basados en su totalidad en algoritmos de procesado de la señal, tanto analógicos como digitales, siendo estos últimos los que están sustituyendo al mundo analógico mediante los procesadores y los ordenadores. Estos algoritmos son la parte más compleja del sistema, y es la creación de nuevos algoritmos la base para lograr sistemas de audio novedosos y funcionales. Se debe destacar que los grupos de desarrollo de sistemas de audio presentan un amplio número de miembros con cometidos diferentes, separando las funciones de programadores e ingenieros de la señal de audio. Es por ello por lo que la simulación de estos algoritmos es fundamental a la hora de desarrollar nuevos y más potentes sistemas de audio. Matlab es una de las herramientas fundamentales para la simulación por ordenador, la cual presenta utilidades para desarrollar proyectos en distintos ámbitos. Sin embargo, en creciente uso actualmente se encuentra el software Simulink, herramienta especializada en la simulación de alto nivel que simplifica la dificultad de la programación en Matlab y permite desarrollar modelos de forma más rápida. Simulink presenta una completa funcionalidad para el desarrollo de algoritmos de procesado digital de audio. Por ello, el objetivo de este proyecto es el estudio de las capacidades de Simulink para generar sistemas de audio funcionales. A su vez, este proyecto pretende profundizar en los métodos de procesado digital de la señal de audio, logrando al final un paquete de sistemas de audio compatible con los programas de edición de audio actuales. ABSTRACT. Modern computer technology has dramatically changed the technological research in multiple areas. The overall process previously used consisted of the development of analog prototypes, creating multiple versions to reach the proper result. This is an expensive process in terms of an economically level and workload. For this reason actual investigation process take advantage of the new technologies to achieve the final objective through simulation. Thanks to the software development for simulation in different areas the growth rate of technological progress has been increased and the cost of research and development projects has been decreased. Hence, simulation allows previously the development of simulated protoypes with a much lower cost to obtain a final product, which will be held in its respective field. This process is not only applied in the case of circuitry products, but is also used in programmed products. Many current programs work with specific algorithms whose performance should be tested beforehand, which allows focusing on the codification of the program. This is the main point of this project, to simulate digital signal processing algorithms before the codification of the final program. Audio systems are entirely based on signal processing, both analog and digital systems, being the digital systems which are replacing the analog world thanks to the processors and computers. This algorithms are the most complex part of every system, and the creation of new algorithms is the most important step to achieve innovative and functional new audio systems. It should be noted that development groups of audio systems have a large number of members with different roles, separating them into programmers and audio signal engineers. For this reason, the simulation of this algorithms is essential when developing new and more powerful audio systems. Matlab is one of the most important tools for computer simulation, which has utilities to develop projects in different areas. However, the use of the Simulink software is constantly growing. It is a simulation tool specialized in high-level simulations which simplifies the difficulty of programming in Matlab and allows the developing of models faster. Simulink presents a full functionality for the development of algorithms for digital audio processing. Therefore, the objective of this project is to study the posibilities of Simulink to generate funcional audio systems. In turn, this projects aims to get deeper into the methods of digital audio signal processing, making at the end a software package of audio systems compatible with the current audio editing software.

Análisis espectral de señales cardiacas mediante frecuencia dominante

La Ingeniería Biomédica surgió en la década de 1950 como una fascinante mezcla interdisciplinaria, en la cual la ingeniería, la biología y la medicina aunaban esfuerzos para analizar y comprender distintas enfermedades. Las señales existentes en este área deben ser analizadas e interpretadas, más allá de las capacidades limitadas de la simple vista y la experiencia humana. Aquí es donde el procesamiento digital de la señal se postula como una herramienta indispensable para extraer la información relevante oculta en dichas señales. La electrocardiografía fue una de las primeras áreas en las que se aplicó el procesado digital de señales hace más de 50 años. Las señales electrocardiográficas continúan siendo, a día de hoy, objeto de estudio por parte de cardiólogos e ingenieros. En esta área, las técnicas de procesamiento de señal han ayudado a encontrar información oculta a simple vista que ha cambiado la forma de tratar ciertas enfermedades que fueron ya diagnosticadas previamente. Desde entonces, se han desarrollado numerosas técnicas de procesado de señales electrocardiográficas, pudiéndose resumir estas en tres grandes categorías: análisis tiempo-frecuencia, análisis de organización espacio-temporal y separación de la actividad atrial del ruido y las interferencias. Este proyecto se enmarca dentro de la primera categoría, análisis tiempo-frecuencia, y en concreto dentro de lo que se conoce como análisis de frecuencia dominante, la cual se va a aplicar al análisis de señales de fibrilación auricular. El proyecto incluye una parte teórica de análisis y desarrollo de algoritmos de procesado de señal, y una parte práctica, de programación y simulación con Matlab. Matlab es una de las herramientas fundamentales para el procesamiento digital de señales por ordenador, la cual presenta importantes funciones y utilidades para el desarrollo de proyectos en este campo. Por ello, se ha elegido dicho software como herramienta para la implementación del proyecto. ABSTRACT. Biomedical Engineering emerged in the 1950s as a fascinating interdisciplinary blend, in which engineering, biology and medicine pooled efforts to analyze and understand different diseases. Existing signals in this area should be analyzed and interpreted, beyond the limited capabilities of the naked eye and the human experience. This is where the digital signal processing is postulated as an indispensable tool to extract the relevant information hidden in these signals. Electrocardiography was one of the first areas where digital signal processing was applied over 50 years ago. Electrocardiographic signals remain, even today, the subject of close study by cardiologists and engineers. In this area, signal processing techniques have helped to find hidden information that has changed the way of treating certain diseases that were already previously diagnosed. Since then, numerous techniques have been developed for processing electrocardiographic signals. These methods can be summarized into three categories: time-frequency analysis, analysis of spatio-temporal organization and separation of atrial activity from noise and interferences. This project belongs to the first category, time-frequency analysis, and specifically to what is known as dominant frequency analysis, which is one of the fundamental tools applied in the analysis of atrial fibrillation signals. The project includes a theoretical part, related to the analysis and development of signal processing algorithms, and a practical part, related to programming and simulation using Matlab. Matlab is one of the fundamental tools for digital signal processing, presenting significant functions and advantages for the development of projects in this field. Therefore, we have chosen this software as a tool for project implementation.

Improvement for detection of microcalcifications through clustering algorithms and artificial neural networks

A new method for detecting microcalcifications in regions of interest (ROIs) extracted from digitized mammograms is proposed. The top-hat transform is a technique based on mathematical morphology operations and, in this paper, is used to perform contrast enhancement of the mi-crocalcifications. To improve microcalcification detection, a novel image sub-segmentation approach based on the possibilistic fuzzy c-means algorithm is used. From the original ROIs, window-based features, such as the mean and standard deviation, were extracted; these features were used as an input vector in a classifier. The classifier is based on an artificial neural network to identify patterns belonging to microcalcifications and healthy tissue. Our results show that the proposed method is a good alternative for automatically detecting microcalcifications, because this stage is an important part of early breast cancer detection

Orthogonal MCMC algorithms

Monte Carlo (MC) methods are widely used in signal processing, machine learning and stochastic optimization. A well-known class of MC methods are Markov Chain Monte Carlo (MCMC) algorithms. In this work, we introduce a novel parallel interacting MCMC scheme, where the parallel chains share information using another MCMC technique working on the entire population of current states. These parallel ?vertical? chains are led by random-walk proposals, whereas the ?horizontal? MCMC uses a independent proposal, which can be easily adapted by making use of all the generated samples. Numerical results show the advantages of the proposed sampling scheme in terms of mean absolute error, as well as robustness w.r.t. to initial values and parameter choice.

Exploring differences between phonetic classes in Sleep Apnoea Syndrome Patients using automatic speech processing techniques

This work is part of an on-going collaborative project between the medical and signal processing communities to promote new research efforts on automatic OSA (Obstructive Apnea Syndrome) diagnosis. In this paper, we explore the differences noted in phonetic classes (interphoneme) across groups (control/apnoea) and analyze their utility for OSA detection

A Radio Frequency Identification System for accurate indoor localization

In this paper we present a novel Radio Frequency Identification (RFID) system for accurate indoor localization. The system is composed of a standard Ultra High Frequency (UHF), ISO-18006C compliant RFID reader, a large set of standard passive RFID tags whose locations are known, and a newly developed tag-like RFID component that is attached to the items that need to be localized. The new semi-passive component, referred to as sensatag (sense-a-tag), has a dual functionality wherein it can sense the communication between the reader and standard tags which are in its proximity, and also communicate with the reader like standard tags using backscatter modulation. Based on the information conveyed by the sensatags to the reader, localization algorithms based on binary sensor principles can be developed. We present results from real measurements that show the accuracy of the proposed system.

Hand Image Segmentation by means of Gaussian Multiscale Aggregation for biometric applications

Applying biometrics to daily scenarios involves demanding requirements in terms of software and hardware. On the contrary, current biometric techniques are also being adapted to present-day devices, like mobile phones, laptops and the like, which are far from meeting the previous stated requirements. In fact, achieving a combination of both necessities is one of the most difficult problems at present in biometrics. Therefore, this paper presents a segmentation algorithm able to provide suitable solutions in terms of precision for hand biometric recognition, considering a wide range of backgrounds like carpets, glass, grass, mud, pavement, plastic, tiles or wood. Results highlight that segmentation accuracy is carried out with high rates of precision (F-measure 88%)), presenting competitive time results when compared to state-of-the-art segmentation algorithms time performance

A practical approach for outdoors distributed target localization in wireless sensor networks

Wireless sensor networks are posed as the new communication paradigm where the use of small, low-complexity, and low-power devices is preferred over costly centralized systems. The spectra of potential applications of sensor networks is very wide, ranging from monitoring, surveillance, and localization, among others. Localization is a key application in sensor networks and the use of simple, efficient, and distributed algorithms is of paramount practical importance. Combining convex optimization tools with consensus algorithms we propose a distributed localization algorithm for scenarios where received signal strength indicator readings are used. We approach the localization problem by formulating an alternative problem that uses distance estimates locally computed at each node. The formulated problem is solved by a relaxed version using semidefinite relaxation technique. Conditions under which the relaxed problem yields to the same solution as the original problem are given and a distributed consensusbased implementation of the algorithm is proposed based on an augmented Lagrangian approach and primaldual decomposition methods. Although suboptimal, the proposed approach is very suitable for its implementation in real sensor networks, i.e., it is scalable, robust against node failures and requires only local communication among neighboring nodes. Simulation results show that running an additional local search around the found solution can yield performance close to the maximum likelihood estimate.

Location-aided distributed primary user identification in a cognitive radio scenario

We address a cognitive radio scenario, where a number of secondary users performs identification of which primary user, if any, is trans- mitting, in a distributed way and using limited location information. We propose two fully distributed algorithms: the first is a direct iden- tification scheme, and in the other a distributed sub-optimal detection based on a simplified Neyman-Pearson energy detector precedes the identification scheme. Both algorithms are studied analytically in a realistic transmission scenario, and the advantage obtained by detec- tion pre-processing is also verified via simulation. Finally, we give details of their fully distributed implementation via consensus aver- aging algorithms.