Predicción de la velocidad crítica de flameo mediante el uso de modelos aeroelásticos extrapolados en tiempo real con datos de vuelo

La Aeroelasticidad fue definida por Arthur Collar en 1947 como "el estudio de la interacción mutua entre fuerzas inerciales, elásticas y aerodinámicas actuando sobre elementos estructurales expuestos a una corriente de aire". Actualmente, esta definición se ha extendido hasta abarcar la influencia del control („Aeroservoelasticidad‟) e, incluso, de la temperatura („Aerotermoelasticidad‟). En el ámbito de la Ingeniería Aeronáutica, los fenómenos aeroelásticos, tanto estáticos (divergencia, inversión de mando) como dinámicos (flameo, bataneo) son bien conocidos desde los inicios de la Aviación. Las lecciones aprendidas a lo largo de la Historia Aeronáutica han permitido establecer criterios de diseño destinados a mitigar la probabilidad de sufrir fenómenos aeroelásticos adversos durante la vida operativa de una aeronave. Adicionalmente, el gran avance experimentado durante esta última década en el campo de la Aerodinámica Computacional y en la modelización aeroelástica ha permitido mejorar la fiabilidad en el cálculo de las condiciones de flameo de una aeronave en su fase de diseño. Sin embargo, aún hoy, los ensayos en vuelo siguen siendo necesarios para validar modelos aeroelásticos, verificar que la aeronave está libre de inestabilidades aeroelásticas y certificar sus distintas envolventes. En particular, durante el proceso de expansión de la envolvente de una aeronave en altitud/velocidad, se requiere predecir en tiempo real las condiciones de flameo y, en consecuencia, evitarlas. A tal efecto, en el ámbito de los ensayos en vuelo, se han desarrollado diversas metodologías que predicen, en tiempo real, las condiciones de flameo en función de condiciones de vuelo ya verificadas como libres de inestabilidades aeroelásticas. De entre todas ellas, aquella que relaciona el amortiguamiento y la velocidad con un parámetro específico definido como „Margen de Flameo‟ (Flutter Margin), permanece como la técnica más común para proceder con la expansión de Envolventes en altitud/velocidad. No obstante, a pesar de su popularidad y facilidad de aplicación, dicha técnica no es adecuada cuando en la aeronave a ensayar se hallan presentes no-linealidades mecánicas como, por ejemplo, holguras. En particular, en vuelos de ensayo dedicados específicamente a expandir la envolvente en altitud/velocidad, las condiciones de „Oscilaciones de Ciclo Límite‟ (Limit Cycle Oscillations, LCOs) no pueden ser diferenciadas de manera precisa de las condiciones de flameo, llevando a una determinación excesivamente conservativa de la misma. La presente Tesis desarrolla una metodología novedosa, basada en el concepto de „Margen de Flameo‟, que permite predecir en tiempo real las condiciones de „Ciclo Límite‟, siempre que existan, distinguiéndolas de las de flameo. En una primera parte, se realiza una revisión bibliográfica de la literatura acerca de los diversos métodos de ensayo existentes para efectuar la expansión de la envolvente de una aeronave en altitud/velocidad, el efecto de las no-linealidades mecánicas en el comportamiento aeroelástico de dicha aeronave, así como una revisión de las Normas de Certificación civiles y militares respecto a este tema. En una segunda parte, se propone una metodología de expansión de envolvente en tiempo real, basada en el concepto de „Margen de Flameo‟, que tiene en cuenta la presencia de no-linealidades del tipo holgura en el sistema aeroelástico objeto de estudio. Adicionalmente, la metodología propuesta se valida contra un modelo aeroelástico bidimensional paramétrico e interactivo programado en Matlab. Para ello, se plantean las ecuaciones aeroelásticas no-estacionarias de un perfil bidimensional en la formulación espacio-estado y se incorpora la metodología anterior a través de un módulo de análisis de señal y otro módulo de predicción. En una tercera parte, se comparan las conclusiones obtenidas con las expuestas en la literatura actual y se aplica la metodología propuesta a resultados experimentales de ensayos en vuelo reales. En resumen, los principales resultados de esta Tesis son: 1. Resumen del estado del arte en los métodos de ensayo aplicados a la expansión de envolvente en altitud/velocidad y la influencia de no-linealidades mecánicas en la determinación de la misma. 2. Revisión de la normas de Certificación Civiles y las normas Militares en relación a la verificación aeroelástica de aeronaves y los límites permitidos en presencia de no-linealidades. 3. Desarrollo de una metodología de expansión de envolvente basada en el Margen de Flameo. 4. Validación de la metodología anterior contra un modelo aeroelástico bidimensional paramétrico e interactivo programado en Matlab/Simulink. 5. Análisis de los resultados obtenidos y comparación con resultados experimentales. ABSTRACT Aeroelasticity was defined by Arthur Collar in 1947 as “the study of the mutual interaction among inertia, elastic and aerodynamic forces when acting on structural elements surrounded by airflow”. Today, this definition has been updated to take into account the Controls („Aeroservoelasticity‟) and even the temperature („Aerothermoelasticity‟). Within the Aeronautical Engineering, aeroelastic phenomena, either static (divergence, aileron reversal) or dynamic (flutter, buzz), are well known since the early beginning of the Aviation. Lessons learned along the History of the Aeronautics have provided several design criteria in order to mitigate the probability of encountering adverse aeroelastic phenomena along the operational life of an aircraft. Additionally, last decade improvements experienced by the Computational Aerodynamics and aeroelastic modelization have refined the flutter onset speed calculations during the design phase of an aircraft. However, still today, flight test remains as a key tool to validate aeroelastic models, to verify flutter-free conditions and to certify the different envelopes of an aircraft. Specifically, during the envelope expansion in altitude/speed, real time prediction of flutter conditions is required in order to avoid them in flight. In that sense, within the flight test community, several methodologies have been developed to predict in real time flutter conditions based on free-flutter flight conditions. Among them, the damping versus velocity technique combined with a Flutter Margin implementation remains as the most common technique used to proceed with the envelope expansion in altitude/airspeed. However, although its popularity and „easy to implement‟ characteristics, several shortcomings can adversely affect to the identification of unstable conditions when mechanical non-linearties, as freeplay, are present. Specially, during test flights devoted to envelope expansion in altitude/airspeed, Limits Cycle Oscillations (LCOs) conditions can not be accurately distinguished from those of flutter and, in consequence, it leads to an excessively conservative envelope determination. The present Thesis develops a new methodology, based on the Flutter Margin concept, that enables in real time the prediction of the „Limit Cycle‟ conditions, whenever they exist, without degrading the capability of predicting the flutter onset speed. The first part of this Thesis presents a review of the state of the art regarding the test methods available to proceed with the envelope expansion of an aircraft in altitude/airspeed and the effect of mechanical non-linearities on the aeroelastic behavior. Also, both civil and military regulations are reviewed with respect aeroelastic investigation of air vehicles. The second part of this Thesis proposes a new methodology to perform envelope expansion in real time based on the Flutter Margin concept when non-linearities, as freeplay, are present. Additionally, this methodology is validated against a Matlab/Slimulink bidimensional aeroelastic model. This model, parametric and interactive, is formulated within the state-space field and it implements the proposed methodology through two main real time modules: A signal processing module and a prediction module. The third part of this Thesis compares the final conclusions derived from the proposed methodology with those stated by the flight test community and experimental results. In summary, the main results provided by this Thesis are: 1. State of the Art review of the test methods applied to envelope expansion in altitude/airspeed and the influence of mechanical non-linearities in its identification. 2. Review of the main civil and military regulations regarding the aeroelastic verification of air vehicles and the limits set when non-linearities are present. 3. Development of a methodology for envelope expansion based on the Flutter Margin concept. 4. A Matlab/Simulink 2D-[aeroelastic model], parametric and interactive, used as a tool to validate the proposed methodology. 5. Conclusions driven from the present Thesis and comparison with experimental results.

A generalized method for flat plates impact detection and location

In the last years, many analyses from acoustic signal processing have been used for different applications. In most cases, these sensor systems are based on the determination of times of flight for signals from every transducer. This paper presents a flat plate generalization method for impact detection and location over linear links or bars-based structures. The use of three piezoelectric sensors allow to achieve the position and impact time while the use of additional sensors lets cover a larger area of detection and avoid wrong timing difference measurements. An experimental setup and some experimental results are briefly presented.

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.

Adaptive spatio-temporal filter for low-cost camera depth maps

In this paper we present an adaptive spatio-temporal filter that aims to improve low-cost depth camera accuracy and stability over time. The proposed system is composed by three blocks that are used to build a reliable depth map of static scenes. An adaptive joint-bilateral filter is used to obtain consistent depth maps by jointly considering depth and video information and by adapting its parameters to different levels of estimated noise. Kalman filters are used to reduce the temporal random fluctuations of the measurements. Finally an interpolation algorithm is used to obtain consistent depth maps in the regions where the depth information is not available. Results show that this approach allows to considerably improve the depth maps quality by considering spatio-temporal information and by adapting its parameters to different levels of noise.

A framework for the analysis and optimization of encoding latency for multiview video

We present a novel framework for the analysis and optimization of encoding latency for multiview video. Firstly, we characterize the elements that have an influence in the encoding latency performance: (i) the multiview prediction structure and (ii) the hardware encoder model. Then, we provide algorithms to find the encoding latency of any arbitrary multiview prediction structure. The proposed framework relies on the directed acyclic graph encoder latency (DAGEL) model, which provides an abstraction of the processing capacity of the encoder by considering an unbounded number of processors. Using graph theoretic algorithms, the DAGEL model allows us to compute the encoding latency of a given prediction structure, and determine the contribution of the prediction dependencies to it. As an example of DAGEL application, we propose an algorithm to reduce the encoding latency of a given multiview prediction structure up to a target value. In our approach, a minimum number of frame dependencies are pruned, until the latency target value is achieved, thus minimizing the degradation of the rate-distortion performance due to the removal of the prediction dependencies. Finally, we analyze the latency performance of the DAGEL derived prediction structures in multiview encoders with limited processing capacity.

Estimating tremor in Vocal Fold Biomechanics for Neurological Disease characterisation

Neurological Diseases (ND) are affecting larger segments of aging population every year. Treatment is dependent on expensive accurate and frequent monitoring. It is well known that ND leave correlates in speech and phonation. The present work shows a method to detect alterations in vocal fold tension during phonation. These may appear either as hypertension or as cyclical tremor. Estimations of tremor may be produced by auto-regressive modeling of the vocal fold tension series in sustained phonation. The correlates obtained are a set of cyclicality coefficients, the frequency and the root mean square amplitude of the tremor. Statistical distributions of these correlates obtained from a set of male and female subjects are presented. Results from five study cases of female voice are also given.

Wavelet description of the Glottal Gap

The Glottal Source correlates reconstructed from the phonated parts of voice may render interesting information with applicability in different fields. One of them is defective closure (gap) detection. Through the paper the background to explain the physical foundations of defective gap are reviewed. A possible method to estimate defective gap is also presented based on a Wavelet Description of the Glottal Source. The method is validated using results from the analysis of a gender-balanced speakers database. Normative values for the different parameters estimated are given. A set of study cases with deficient glottal closure is presented and discussed.

Temporal task allocation in periodic environments. An approach based on synchronization

In this paper, we study a robot swarm that has to perform task allocation in an environment that features periodic properties. In this environment, tasks appear in different areas following periodic temporal patterns. The swarm has to reallocate its workforce periodically, performing a temporal task allocation that must be synchronized with the environment to be effective. We tackle temporal task allocation using methods and concepts that we borrow from the signal processing literature. In particular, we propose a distributed temporal task allocation algorithm that synchronizes robots of the swarm with the environment and with each other. In this algorithm, robots use only local information and a simple visual communication protocol based on light blinking. Our results show that a robot swarm that uses the proposed temporal task allocation algorithm performs considerably more tasks than a swarm that uses a greedy algorithm.

Wavelets in High-Capacity DWDM Systems and Modal-Division Multiplexed Transmissions

Optical communications receivers using wavelet signals processing is proposed in this paper for dense wavelength-division multiplexed (DWDM) systems and modal-division multiplexed (MDM) transmissions. The optical signal-to-noise ratio (OSNR) required to demodulate polarization-division multiplexed quadrature phase shift keying (PDM-QPSK) modulation format is alleviated with the wavelet denoising process. This procedure improves the bit error rate (BER) performance and increasing the transmission distance in DWDM systems. Additionally, the wavelet-based design relies on signal decomposition using time-limited basis functions allowing to reduce the computational cost in Digital-Signal-Processing (DSP) module. Attending to MDM systems, a new scheme of encoding data bits based on wavelets is presented to minimize the mode coupling in few-mode (FWF) and multimode fibers (MMF). The Shifted Prolate Wave Spheroidal (SPWS) functions are proposed to reduce the modal interference.

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.

A new approach for the glotis segmentation using Snakes

The present work describes a new methodology for the automatic detection of the glottal space from laryngeal images based on active contour models (snakes). In order to obtain an appropriate image for the use of snakes based techniques, the proposed algorithm combines a pre-processing stage including some traditional techniques (thresholding and median filter) with more sophisticated ones such as anisotropic filtering. The value selected for the thresholding was fixed to the 85% of the maximum peak of the image histogram, and the anisotropic filter permits to distinguish two intensity levels, one corresponding to the background and the other one to the foreground (glottis). The initialization carried out is based on the magnitude obtained using the Gradient Vector Flow field, ensuring an automatic process for the selection of the initial contour. The performance of the algorithm is tested using the Pratt coefficient and compared against a manual segmentation. The results obtained suggest that this method provided results comparable with other techniques such as the proposed in (Osma-Ruiz et al., 2008).

Fully adaptive gaussian mixture metropolis-hastings algorithm

Markov Chain Monte Carlo methods are widely used in signal processing and communications for statistical inference and stochastic optimization. In this work, we introduce an efficient adaptive Metropolis-Hastings algorithm to draw samples from generic multimodal and multidimensional target distributions. The proposal density is a mixture of Gaussian densities with all parameters (weights, mean vectors and covariance matrices) updated using all the previously generated samples applying simple recursive rules. Numerical results for the one and two-dimensional cases are provided.

Cross-Products LASSO

Negative co-occurrence is a common phenomenon in many signal processing applications. In some cases the signals involved are sparse, and this information can be exploited to recover them. In this paper, we present a sparse learning approach that explicitly takes into account negative co-occurrence. This is achieved by adding a novel penalty term to the LASSO cost function based on the cross-products between the reconstruction coefficients. Although the resulting optimization problem is non-convex, we develop a new and efficient method for solving it based on successive convex approximations. Results on synthetic data, for both complete and overcomplete dictionaries, are provided to validate the proposed approach.

Efficient Monte Carlo optimization for multi-label classifier chains

Multi-label classification (MLC) is the supervised learning problem where an instance may be associated with multiple labels. Modeling dependencies between labels allows MLC methods to improve their performance at the expense of an increased computational cost. In this paper we focus on the classifier chains (CC) approach for modeling dependencies. On the one hand, the original CC algorithm makes a greedy approximation, and is fast but tends to propagate errors down the chain. On the other hand, a recent Bayes-optimal method improves the performance, but is computationally intractable in practice. Here we present a novel double-Monte Carlo scheme (M2CC), both for finding a good chain sequence and performing efficient inference. The M2CC algorithm remains tractable for high-dimensional data sets and obtains the best overall accuracy, as shown on several real data sets with input dimension as high as 1449 and up to 103 labels.

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.