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.

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

A region based approach to background modeling in a wavelet multi-resolution framework

In the field of detection and monitoring of dynamic objects in quasi-static scenes, background subtraction techniques where background is modeled at pixel-level, although showing very significant limitations, are extensively used. In this work we propose a novel approach to background modeling that operates at region-level in a wavelet based multi-resolution framework. Based on a segmentation of the background, characterization is made for each region independently as a mixture of K Gaussian modes, considering the model of the approximation and detail coefficients at the different wavelet decomposition levels. Background region characterization is updated along time, and the detection of elements of interest is carried out computing the distance between background region models and those of each incoming image in the sequence. The inclusion of the context in the modeling scheme through each region characterization makes the model robust, being able to support not only gradual illumination and long-term changes, but also sudden illumination changes and the presence of strong shadows in the scene

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.

Contributions to the contextualization of human-machine spoken interaction systems

The design and development of spoken interaction systems has been a thoroughly studied research scope for the last decades. The aim is to obtain systems with the ability to interact with human agents with a high degree of naturalness and efficiency, allowing them to carry out the actions they desire using speech, as it is the most natural means of communication between humans. To achieve that degree of naturalness, it is not enough to endow systems with the ability to accurately understand the user’s utterances and to properly react to them, even considering the information provided by the user in his or her previous interactions. The system has also to be aware of the evolution of the conditions under which the interaction takes place, in order to act the most coherent way as possible at each moment. Consequently, one of the most important features of the system is that it has to be context-aware. This context awareness of the system can be reflected in the modification of the behaviour of the system taking into account the current situation of the interaction. For instance, the system should decide which action it has to carry out, or the way to perform it, depending on the user that requests it, on the way that the user addresses the system, on the characteristics of the environment in which the interaction takes place, and so on. In other words, the system has to adapt its behaviour to these evolving elements of the interaction. Moreover that adaptation has to be carried out, if possible, in such a way that the user: i) does not perceive that the system has to make any additional effort, or to devote interaction time to perform tasks other than carrying out the requested actions, and ii) does not have to provide the system with any additional information to carry out the adaptation, which could imply a lesser efficiency of the interaction, since users should devote several interactions only to allow the system to become adapted. In the state-of-the-art spoken dialogue systems, researchers have proposed several disparate strategies to adapt the elements of the system to different conditions of the interaction (such as the acoustic characteristics of a specific user’s speech, the actions previously requested, and so on). Nevertheless, to our knowledge there is not any consensus on the procedures to carry out these adaptation. The approaches are to an extent unrelated from one another, in the sense that each one considers different pieces of information, and the treatment of that information is different taking into account the adaptation carried out. In this regard, the main contributions of this Thesis are the following ones: Definition of a contextualization framework. We propose a unified approach that can cover any strategy to adapt the behaviour of a dialogue system to the conditions of the interaction (i.e. the context). In our theoretical definition of the contextualization framework we consider the system’s context as all the sources of variability present at any time of the interaction, either those ones related to the environment in which the interaction takes place, or to the human agent that addresses the system at each moment. Our proposal relies on three aspects that any contextualization approach should fulfill: plasticity (i.e. the system has to be able to modify its behaviour in the most proactive way taking into account the conditions under which the interaction takes place), adaptivity (i.e. the system has also to be able to consider the most appropriate sources of information at each moment, both environmental and user- and dialogue-dependent, to effectively adapt to the conditions aforementioned), and transparency (i.e. the system has to carry out the contextualizaton-related tasks in such a way that the user neither perceives them nor has to do any effort in providing the system with any information that it needs to perform that contextualization). Additionally, we could include a generality aspect to our proposed framework: the main features of the framework should be easy to adopt in any dialogue system, regardless of the solution proposed to manage the dialogue. Once we define the theoretical basis of our contextualization framework, we propose two cases of study on its application in a spoken dialogue system. We focus on two aspects of the interaction: the contextualization of the speech recognition models, and the incorporation of user-specific information into the dialogue flow. One of the modules of a dialogue system that is more prone to be contextualized is the speech recognition system. This module makes use of several models to emit a recognition hypothesis from the user’s speech signal. Generally speaking, a recognition system considers two types of models: an acoustic one (that models each of the phonemes that the recognition system has to consider) and a linguistic one (that models the sequences of words that make sense for the system). In this work we contextualize the language model of the recognition system in such a way that it takes into account the information provided by the user in both his or her current utterance and in the previous ones. These utterances convey information useful to help the system in the recognition of the next utterance. The contextualization approach that we propose consists of a dynamic adaptation of the language model that is used by the recognition system. We carry out this adaptation by means of a linear interpolation between several models. Instead of training the best interpolation weights, we make them dependent on the conditions of the dialogue. In our approach, the system itself will obtain these weights as a function of the reliability of the different elements of information available, such as the semantic concepts extracted from the user’s utterance, the actions that he or she wants to carry out, the information provided in the previous interactions, and so on. One of the aspects more frequently addressed in Human-Computer Interaction research is the inclusion of user specific characteristics into the information structures managed by the system. The idea is to take into account the features that make each user different from the others in order to offer to each particular user different services (or the same service, but in a different way). We could consider this approach as a user-dependent contextualization of the system. In our work we propose the definition of a user model that contains all the information of each user that could be potentially useful to the system at a given moment of the interaction. In particular we will analyze the actions that each user carries out throughout his or her interaction. The objective is to determine which of these actions become the preferences of that user. We represent the specific information of each user as a feature vector. Each of the characteristics that the system will take into account has a confidence score associated. With these elements, we propose a probabilistic definition of a user preference, as the action whose likelihood of being addressed by the user is greater than the one for the rest of actions. To include the user dependent information into the dialogue flow, we modify the information structures on which the dialogue manager relies to retrieve information that could be needed to solve the actions addressed by the user. Usage preferences become another source of contextual information that will be considered by the system towards a more efficient interaction (since the new information source will help to decrease the need of the system to ask users for additional information, thus reducing the number of turns needed to carry out a specific action). To test the benefits of the contextualization framework that we propose, we carry out an evaluation of the two strategies aforementioned. We gather several performance metrics, both objective and subjective, that allow us to compare the improvements of a contextualized system against the baseline one. We will also gather the user’s opinions as regards their perceptions on the behaviour of the system, and its degree of adaptation to the specific features of each interaction. Resumen El diseño y el desarrollo de sistemas de interacción hablada ha sido objeto de profundo estudio durante las pasadas décadas. El propósito es la consecución de sistemas con la capacidad de interactuar con agentes humanos con un alto grado de eficiencia y naturalidad. De esta manera, los usuarios pueden desempeñar las tareas que deseen empleando la voz, que es el medio de comunicación más natural para los humanos. A fin de alcanzar el grado de naturalidad deseado, no basta con dotar a los sistemas de la abilidad de comprender las intervenciones de los usuarios y reaccionar a ellas de manera apropiada (teniendo en consideración, incluso, la información proporcionada en previas interacciones). Adicionalmente, el sistema ha de ser consciente de las condiciones bajo las cuales transcurre la interacción, así como de la evolución de las mismas, de tal manera que pueda actuar de la manera más coherente en cada instante de la interacción. En consecuencia, una de las características primordiales del sistema es que debe ser sensible al contexto. Esta capacidad del sistema de conocer y emplear el contexto de la interacción puede verse reflejada en la modificación de su comportamiento debida a las características actuales de la interacción. Por ejemplo, el sistema debería decidir cuál es la acción más apropiada, o la mejor manera de llevarla a término, dependiendo del usuario que la solicita, del modo en el que lo hace, etcétera. En otras palabras, el sistema ha de adaptar su comportamiento a tales elementos mutables (o dinámicos) de la interacción. Dos características adicionales son requeridas a dicha adaptación: i) el usuario no ha de percibir que el sistema dedica recursos (temporales o computacionales) a realizar tareas distintas a las que aquél le solicita, y ii) el usuario no ha de dedicar esfuerzo alguno a proporcionar al sistema información adicional para llevar a cabo la interacción. Esto último implicaría una menor eficiencia de la interacción, puesto que los usuarios deberían dedicar parte de la misma a proporcionar información al sistema para su adaptación, sin ningún beneficio inmediato. En los sistemas de diálogo hablado propuestos en la literatura, se han propuesto diferentes estrategias para llevar a cabo la adaptación de los elementos del sistema a las diferentes condiciones de la interacción (tales como las características acústicas del habla de un usuario particular, o a las acciones a las que se ha referido con anterioridad). Sin embargo, no existe una estrategia fija para proceder a dicha adaptación, sino que las mismas no suelen guardar una relación entre sí. En este sentido, cada una de ellas tiene en cuenta distintas fuentes de información, la cual es tratada de manera diferente en función de las características de la adaptación buscada. Teniendo en cuenta lo anterior, las contribuciones principales de esta Tesis son las siguientes: Definición de un marco de contextualización. Proponemos un criterio unificador que pueda cubrir cualquier estrategia de adaptación del comportamiento de un sistema de diálogo a las condiciones de la interacción (esto es, el contexto de la misma). En nuestra definición teórica del marco de contextualización consideramos el contexto del sistema como todas aquellas fuentes de variabilidad presentes en cualquier instante de la interacción, ya estén relacionadas con el entorno en el que tiene lugar la interacción, ya dependan del agente humano que se dirige al sistema en cada momento. Nuestra propuesta se basa en tres aspectos que cualquier estrategia de contextualización debería cumplir: plasticidad (es decir, el sistema ha de ser capaz de modificar su comportamiento de la manera más proactiva posible, teniendo en cuenta las condiciones en las que tiene lugar la interacción), adaptabilidad (esto es, el sistema ha de ser capaz de considerar la información oportuna en cada instante, ya dependa del entorno o del usuario, de tal manera que adecúe su comportamiento de manera eficaz a las condiciones mencionadas), y transparencia (que implica que el sistema ha de desarrollar las tareas relacionadas con la contextualización de tal manera que el usuario no perciba la manera en que dichas tareas se llevan a cabo, ni tampoco deba proporcionar al sistema con información adicional alguna). De manera adicional, incluiremos en el marco propuesto el aspecto de la generalidad: las características del marco de contextualización han de ser portables a cualquier sistema de diálogo, con independencia de la solución propuesta en los mismos para gestionar el diálogo. Una vez hemos definido las características de alto nivel de nuestro marco de contextualización, proponemos dos estrategias de aplicación del mismo a un sistema de diálogo hablado. Nos centraremos en dos aspectos de la interacción a adaptar: los modelos empleados en el reconocimiento de habla, y la incorporación de información específica de cada usuario en el flujo de diálogo. Uno de los módulos de un sistema de diálogo más susceptible de ser contextualizado es el sistema de reconocimiento de habla. Este módulo hace uso de varios modelos para generar una hipótesis de reconocimiento a partir de la señal de habla. En general, un sistema de reconocimiento emplea dos tipos de modelos: uno acústico (que modela cada uno de los fonemas considerados por el reconocedor) y uno lingüístico (que modela las secuencias de palabras que tienen sentido desde el punto de vista de la interacción). En este trabajo contextualizamos el modelo lingüístico del reconocedor de habla, de tal manera que tenga en cuenta la información proporcionada por el usuario, tanto en su intervención actual como en las previas. Estas intervenciones contienen información (semántica y/o discursiva) que puede contribuir a un mejor reconocimiento de las subsiguientes intervenciones del usuario. La estrategia de contextualización propuesta consiste en una adaptación dinámica del modelo de lenguaje empleado en el reconocedor de habla. Dicha adaptación se lleva a cabo mediante una interpolación lineal entre diferentes modelos. En lugar de entrenar los mejores pesos de interpolación, proponemos hacer los mismos dependientes de las condiciones actuales de cada diálogo. El propio sistema obtendrá estos pesos como función de la disponibilidad y relevancia de las diferentes fuentes de información disponibles, tales como los conceptos semánticos extraídos a partir de la intervención del usuario, o las acciones que el mismo desea ejecutar. Uno de los aspectos más comúnmente analizados en la investigación de la Interacción Persona-Máquina es la inclusión de las características específicas de cada usuario en las estructuras de información empleadas por el sistema. El objetivo es tener en cuenta los aspectos que diferencian a cada usuario, de tal manera que el sistema pueda ofrecer a cada uno de ellos el servicio más apropiado (o un mismo servicio, pero de la manera más adecuada a cada usuario). Podemos considerar esta estrategia como una contextualización dependiente del usuario. En este trabajo proponemos la definición de un modelo de usuario que contenga toda la información relativa a cada usuario, que pueda ser potencialmente utilizada por el sistema en un momento determinado de la interacción. En particular, analizaremos aquellas acciones que cada usuario decide ejecutar a lo largo de sus diálogos con el sistema. Nuestro objetivo es determinar cuáles de dichas acciones se convierten en las preferencias de cada usuario. La información de cada usuario quedará representada mediante un vector de características, cada una de las cuales tendrá asociado un valor de confianza. Con ambos elementos proponemos una definición probabilística de una preferencia de uso, como aquella acción cuya verosimilitud es mayor que la del resto de acciones solicitadas por el usuario. A fin de incluir la información dependiente de usuario en el flujo de diálogo, llevamos a cabo una modificación de las estructuras de información en las que se apoya el gestor de diálogo para recuperar información necesaria para resolver ciertos diálogos. En dicha modificación las preferencias de cada usuario pasarán a ser una fuente adicional de información contextual, que será tenida en cuenta por el sistema en aras de una interacción más eficiente (puesto que la nueva fuente de información contribuirá a reducir la necesidad del sistema de solicitar al usuario información adicional, dando lugar en consecuencia a una reducción del número de intervenciones necesarias para llevar a cabo una acción determinada). Para determinar los beneficios de las aplicaciones del marco de contextualización propuesto, llevamos a cabo una evaluación de un sistema de diálogo que incluye las estrategias mencionadas. Hemos recogido diversas métricas, tanto objetivas como subjetivas, que nos permiten determinar las mejoras aportadas por un sistema contextualizado en comparación con el sistema sin contextualizar. De igual manera, hemos recogido las opiniones de los participantes en la evaluación acerca de su percepción del comportamiento del sistema, y de su capacidad de adaptación a las condiciones concretas de cada interacción.

Lifetime maximization for beamforming applications in wireless sensor networks

Energy efficiency is a major design issue in the context of Wireless Sensor Networks (WSN). If data is to be sent to a far-away base station, collaborative beamforming by the sensors may help to dis- tribute the load among the nodes and reduce fast battery depletion. However, collaborative beamforming techniques are far from opti- mality and in many cases may be wasting more power than required. In this contribution we consider the issue of energy efficiency in beamforming applications. Using a convex optimization framework, we propose the design of a virtual beamformer that maximizes the network's lifetime while satisfying a pre-specified Quality of Service (QoS) requirement. A distributed consensus-based algorithm for the computation of the optimal beamformer is also provided

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.

Rectangular room dimensions estimation using narrowband signal and sectorized antennas

A system for estimation of unknown rectangular room dimensions based on two radio transceivers, both capable of full duplex operations, is presented. The approach is based on CIR measurements taken at the same place where the signal is transmitted (generated), commonly known as self- to-self CIR. Another novelty is the receiver antenna design which consists of eight sectorized antennas with 45° aperture in the horizontal plane, whose total coverage corresponds to the isotropic one. The dimensions of a rectangular room are reconstructed directly from radio impulse responses by extracting the information regarding features like round trip time, received signal strength and reverberation time. Using radar approach the estimation of walls and corners positions are derived. Additionally, the analysis of the absorption coefficient of the test environment is conducted and a typical coefficient for office room with furniture is proposed. Its accuracy is confirmed through the results of volume estimation. Tests using measured data were performed, and the simulation results confirm the feasibility of the approach.

Design of universal multicoset sampling patterns for compressed sensing of multiband sparse signals

Many problems in digital communications involve wideband radio signals. As the most recent example, the impressive advances in Cognitive Radio systems make even more necessary the development of sampling schemes for wideband radio signals with spectral holes. This is equivalent to considering a sparse multiband signal in the framework of Compressive Sampling theory. Starting from previous results on multicoset sampling and recent advances in compressive sampling, we analyze the matrix involved in the corresponding reconstruction equation and define a new method for the design of universal multicoset codes, that is, codes guaranteeing perfect reconstruction of the sparse multiband signal.

Structural health monitoring network system with wireless communications inside closed aerospace structures

Structural Health Monitoring (SHM) requires integrated "all in one" electronic devices capable of performing analysis of structural integrity and on-board damage detection in aircraft?s structures. PAMELA III (Phased Array Monitoring for Enhanced Life Assessment, version III) SHM embedded system is an example of this device type. This equipment is capable of generating excitation signals to be applied to an array of integrated piezoelectric Phased Array (PhA) transducers stuck to aircraft structure, acquiring the response signals, and carrying out the advanced signal processing to obtain SHM maps. PAMELA III is connected with a host computer in order to receive the configuration parameters and sending the obtained SHM maps, alarms and so on. This host can communicate with PAMELA III through an Ethernet interface. To avoid the use of wires where necessary, it is possible to add Wi-Fi capabilities to PAMELA III, connecting a Wi-Fi node working as a bridge, and to establish a wireless communication between PAMELA III and the host. However, in a real aircraft scenario, several PAMELA III devices must work together inside closed structures. In this situation, it is not possible for all PAMELA III devices to establish a wireless communication directly with the host, due to the signal attenuation caused by the different obstacles of the aircraft structure. To provide communication among all PAMELA III devices and the host, a wireless mesh network (WMN) system has been implemented inside a closed aluminum wingbox. In a WMN, as long as a node is connected to at least one other node, it will have full connectivity to the entire network because each mesh node forwards packets to other nodes in the network as required. Mesh protocols automatically determine the best route through the network and can dynamically reconfigure the network if a link drops out. The advantages and disadvantages on the use of a wireless mesh network system inside closed aerospace structures are discussed.

Integrated electronic system for ultrasonic structural health monitoring

A fully integrated on-board electronic system that can perform in-situ structural health monitoring (SHM) of aircraft?s structures using specifically designed equipment for SHM based on guided wave ultrasonic method or Lamb waves? method is introduced. This equipment is called Phased Array Monitoring for Enhanced Life Assessment (PAMELA III) and is an essential part of overall PAMELA SHM? system. PAMELA III can generate any kind of excitation signals, acquire the response signals that propagate throughout the structure being tested, and perform the signal processing for damage detection directly on the structure without need to send the huge amount of raw signals but only the final SHM maps. It monitors the structure by means of an array of integrated Phased Array (PhA) transducers preferably bonded onto the host structure. The PAMELA III hardware for SHM mapping has been designed, built and subjected to laboratory tests, using aluminum and CFRP structures. The 12 channel system has been designed to be low weight (265 grams only), to have a small form factor, to be directly mounted above the integrated PhA transducers without need for cables and to be EMI protected so that the equipment can be taken on board an aircraft to perform required SHM analyses by use of embedded SHM algorithms. Moreover, the autonomous, automatic and on real-time working procedure makes it suitable for the avionic field, sending the corresponding alerts, maps and reports to external equipment.

El sistema de grabación y reproducción sonora ambiofónico

Este proyecto fin de carrera trata del sistema de grabación y reproducción sonora ambiofónico, destacar que este sistema y la tecnología que emplea es de dominio público. La ambiofonía se basa en un amalgama de investigaciones recientes y de los ya bien sabidos principios psicoacústicos y binaurales. Estos avances han expandido nuevas fronteras en lo concerniente a la grabación y reproducción de audio, así como de presentar al oyente un campo sonoro a la entrada de sus oídos lo más parecido posible al campo sonoro al que se expondría al oyente en el momento y lugar de la toma de sonido, es decir, reconstruye un campo sonora binaural. Este sistema ha podido desarrollarse, de una manera bastante satisfactoria, gracias a todos los estudios y textos anteriores en materia de psicoacústica y del mecanismo de escucha humano. Otro factor gracias al cual es posible y asequible, tanto el desarrollo como el disfrute de esta tecnología, es el hecho que en nuestros días es muy económico disponer de ordenadores lo suficientemente potentes y rápidos para realizar el procesado de señales que se requiere de una manera bastante rápida. Los desarrolladores de dicha tecnología han publicado diversos documentos y archivos descargables de la red con aplicaciones para la implementación de sistemas ambiofónicos de manera gratuita para uso privado. El sistema ambiofónico se basa en la combinación de factores psicoacústicos ignorados o subestimados y lo ya sabido sobre las propiedades acústicas de salas, tanto de salas en las que tienen lugar las ejecuciones musicales (auditorios, teatros, salas de conciertos...), como de salas de escucha (salones de domicilios, controles de estudios...). En la parte práctica del proyecto se van a realizar una serie de grabaciones musicales empleando tanto técnicas estereofónicas tradicionales como ambiofónicas de grabación con el fin de describir y comparar ambas técnicas microfónicas. También servirá para estudiar hasta que punto es favorable subjetivamente para el oyente el hecho de realizar la toma de sonido teniendo en cuenta las propiedades del sistema de reproducción ambiofónico. Esta comparación nos dará una idea de hasta donde se puede llegar, en cuanto a sensación de realidad para el oyente, al tener en cuenta durante el proceso de grabación efectos como la respuesta del pabellón auditivo del oyente, la cual es única, y que posteriormente la diafonía interaural va a ser cancelada mediante un procesado digital de señal. ABSTRACT. This final project is about the ambiophonic recording and playback system, note that this system and the technology it uses is of public domain. Ambiophonics is based on an amalgam of recent research and to the well known and binaural psychoacoustic principles. These advances have expanded new frontiers with regard to the audio recording and playback, as well as to present the listener a sound field at the entrance of their ears as close as possible to the sound field that would the listener be exposed to at the time and place of the mucial interpretation, so we can say that ambiophonics reconstructs a binaural sound field . This system has been developed, in a fairly satisfactory way, thanks to all the studies and previous texts on psychoacoustics and human listening mechanism. Another factor by which it is possible and affordable, both the development and the enjoyment of this technology, is the fact that in our days is inexpensive to usres to own computers that are powerful and fast enough to perform the signal processing that is required in a short time. The developers of this technology have published several documents and downloadable files on the network with applications for ambiophonics system implementation for free. Ambiophonics is based on a combination of factors ignored or underestimated psychoacousticly and what is already known about the acoustic properties of rooms, including rooms where musical performances take place (auditoriums, theaters, concert halls...), and listening rooms (concet halls, studios controls...). In the practical part of the project will be making a series of musical recordings using both traditional stereo recording techniques and recording techiniques compatible with ambiophonics in order to describe and compare both recording techniques. It will also examine to what extent is subjectively favorable for the listener the fact of considering the playback system properties of ambiophonics during the recording stage. This comparison will give us an idea of how far can we get, in terms of sense of reality to the listener, keeping in mind during the recording process the effects introduced by the response of the ear of the listener, which is unique, and that the subsequently interaural crosstalk will be canceled by a digital signal processing.

Design and Validation of an Optimized MB-OFDM Ultra Wideband Transceiver System

Esta tesis está incluida dentro del campo del campo de Multiband Orthogonal Frequency Division Multiplexing Ultra Wideband (MB-OFDM UWB), el cual ha adquirido una gran importancia en las comunicaciones inalámbricas de alta tasa de datos en la última década. UWB surgió con el objetivo de satisfacer la creciente demanda de conexiones inalámbricas en interiores y de uso doméstico, con bajo coste y alta velocidad. La disponibilidad de un ancho de banda grande, el potencial para alta velocidad de transmisión, baja complejidad y bajo consumo de energía, unido al bajo coste de implementación, representa una oportunidad única para que UWB se convierta en una solución ampliamente utilizada en aplicaciones de Wireless Personal Area Network (WPAN). UWB está definido como cualquier transmisión que ocupa un ancho de banda de más de 20% de su frecuencia central, o más de 500 MHz. En 2002, la Comisión Federal de Comunicaciones (FCC) definió que el rango de frecuencias de transmisión de UWB legal es de 3.1 a 10.6 GHz, con una energía de transmisión de -41.3 dBm/Hz. Bajo las directrices de FCC, el uso de la tecnología UWB puede aportar una enorme capacidad en las comunicaciones de corto alcance. Considerando las ecuaciones de capacidad de Shannon, incrementar la capacidad del canal requiere un incremento lineal en el ancho de banda, mientras que un aumento similar de la capacidad de canal requiere un aumento exponencial en la energía de transmisión. En los últimos años, s diferentes desarrollos del UWB han sido extensamente estudiados en diferentes áreas, entre los cuales, el protocolo de comunicaciones inalámbricas MB-OFDM UWB está considerado como la mejor elección y ha sido adoptado como estándar ISO/IEC para los WPANs. Combinando la modulación OFDM y la transmisión de datos utilizando las técnicas de salto de frecuencia, el sistema MB-OFDM UWB es capaz de soportar tasas de datos con que pueden variar de los 55 a los 480 Mbps, alcanzando una distancia máxima de hasta 10 metros. Se esperara que la tecnología MB-OFDM tenga un consumo energético muy bajo copando un are muy reducida en silicio, proporcionando soluciones de bajo coste que satisfagan las demandas del mercado. Para cumplir con todas estas expectativas, el desarrollo y la investigación del MBOFDM UWB deben enfrentarse a varios retos, como son la sincronización de alta sensibilidad, las restricciones de baja complejidad, las estrictas limitaciones energéticas, la escalabilidad y la flexibilidad. Tales retos requieren un procesamiento digital de la señal de última generación, capaz de desarrollar sistemas que puedan aprovechar por completo las ventajas del espectro UWB y proporcionar futuras aplicaciones inalámbricas en interiores. Esta tesis se centra en la completa optimización de un sistema de transceptor de banda base MB-OFDM UWB digital, cuyo objetivo es investigar y diseñar un subsistema de comunicación inalámbrica para la aplicación de las Redes de Sensores Inalámbricas Visuales. La complejidad inherente de los procesadores FFT/IFFT y el sistema de sincronización así como la alta frecuencia de operación para todos los elementos de procesamiento, se convierten en el cuello de la botella para el diseño y la implementación del sistema de UWB digital en base de banda basado en MB-OFDM de baja energía. El objetivo del transceptor propuesto es conseguir baja energía y baja complejidad bajo la premisa de un alto rendimiento. Las optimizaciones están realizadas tanto a nivel algorítmico como a nivel arquitectural para todos los elementos del sistema. Una arquitectura hardware eficiente en consumo se propone en primer lugar para aquellos módulos correspondientes a núcleos de computación. Para el procesado de la Transformada Rápida de Fourier (FFT/IFFT), se propone un algoritmo mixed-radix, basado en una arquitectura con pipeline y se ha desarrollado un módulo de Decodificador de Viterbi (VD) equilibrado en coste-velocidad con el objetivo de reducir el consumo energético e incrementar la velocidad de procesamiento. También se ha implementado un correlador signo-bit simple basado en la sincronización del tiempo de símbolo es presentado. Este correlador es usado para detectar y sincronizar los paquetes de OFDM de forma robusta y precisa. Para el desarrollo de los subsitemas de procesamiento y realizar la integración del sistema completo se han empleado tecnologías de última generación. El dispositivo utilizado para el sistema propuesto es una FPGA Virtex 5 XC5VLX110T del fabricante Xilinx. La validación el propuesta para el sistema transceptor se ha implementado en dicha placa de FPGA. En este trabajo se presenta un algoritmo, y una arquitectura, diseñado con filosofía de co-diseño hardware/software para el desarrollo de sistemas de FPGA complejos. El objetivo principal de la estrategia propuesta es de encontrar una metodología eficiente para el diseño de un sistema de FPGA configurable optimizado con el empleo del mínimo esfuerzo posible en el sistema de procedimiento de verificación, por tanto acelerar el periodo de desarrollo del sistema. La metodología de co-diseño presentada tiene la ventaja de ser fácil de usar, contiene todos los pasos desde la propuesta del algoritmo hasta la verificación del hardware, y puede ser ampliamente extendida para casi todos los tipos de desarrollos de FPGAs. En este trabajo se ha desarrollado sólo el sistema de transceptor digital de banda base por lo que la comprobación de señales transmitidas a través del canal inalámbrico en los entornos reales de comunicación sigue requiriendo componentes RF y un front-end analógico. No obstante, utilizando la metodología de co-simulación hardware/software citada anteriormente, es posible comunicar el sistema de transmisor y el receptor digital utilizando los modelos de canales propuestos por IEEE 802.15.3a, implementados en MATLAB. Por tanto, simplemente ajustando las características de cada modelo de canal, por ejemplo, un incremento del retraso y de la frecuencia central, podemos estimar el comportamiento del sistema propuesto en diferentes escenarios y entornos. Las mayores contribuciones de esta tesis son: • Se ha propuesto un nuevo algoritmo 128-puntos base mixto FFT usando la arquitectura pipeline multi-ruta. Los complejos multiplicadores para cada etapa de procesamiento son diseñados usando la arquitectura modificada shiftadd. Los sistemas word length y twiddle word length son comparados y seleccionados basándose en la señal para cuantización del SQNR y el análisis de energías. • El desempeño del procesador IFFT es analizado bajo diferentes situaciones aritméticas de bloques de punto flotante (BFP) para el control de desbordamiento, por tanto, para encontrar la arquitectura perfecta del algoritmo IFFT basado en el procesador FFT propuesto. • Para el sistema de receptor MB-OFDM UWB se ha empleado una sincronización del tiempo innovadora, de baja complejidad y esquema de compensación, que consiste en funciones de Detector de Paquetes (PD) y Estimación del Offset del tiempo. Simplificando el cross-correlation y maximizar las funciones probables solo a sign-bit, la complejidad computacional se ve reducida significativamente. • Se ha propuesto un sistema de decodificadores Viterbi de 64 estados de decisión-débil usando velocidad base-4 de arquitectura suma-comparaselecciona. El algoritmo Two-pointer Even también es introducido en la unidad de rastreador de origen con el objetivo de conseguir la eficiencia en el hardware. • Se han integrado varias tecnologías de última generación en el completo sistema transceptor basebanda , con el objetivo de implementar un sistema de comunicación UWB altamente optimizado. • Un diseño de flujo mejorado es propuesto para el complejo sistema de implementación, el cual puede ser usado para diseños de Cadena de puertas de campo programable general (FPGA). El diseño mencionado no sólo reduce dramáticamente el tiempo para la verificación funcional, sino también provee un análisis automático como los errores del retraso del output para el sistema de hardware implementado. • Un ambiente de comunicación virtual es establecido para la validación del propuesto sistema de transceptores MB-OFDM. Este método es provisto para facilitar el uso y la conveniencia de analizar el sistema digital de basebanda sin parte frontera analógica bajo diferentes ambientes de comunicación. Esta tesis doctoral está organizada en seis capítulos. En el primer capítulo se encuentra una breve introducción al campo del UWB, tanto relacionado con el proyecto como la motivación del desarrollo del sistema de MB-OFDM. En el capítulo 2, se presenta la información general y los requisitos del protocolo de comunicación inalámbrica MBOFDM UWB. En el capítulo 3 se habla de la arquitectura del sistema de transceptor digital MB-OFDM de banda base . El diseño del algoritmo propuesto y la arquitectura para cada elemento del procesamiento está detallado en este capítulo. Los retos de diseño del sistema que involucra un compromiso de discusión entre la complejidad de diseño, el consumo de energía, el coste de hardware, el desempeño del sistema, y otros aspectos. En el capítulo 4, se ha descrito la co-diseñada metodología de hardware/software. Cada parte del flujo del diseño será detallado con algunos ejemplos que se ha hecho durante el desarrollo del sistema. Aprovechando esta estrategia de diseño, el procedimiento de comunicación virtual es llevado a cabo para probar y analizar la arquitectura del transceptor propuesto. Los resultados experimentales de la co-simulación y el informe sintético de la implementación del sistema FPGA son reflejados en el capítulo 5. Finalmente, en el capítulo 6 se incluye las conclusiones y los futuros proyectos, y también los resultados derivados de este proyecto de doctorado. ABSTRACT In recent years, the Wireless Visual Sensor Network (WVSN) has drawn great interest in wireless communication research area. They enable a wealth of new applications such as building security control, image sensing, and target localization. However, nowadays wireless communication protocols (ZigBee, Wi-Fi, and Bluetooth for example) cannot fully satisfy the demands of high data rate, low power consumption, short range, and high robustness requirements. New communication protocol is highly desired for such kind of applications. The Ultra Wideband (UWB) wireless communication protocol, which has increased in importance for high data rate wireless communication field, are emerging as an important topic for WVSN research. UWB has emerged as a technology that offers great promise to satisfy the growing demand for low-cost, high-speed digital wireless indoor and home networks. The large bandwidth available, the potential for high data rate transmission, and the potential for low complexity and low power consumption, along with low implementation cost, all present a unique opportunity for UWB to become a widely adopted radio solution for future Wireless Personal Area Network (WPAN) applications. UWB is defined as any transmission that occupies a bandwidth of more than 20% of its center frequency, or more than 500 MHz. In 2002, the Federal Communications Commission (FCC) has mandated that UWB radio transmission can legally operate in the range from 3.1 to 10.6 GHz at a transmitter power of －41.3 dBm/Hz. Under the FCC guidelines, the use of UWB technology can provide enormous capacity over short communication ranges. Considering Shannon’s capacity equations, increasing the channel capacity requires linear increasing in bandwidth, whereas similar channel capacity increases would require exponential increases in transmission power. In recent years, several different UWB developments has been widely studied in different area, among which, the MB-OFDM UWB wireless communication protocol is considered to be the leading choice and has recently been adopted in the ISO/IEC standard for WPANs. By combing the OFDM modulation and data transmission using frequency hopping techniques, the MB-OFDM UWB system is able to support various data rates, ranging from 55 to 480 Mbps, over distances up to 10 meters. The MB-OFDM technology is expected to consume very little power and silicon area, as well as provide low-cost solutions that can satisfy consumer market demands. To fulfill these expectations, MB-OFDM UWB research and development have to cope with several challenges, which consist of high-sensitivity synchronization, low- complexity constraints, strict power limitations, scalability, and flexibility. Such challenges require state-of-the-art digital signal processing expertise to develop systems that could fully take advantages of the UWB spectrum and support future indoor wireless applications. This thesis focuses on fully optimization for the MB-OFDM UWB digital baseband transceiver system, aiming at researching and designing a wireless communication subsystem for the Wireless Visual Sensor Networks (WVSNs) application. The inherent high complexity of the FFT/IFFT processor and synchronization system, and high operation frequency for all processing elements, becomes the bottleneck for low power MB-OFDM based UWB digital baseband system hardware design and implementation. The proposed transceiver system targets low power and low complexity under the premise of high performance. Optimizations are made at both algorithm and architecture level for each element of the transceiver system. The low-power hardwareefficient structures are firstly proposed for those core computation modules, i.e., the mixed-radix algorithm based pipelined architecture is proposed for the Fast Fourier Transform (FFT/IFFT) processor, and the cost-speed balanced Viterbi Decoder (VD) module is developed, in the aim of lowering the power consumption and increasing the processing speed. In addition, a low complexity sign-bit correlation based symbol timing synchronization scheme is presented so as to detect and synchronize the OFDM packets robustly and accurately. Moreover, several state-of-the-art technologies are used for developing other processing subsystems and an entire MB-OFDM digital baseband transceiver system is integrated. The target device for the proposed transceiver system is Xilinx Virtex 5 XC5VLX110T FPGA board. In order to validate the proposed transceiver system in the FPGA board, a unified algorithm-architecture-circuit hardware/software co-design environment for complex FPGA system development is presented in this work. The main objective of the proposed strategy is to find an efficient methodology for designing a configurable optimized FPGA system by using as few efforts as possible in system verification procedure, so as to speed up the system development period. The presented co-design methodology has the advantages of easy to use, covering all steps from algorithm proposal to hardware verification, and widely spread for almost all kinds of FPGA developments. Because only the digital baseband transceiver system is developed in this thesis, the validation of transmitting signals through wireless channel in real communication environments still requires the analog front-end and RF components. However, by using the aforementioned hardware/software co-simulation methodology, the transmitter and receiver digital baseband systems get the opportunity to communicate with each other through the channel models, which are proposed from the IEEE 802.15.3a research group, established in MATLAB. Thus, by simply adjust the characteristics of each channel model, e.g. mean excess delay and center frequency, we can estimate the transmission performance of the proposed transceiver system through different communication situations. The main contributions of this thesis are: • A novel mixed radix 128-point FFT algorithm by using multipath pipelined architecture is proposed. The complex multipliers for each processing stage are designed by using modified shift-add architectures. The system wordlength and twiddle word-length are compared and selected based on Signal to Quantization Noise Ratio (SQNR) and power analysis. • IFFT processor performance is analyzed under different Block Floating Point (BFP) arithmetic situations for overflow control, so as to find out the perfect architecture of IFFT algorithm based on the proposed FFT processor. • An innovative low complex timing synchronization and compensation scheme, which consists of Packet Detector (PD) and Timing Offset Estimation (TOE) functions, for MB-OFDM UWB receiver system is employed. By simplifying the cross-correlation and maximum likelihood functions to signbit only, the computational complexity is significantly reduced. • A 64 state soft-decision Viterbi Decoder system by using high speed radix-4 Add-Compare-Select architecture is proposed. Two-pointer Even algorithm is also introduced into the Trace Back unit in the aim of hardware-efficiency. • Several state-of-the-art technologies are integrated into the complete baseband transceiver system, in the aim of implementing a highly-optimized UWB communication system. • An improved design flow is proposed for complex system implementation which can be used for general Field-Programmable Gate Array (FPGA) designs. The design method not only dramatically reduces the time for functional verification, but also provides automatic analysis such as errors and output delays for the implemented hardware systems. • A virtual communication environment is established for validating the proposed MB-OFDM transceiver system. This methodology is proved to be easy for usage and convenient for analyzing the digital baseband system without analog frontend under different communication environments. This PhD thesis is organized in six chapters. In the chapter 1 a brief introduction to the UWB field, as well as the related work, is done, along with the motivation of MBOFDM system development. In the chapter 2, the general information and requirement of MB-OFDM UWB wireless communication protocol is presented. In the chapter 3, the architecture of the MB-OFDM digital baseband transceiver system is presented. The design of the proposed algorithm and architecture for each processing element is detailed in this chapter. Design challenges of such system involve trade-off discussions among design complexity, power consumption, hardware cost, system performance, and some other aspects. All these factors are analyzed and discussed. In the chapter 4, the hardware/software co-design methodology is proposed. Each step of this design flow will be detailed by taking some examples that we met during system development. Then, taking advantages of this design strategy, the Virtual Communication procedure is carried out so as to test and analyze the proposed transceiver architecture. Experimental results from the co-simulation and synthesis report of the implemented FPGA system are given in the chapter 5. The chapter 6 includes conclusions and future work, as well as the results derived from this PhD work.

Photonic processing subsystem based on visual cortex architecture

Optical signal processing in any living being is more complex than the one obtained in artificial systems. Cortex architecture, although only partly known, gives some useful ideas to be employed in communications. To analyze some of these structures is the objective of this paper. One of the main possibilities reported is handling signals in a parallel way. As it is shown, according to the signal characteristics each signal impinging onto a single input may be routed to a different output. At the same time, identical signals, coming to different inputs, may be routed to the same output without internal conflicts. This is due to the change of some of their characteristics in the way out when going through the intermediate levels. The simulation of this architecture is based on simple logic cells. The basis for the proposed architecture is the five layers of the mammalian retina and the first levels of the visual cortex.

Low-power vertical cavity NAND gate

The study of the Vertical-Cavity Semiconductor Optical Amplifiers (VCSOAs) for optical signal processing applications is increasing his interest. Due to their particular structure, the VCSOAs present some advantages when compared to their edge-emitting counterparts including low manufacturing costs, high coupling efficiency to optical fibers and the ease to fabricate 2-D arrays of this kind of devices. As a consequence, all-optical logic gates based on VCSOAs may be very promising devices for their use in optical computing and optical switching in communications. Moreover, since all the boolean logic functions can be implemented by combining NAND logic gates, the development of a Vertical-Cavity NAND gate would be of particular interest. In this paper, the characteristics of the dispersive optical bistability appearing on a VCSOA operated in reflection are studied. A progressive increment of the number of layers compounding the top Distributed Bragg Reflector (DBR) of the VCSOA results on a change on the shape of the appearing bistability from an S-shape to a clockwise bistable loop. This resulting clockwise bistability has high on-off contrast ratio and input power requirements one order of magnitude lower than those needed for edge-emitting devices. Based on these results, an all-optical vertical-cavity NAND gate with high on-off contrast ratio and an input power for operation of only 10|i\V will be reported in this paper.