The challenges of measuring speech signals in the presence of a strong background noise cannot be easily addressed with traditional acoustic technology. A recent solution to the problem considers combining acoustic sensor measurements with real-time, non-acoustic detection of an aspect of the speech production process. While significant advancements have been made in that area using low-power radar-based techniques, drawbacks inherent to the operation of such sensors are yet to be surmounted. Therefore, one imperative scientific objective is to devise new, non-invasive non-acoustic sensor topologies that offer improvements regarding sensitivity, robustness, and acoustic bandwidth. This project investigates a novel design that directly senses the glottal flow waveform by measuring variations in the electromagnetic properties of neck tissues during voiced segments of speech. The approach is to explore two distinct sensor configurations, namely the“six-element" and the“parallel-plate" resonator. The research focuses on the modeling aspect of the biological load and the resonator prototypes using multi-transmission line (MTL) and finite element (FE) simulation tools. Finally, bench tests performed with both prototypes on phantom loads as well as human subjects are presented.

This study embarks on exploiting the Thickness Shear Mode (TSM) acoustic wave sensor and the ElectroMagnetic Piezoelectric Acoustic Sensor (EMPAS) towards the study of aptamer-to-cocaine binding in a label-free direct approach. The high sensitivity and selectivity offered by the EMPAS in combination with alkyltrichlorosilane-based self-assembled monolayers proved superior towards the detection of cocaine. The most efficient method for the attachment of the aptamers onto the sensor surface to construct highly dense populations of the aptamer molecules with retained biomolecule activity is shown to be dependent on the composition of immobilizing solution and on the amount of spacing provided in the plane of the aptamer molecules. The distinct ligand-induced binding mechanisms and regeneration capabilities of the two anti-cocaine aptamers are monitored with the EMPAS. Utilizing this sensor to monitor cocaine-aptamer interactions will serve as the first piezoelectric aptasensor for the detection of a small molecule.

This study embarks on exploiting the Thickness Shear Mode (TSM) acoustic wave sensor and the ElectroMagnetic Piezoelectric Acoustic Sensor (EMPAS) towards the study of aptamer-to-cocaine binding in a label-free direct approach. The high sensitivity and selectivity offered by the EMPAS in combination with alkyltrichlorosilane-based self-assembled monolayers proved superior towards the detection of cocaine. The most efficient method for the attachment of the aptamers onto the sensor surface to construct highly dense populations of the aptamer molecules with retained biomolecule activity is shown to be dependent on the composition of immobilizing solution and on the amount of spacing provided in the plane of the aptamer molecules. The distinct ligand-induced binding mechanisms and regeneration capabilities of the two anti-cocaine aptamers are monitored with the EMPAS. Utilizing this sensor to monitor cocaine-aptamer interactions will serve as the first piezoelectric aptasensor for the detection of a small molecule.

The master´s thesis describes the basic information on the reporting of water losses. Means and methods to reduce water losses. Devices used to monitoring water supply, or detect malfunctioning of water supply line. Describes and evaluates the functionality of UFR on the laboratory model.

[ES] Los sistemas de audio han experimentado un gran desarrollo en los últimos años gracias al aumento de dispositivos con procesadores de alto rendimiento capaces de realizar un procesamiento cada vez más eficiente. Además, las comunicaciones inalámbricas permiten a los dispositivos de una red estar ubicados en diferentes lugares sin limitaciones físicas. La combinación de estas tecnologías ha dado lugar a la aparición de las redes de sensores acústicos (ASN). Una ASN está compuesta por nodos equipados con transductores de audio, como micrófonos o altavoces. En el caso de la monitorización acústica del campo, sólo es necesario incorporar sensores acústicos a los nodos ASN. Sin embargo, en el caso de las aplicaciones de control, los nodos deben interactuar con el campo acústico a través de altavoces. La ASN puede implementarse mediante dispositivos de bajo coste, como Raspberry Pi o dispositivos móviles, capaces de gestionar varios micrófonos y altavoces y de ofrecer una buena capacidad de cálculo. Además, estos dispositivos pueden comunicarse mediante conexiones inalámbricas, como Wi-Fi o Bluetooth. Por lo tanto, en esta tesis, se propone una ASN compuesta por dispositivos móviles conectados a altavoces inalámbricos mediante un enlace Bluetooth. Además, el problema de la sincronización entre los dispositivos de una ASN es uno de los principales retos a abordar, ya que el rendimiento del procesamiento de audio es muy sensible a la falta de sincronismo. Por lo tanto, también se lleva a cabo un análisis del problema de sincronización entre dispositivos conectados a altavoces inalámbricos en una ASN. En este sentido, una de las principales aportaciones es el análisis de la latencia de audio cuando los nodos acústicos de la ASN están formados por dispositivos móviles que se comunican altavoces mediante enlaces Bluetooth. Una segunda contribución significativa de esta tesis es la implementación de un método para sincronizar los diferentes dispositivos de una ASN, junto con un estudio de sus limitaciones. Por último, se ha introducido el método propuesto para implementar aplicaciones de zonas sonoras personales (PSZ). Por lo tanto, la implementación y el análisis del rendimiento de diferentes aplicaciones de audio sobre una ASN compuesta por dispositivos móviles y altavoces inalámbricos es también una contribución significativa en el área de las ASN. Cuando el entorno acústico afecta negativamente a la percepción de la señal de audio emitida por los altavoces de la ASN, se uti­lizan técnicas de ecualización para mejorar la percepción de la señal de audio. Para ello, en esta tesis se implementa un sistema de ecualización inteligente. Para ello, se emplean algoritmos psicoacústicos para implementar un procesamiento inteligente basado en el sis­tema auditivo humano capaz de adaptarse a los cambios del entorno. Por ello, otra contribución importante de esta tesis es el análisis del enmas­caramiento espectral entre dos sonidos complejos. Este análisis permitirá calcular el umbral de enmascaramiento de un sonido con más precisión que los métodos utilizados actualmente. Este método se utiliza para implementar una aplicación de ecualización perceptiva que pretende mejorar la percepción de la señal de audio en presencia de un ruido ambien­tal. Para ello, esta tesis propone dos algoritmos de ecualización diferentes: 1) la pre-ecualización de la señal de audio para que se perciba por encima del umbral de enmascaramiento del ruido ambiental y 2) diseñar un con­trol de ruido ambiental perceptivo en los sistemas de ecualización activa de ruido (ANE), para que el nivel de ruido ambiental percibido esté por debajo del umbral de enmascaramiento de la señal de audio. Por lo tanto, la ultima aportación de esta tesis es la implementación de una aplicación de ecualización perceptiva con los dos diferentes algorit­mos de ecualización embebidos y el análisis de su rendimiento a través del banco de pruebas realizado en el laboratorio GTAC-iTEAM. / [CA] El sistemes de so han experimentat un gran desenvolupament en els últims anys gràcies a l'augment de dispositius amb processadors d'alt rendiment capaços de realitzar un processament d'àudio cada vegada més eficient. D'altra banda, l'expansió de les comunicacions inalàmbriques ha permès implementar xarxes en les quals els dispositius poden estar situats a difer­ents llocs sense limitacions físiques. La combinació d'aquestes tecnologies ha donat lloc a l'aparició de les xarxes de sensors acústics (ASN). Una ASN està composta per nodes equipats amb transductors d'àudio, com micr`ofons o altaveus. En el cas del monitoratge del camp acústic, només cal incorporar sensors acústics als nodes de l'ASN. No obstant això, en el cas de les aplicacions de control, els nodes han d'interactuar amb el camp acústic a través d'altaveus. Una ASN pot implementar-se mitjant¿cant dispositius de baix cost, com ara Raspberry Pi o dispositius mòbils, capaços de gestionar di­versos micròfons i altaveus i d'oferir una bona capacitat computacional. A més, aquests dispositius poden comunicar-se a través de connexions inalàmbriques, com Wi-Fi o Bluetooth. Per això, en aquesta tesi es proposa una ASN composta per dispositius mòbils connectats a altaveus inalàmbrics a través d'un enllaç Bluetooth. El problema de la sincronització entre els dispositius d'una ASN és un dels principals reptes a abordar ja que el rendiment del processament d'àudio és molt sensible a la falta de sincronisme. Per tant, també es duu a terme una anàlisi profunda del problema de la sincronització entre els dispositius comercials connectats als altaveus inalàmbrics en una ASN. En aquest sentit, una de les principals contribucions és l'anàlisi de la latència d'àudio quan els nodes acústics en l'ASN estan compostos per dispositius mòbils que es comuniquen amb els altaveus corresponents mitjançant enllaços Bluetooth. Una segona contribuciò sig­nificativa d'aquesta tesi és la implementació d'un mètode per sincronitzar els diferents dispositius d'una ASN, juntament amb un estudi de les seves limitacions. Finalment, s'ha introduït el mètode proposat per implemen­tar aplicacions de zones de so personal. Per tant, la implementació i l'anàlisi del rendiment de diferents aplicacions d'àudio sobre una ASN composta per dispositius mòbils i al­taveus inalàmbrics és també una contribució significativa a l'àrea de les ASN. Quan l'entorn acústic afecta negativament a la percepció del senyal d'àudio emesa pels altaveus de l'ASN, es fan servir tècniques d'equalització per a millorar la percepció del senyal d'àudio. En consequència, en aquesta tesi s'implementa un sistema d'equalització intel·ligent. Per això, s'utilitzen algoritmes psicoacústics per implementar un processament intel·ligent basat en el sistema audi­tiu humà capaç d'adaptar-se als canvis de l'entorn. Per aquest motiu, una altra contribució important d'aquesta tesi és l'anàlisi de l'emmascarament espectral entre dos sons complexos. Aquesta anàlisi permetrà calcular el llindar d'emmascarament d'un so sobre amb més precisió que els mètodes utilitzats actualment. Aquest mètode s'utilitza per a imple­mentar una aplicació d'equalització perceptual que pretén millorar la per­cepció del senyal d'àudio en presència d'un soroll ambiental. Per això, aquesta tesi proposa dos algoritmes d'equalització diferents: 1) la pree­qualització del senyal d'àudio perquè es percebi per damunt del llindar d'emmascarament del soroll ambiental i 2) dissenyar un control de soroll ambiental perceptiu en els sistemes d'equalització activa de soroll (ANE) de manera que el nivell de soroll ambiental percebut estiga per davall del llindar d'emmascarament del senyal d'àudio. Per tant, l'última aportació d'aquesta tesi és la implementació d'una aplicació d'equalització perceptiva amb els dos algoritmes d'equalització embeguts i l'anàlisi del seu rendiment a través del banc de proves realitzat al laboratori GTAC-iTEAM. / [EN] Audio systems have been extensively developed in recent years thanks to the increase of devices with high-performance processors able to per­form more efficient processing. In addition, wireless communications allow devices in a network to be located in different places without physical limitations. The combination of these technologies has led to the emergence of Acoustic Sensor Networks (ASN). An ASN is com­posed of nodes equipped with audio transducers, such as microphones or speakers. In the case of acoustic field monitoring, only acoustic sensors need to be incorporated into the ASN nodes. However, in the case of control applications, the nodes must interact with the acoustic field through loudspeakers. ASN can be implemented through low-cost devices, such as Rasp­berry Pi or mobile devices, capable of managing multiple mi­crophones and loudspeakers and offering good computational capacity. In addition, these devices can communicate through wireless connections, such as Wi-Fi or Bluetooth. Therefore, in this dissertation, an ASN composed of mobile devices connected to wireless speak­ers through a Bluetooth link is proposed. Additionally, the problem of syn­chronization between the devices in an ASN is one of the main challenges to be addressed since the audio processing performance is very sensitive to the lack of synchronism. Therefore, an analysis of the synchroniza­tion problem between devices connected to wireless speakers in an ASN is also carried out. In this regard, one of the main contributions is the analysis of the audio latency of mobile devices when the acoustic nodes in the ASN are comprised of mobile devices communicating with the corresponding loudspeakers through Bluetooth links. A second significant contribution of this dissertation is the implementation of a method to synchronize the different devices of an ASN, together with a study of its limitations. Finally, the proposed method has been introduced in order to implement personal sound zones (PSZ) applications. Therefore, the imple­mentation and analysis of the performance of different audio applications over an ASN composed of mobile devices and wireless speakers is also a significant contribution in the area of ASN. In cases where the acoustic environment negatively affects the percep­tion of the audio signal emitted by the ASN loudspeakers, equalization techniques are used with the objective of enhancing the perception thresh­old of the audio signal. For this purpose, a smart equalization system is implemented in this dissertation. In this regard, psychoacous­tic algorithms are employed to implement a smart processing based on the human hearing system capable of adapting to changes in the envi­ronment. Therefore, another important contribution of this thesis focuses on the analysis of the spectral masking between two complex sounds. This analysis will allow to calculate the masking threshold of one sound over the other in a more accurate way than the currently used methods. This method is used to implement a perceptual equalization application that aims to improve the perception threshold of the audio signal in presence of ambient noise. To this end, this thesis proposes two different equalization algorithms: 1) pre-equalizing the audio signal so that it is perceived above the ambient noise masking threshold and 2) designing a perceptual control of ambient noise in active noise equalization (ANE) systems, so that the perceived ambient noise level is below the masking threshold of the audio signal. Therefore, the last contribution of this dissertation is the imple­mentation of a perceptual equalization application with the two different embedded equalization algorithms and the analysis of their performance through the testbed carried out in the GTAC-iTEAM laboratory. / This work has received financial support of the following projects: • SSPRESING: Smart Sound Processing for the Digital Living (Reference: TEC2015-67387-C4-1-R. Entity: Ministerio de Economia y Empresa. Spain). • FPI: Ayudas para contratos predoctorales para la formación de doctores (Reference: BES-2016-077899. Entity: Agencia Estatal de Investigación. Spain). DANCE: Dynamic Acoustic Networks for Changing Environments (Reference: RTI2018-098085-B-C41-AR. Entity: Agencia Estatal de Investigación. Spain). • DNOISE: Distributed Network of Active Noise Equalizers for Multi-User Sound Control (Reference: H2020-FETOPEN-4-2016-2017. Entity: I+D Colaborativa competitiva. Comisión de las comunidades europea). / Estreder Campos, J. (2022). Smart Sound Control in Acoustic Sensor Networks: a Perceptual Perspective [Tesis doctoral]. Universitat Politècnica de València. https://doi.org/10.4995/Thesis/10251/181597

Dans ce travail, une approche de reconnaissance de l’orateur en utilisant un microphone de contact est développée et présentée. L'élément passif de contact est construit à partir d'un matériau piézoélectrique. La position du transducteur piézoélectrique sur le cou de l'individu peut affecter grandement la qualité du signal recueilli et par conséquent les informations qui en sont extraites. Ainsi, le milieu multicouche dans lequel les vibrations des cordes vocales se propagent avant d'être détectées par le transducteur est modélisé. Le meilleur emplacement sur le cou de l’individu pour attacher un élément transducteur particulier est déterminé en mettant en œuvre des techniques de simulation Monte Carlo et, par conséquent, les résultats de la simulation sont vérifiés en utilisant des expériences réelles. La reconnaissance est basée sur le signal généré par les vibrations des cordes vocales lorsqu'un individu parle et non sur le signal vocal à la sortie des lèvres qui est influencé par les résonances dans le conduit vocal. Par conséquent, en raison de la nature variable du signal recueilli, l'analyse a été effectuée en appliquant la technique de transformation de Fourier à court terme pour décomposer le signal en ses composantes de fréquence. Ces fréquences représentent les vibrations des cordes vocales (50-1000 Hz). Les caractéristiques en termes d'intervalle de fréquences sont extraites du spectrogramme résultant. Ensuite, un vecteur 1-D est formé à des fins d'identification. L'identification de l’orateur est effectuée en utilisant deux critères d'évaluation qui sont la mesure de la similarité de corrélation et l'analyse en composantes principales (ACP) en conjonction avec la distance euclidienne. Les résultats montrent qu'un pourcentage élevé de reconnaissance est atteint et que la performance est bien meilleure que de nombreuses techniques existantes dans la littérature. / In this work, a speaker recognition approach using a contact microphone is developed and presented. The contact passive element is constructed from a piezoelectric material. In this context, the position of the piezoelectric transducer on the individual’s neck may greatly affect the quality of the collected signal and consequently the information extracted from it. Thus, the multilayered medium in which the sound propagates before being detected by the transducer is modeled. The best location on the individual’ neck to place a particular transducer element is determined by implementing Monte Carlo simulation techniques and consequently, the simulation results are verified using real experiments. The recognition is based on the signal generated from the vocal cords’ vibrations when an individual is speaking and not on the vocal signal at the output of the lips that is influenced by the resonances in the vocal tract. Therefore, due to the varying nature of the collected signal, the analysis was performed by applying the Short Term Fourier Transform technique to decompose the signal into its frequency components. These frequencies represent the vocal folds’ vibrations (50-1000 Hz). The features in terms of frequencies’ interval are extracted from the resulting spectrogram. Then, a 1-D vector is formed for identification purposes. The identification of the speaker is performed using two evaluation criteria, namely, the correlation similarity measure and the Principal Component Analysis (PCA) in conjunction with the Euclidean distance. The results show that a high percentage of recognition is achieved and the performance is much better than many existing techniques in the literature.

Este trabajo de tesis recoge las contribuciones aportadas en el campo de los sensores de fibra óptica en tres ámbitos de trabajo. Un sistema sensor de alta temperatura sin contacto con transductores de fibras ópticas de sílice se ha presentado. Dos topologías de "ojo abierto" y "ojo cerrado" y un algoritmo de decodificación "pirometría de banda dual" propuesto en este trabajo son las novedosas aportaciones presentadas. El sistema ha sido validado en laboratorio y en pruebas de campo de forma exitosa. Para sensores de temperatura con fibras dopadas con Erbio se ha demostrado que su máxima sensibilidad tiene lugar cuando la longitud utilizada es superior a la óptima exigida por el bombeo. Finalmente se ha modelado y validado en laboratorio una red de sensores acústicos basados en interferómetros Fabry-Perot de cavidad larga determinando a partir del modelo obtenido la fase mínima detectable y la visibilidad máxima del sistema. / The thesis collects contributions in the field of optic fibre sensors for three areas. A high temperature sensor system without contact based on silica optic fibre transducers has been submitted. Two topologies "open eye" and "closed eye" and a dual band pyrometer decoding algorithm are the submitted innovative contributions. The system has been validated successful in the laboratory and field tests so. Another contribution has been the demonstration that the maximum sensitivity for temperature sensors based on Er-doped fibres occurs when the used length is more than the optimum required by pumping. Finally a network of acoustic sensors based on Fabry-Perot fibre interferometers has been modelled and validated in the laboratory. The minimum detectable phase and maximum visibility of the system has been obtained from the proposed and validated model

The aim of master's thesis is to get familiarized with the problems of measurement and analysis of seismic waves. Theoretical part deals with the description of seismic waves, especially their types, sources and properties. Attention was afterwards focused on the measurement systems of these waves, emphasis was placed on their principles and advantages. The practical part discusses methods of noise reduction and highlighting of significant events in measured data. At the end, individual methods are implemented into user-friendly graphical interface.

Les réseaux de capteurs acoustiques sous-marins (UW-ASN) sont les plus nouveaux achèvements technologiques en termes de communication. Les UW-ASN visent à observer et à explorer les lacs, les rivières, les mers et les océans. Récemment, ils ont été soumis à une attention particulière en raison de leur grand potentiel en termes d'applications prometteuses dans divers domaines (militaires, environnementaux, scientifiques ...) et aux nouvelles questions scientifiques qu'ils suscitent. Un problème majeur dans les UW-ASN est l'épuisement rapide de l'énergie, car une grande puissance est nécessaire pour la communication acoustique, tandis que le budget de la batterie des capteurs est limité. Par conséquent, les protocoles de communication énergétiques revêtent une importance primordiale pour faire usage judiciaire du budget énergétique disponible. Dans ce contexte, cette thèse vise à étudier les principales caractéristiques des capteurs acoustiques sous-marins difficiles afin de concevoir des protocoles de communication énergétiques, plus spécifiquement au niveau routage et MAC. Tout d'abord, nous abordons le problème des trous énergétiques dans UW-ASN. Le problème du « sink-hole » se produit lorsque les capteurs les plus proches du sink épuisent leur énergie plus rapidement en raison de leur charge plus lourde. En effet, ces capteurs, en particulier ceux qui sont à un seul saut du sinkstatique, agissent comme des relais pour tous les autres capteurs, ce qui leur épuise sévèrement l’énergie.A la couche de routage,en particulier, nous proposons de distribuer la charge transmise par chaque capteur parmi plusieurs voisins potentiels, en supposant que les capteurs peuvent ajuster leur gamme de communication entre deux niveaux lorsqu'ils envoient ou transmettent des données. Plus précisément, nous déterminons pour chaque capteur l'ensemble des prochains sauts avec les poids de charge associés qui entraînent un épuisement équitable d'énergie entre tous les capteurs du réseau. Ensuite, nous étendons notre stratégie de routage équilibrée en supposant que chaque capteur n'est pas seulement capable d'ajuster sa puissance d'émission à 2 niveaux mais aussi jusqu'à n niveaux où n> 2. Par conséquent, à la couche de routage, pour chaque valeur possible de n, nous déterminons pour chaque capteur l'ensemble des éventuels sauts avec les poids de charge associés qui mènent à une consommation d'énergie équitable chez tous les capteurs du réseau. En outre, nous obtenons le nombre optimal de puissances de transmission n qui équilibre la consommation d'énergie de tous les capteurs pour chaque configuration de réseau. En plus de cela, il convient de souligner que notre protocole de routage étendu utilise un modèle de canal à variation de temps plus réaliste qui tient compte de la plupart des caractéristiques fondamentales de la propagation acoustique sous-marine. Les résultats analytiques montrent que notre protocole de routage assure une réduction importante de la consommation d’énergie. Deuxièmement, pour atténuer les impacts de collision spectaculaires gaspillant l’énergie, nous concevons un protocole MAC multicanal (MC-UWMAC) évitant les collisions pour les UW-ASNs. MC-UWMAC fonctionne avec un canal de contrôle (décomposé en créneaux de temps) et un ensemble de canaux de données à bande passante égale. Les créneaux du canal de contrôle sont dédiés à l’échange RTS / CTS permettant à une paire de capteurs communicants de s'accorder sur l'heure de début de la communication sur un canal de données pré-alloué. Dans cette thèse, nous proposons deux nouvelles procédures associées d'allocation des créneaux du canal de contrôle et d'attribution des canaux de données sans nécessiter de frais de négociation supplémentaires. En conséquence, chaque capteur peut initier l'échange RTS / CTS uniquement à son créneau assigné, calculé à l'aide d'une procédure d'allocation basée sur une partition virtuelle de grille de la zone de déploiement. (...) / UnderWaterAcoustic Sensor Networks (UW-ASNs) are the newest technological achievement in terms of communication. Composed of a set of communicating underwater sensors, UW-ASNs are intended to observe and explore lakes, rivers, seas and oceans. Recently, they have been subject to a special attention due to their great potential in terms of promising applications in various domains (military, environmental, scientific...) and to the new scientific issues they raise. A great challenging issue in UW-ASNs is the fast energy depletion since high power is needed for acoustic communication while sensors battery budget is limited. Hence, energy-efficient networking protocols are of a paramount importance to make judicious use of the available energy budget while considering the distinguishing underwater environment characteristics. In this context, this thesis aims at studying the main challenging underwater acoustic sensors characteristics to design energy-efficient communication protocols specifically at the routing and MAC layers. First, we address the problem of energy holes in UW-ASNs. The sink-hole problem occurs when the closest nodes to sink drain their energy faster due to their heavier load. Indeed, those sensors especially the ones that are 1-hop away from the static sink act as relays to it on behalf of all other sensors, thus suffering from severe energy depletion. In particular, at the routing layer, we propose to distribute the transmission load at each sensor among several potential neighbors, assuming that sensors can adjust their communication range among two levels when they send or forward data. Specifically, we determine for each sensor the set of next hops with the associated load weights that lead to a fair energy depletion among all sensors in the network. Then, we extend our balanced routing strategy by assuming that each sensor node is not only able to adjust its transmission power to 2 levels but eventually up to n levels where n > 2. Consequently, at the routing layer, for each possible value of n, we determine for each sensor the set of possible next hops with the associated load weights that lead to a fair energy consumption among all sensors in the network. Moreover, we derive the optimal number of transmission powers n that balances the energy consumption among all sensors for each network configuration. In addition to that, it is worth pointing out that our extended routing protocol uses a more realistic time varying channel model that takes into account most of the fundamental characteristics of the underwater acoustic propagation. Analytical results show that further energy saving is achieved by our extended routing scheme. Second, to mitigate the dramatic collision impacts, we design a collision avoidance energy efficient multichannel MAC protocol (MC-UWMAC) for UW-ASNs. MC-UWMAC operates on single slotted control and a set of equal-bandwidth data channels. Control channel slots are dedicated to RTS/CTS handshaking allowing a communicating node pair to agree on the start time of communication on a pre-allocated data channel. In this thesis, we propose two novel coupled slot assignment and data channels allocation procedures without requiring any extra negotiation overhead. Accordingly, each node can initiate RTS/CTS exchange only at its assigned slot calculated using a slot allocation procedure based on a grid virtual partition of the deployment area. Moreover, for each communicating pair of nodes, one data channel is allocated using a channel allocation procedure based on our newly designed concept of singleton- intersecting quorum. Accordingly, each pair of communicating nodes will have at their disposal a unique 2-hop conflict free data channel. Compared with existing MAC protocol, MC-UWMAC reduces experienced collisions and improves network throughput while minimizing energy consumption.

[pt] Diferentes tipos de lasers podem ser usados para gerar pulsos de luz com uma ampla faixa de durações de pulso, energias e potências de pico. As técnicas de Q-switching e mode-locked são relatadas há anos por vários autores e pesquisadores e são frequentemente utilizadas na geração de lasers de pulso ultracurto com duração de pulso no domínio do tempo na faixa de nanossegundos até femtossegundos. Uma configuração, com ganho fornecido por um amplificador óptico semicondutor (SOA) e amplificador de fibra dopada com érbio (EDFA) é proposta e emprega a técnica de gerenciamento de dispersão para gerar um trem de pulsos ópticos exibindo alta potência de pico, taxa de repetição ultra-baixa e largura temporal curta, habilitando que este laser seja usado como uma fonte para aplicações de alta resolução em reflectometria óptica no domínio do tempo (OTDR). A operação mode-locked é conhecida por ocorrer apenas em lasers ordenados padrão por um longo tempo e até recentemente foi encontrado também em lasers de fibra aleatórios desordenados (RFL). Embora tenha havido progresso no sentido de travar modos espaciais e longitudinais em lasers aleatórios, a literatura carece de relatos sobre geração de pulsos limitada por transformada de Fourier, apesar das muitas décadas de campo. O autor demonstra experimentalmente um mode-locked random fiber laser (MLRFL) operando como um refletômetro óptico de domínio do tempo sensível à fase. Aqui, a saída total do laser fornece o sinal de detecção, em contraste com o pequeno sinal retroespalhado medido em um OTDR convencional. O laser opera como um sensor acústico distribuído (DAS) e sensor de temperatura distribuído (DTS). / [en] Different types of lasers can be used to generate light pulses with a wide range of pulse durations, energies and peak powers. Q-switching and mode-locked techniques have been reported for years by several authors and researchers and are frequently used in the generation of ultra-short-pulse lasers with time-domain pulse durations from the nanosecond to femtosecond range. A configuration, with gain provided by a semiconductor optical amplifier (SOA) and erbium-doped fiber amplifier (EDFA) is proposed and employ the dispersion management technique to generate a train of optical pulses exhibiting high-peak-power, ultralow repetition rate, and fast temporal width, enabling this laser to be used as a source for high-resolution optical time domain reflectometer (OTDR) applications. The mode-locking operation has been known to occur only in standard ordered lasers for a long time and until recently it was found to also occur in disordered random fiber lasers (RFL). Although progress has been made towards locking spatial and longitudinal modes in random lasers, the literature lacks reports on Fourier transform-limited pulse generation despite the many decades of the field. The author experimentally demonstrates a mode-locked random fiber laser (MLRFL) operating as a lasing phase-sensitive optical time domain reflectometer based on random feedback from a sensing fiber. Here, the full output of the laser provides the sensing signal, in contrast to the small backscattered signal measured in a conventional OTDR. The laser operates as a distributed acoustic sensor (DAS) and distributed temperature sensor (DTS).