Dans les systèmes d'alertes publiques, l’étude de la diffusion des informations dans le réseau est essentielle. Les systèmes de diffusion des messages d'alertes doivent atteindre beaucoup de nœuds en peu de temps. Dans les réseaux de communication basés sur les interactions “device to device”, on s'est récemment beaucoup intéressé à la diffusion des informations et le besoin d'auto-organisation a été mis en évidence. L'auto-organisation conduit à des comportements locaux et des interactions qui ont un effet sur le réseau global et présentent un avantage de scalabilité. Ces réseaux auto-organisés peuvent être autonomes et utiliser peu d'espace mémoire. On peut développer des caractères auto-organisés dans les réseaux de communication en utilisant des idées venant de phénomènes naturels. Il semble intéressant de chercher à obtenir les propriétés des “small world” pour améliorer la diffusion des informations dans le réseau. Dans les modèles de “small world” on réalise un recâblage des liens dans le réseau en changeant la taille et la direction des liens existants. Dans un environnement sans-fils autonome une organisation de ce type peut être créée en utilisant le flocking, l'inhibition latérale et le “beamforming”. Dans ce but, l'auteur utilise d'abord l'analogie avec l'inhibition latérale, le flocking et le “beamforming” pour montrer comment la diffusion des informations peut être améliorée. L'analogue de l'inhibition latérale est utilisé pour créer des régions virtuelles dans le réseau. Puis en utilisant l'analogie avec les règles du flocking, on caractérise les propriétés des faisceaux permettant aux nœuds de communiquer dans les régions. Nous prouvons que les propriétés des “small world” sont vérifiées en utilisant la mesure des moyennes des longueurs des chemins. Cependant l'algorithme proposé est valable pour les réseaux statiques alors que dans les cas introduisant de la mobilité, les concepts d'inhibition latérale et de flocking nécessiteraient beaucoup plus de temps. Dans le cas d'un réseau mobile la structure du réseau change fréquemment. Certaines connexions intermittentes impactent fortement la diffusion des informations. L'auteur utilise le concept de stabilité avec le “beamforming” pour montrer comment on peut améliorer la diffusion des informations. Dans son algorithme il prévoit d'abord la stabilité du nœud en utilisant des informations locales et il utilise ce résultat pour identifier les nœuds qui réaliseront du beamforming. Dans l'algorithme, les nœuds de stabilité faible sont autorisés à faire du beamforming vers les nœuds de forte stabilité. La frontière entre forte et faible stabilité est fixée par un seuil. Cet algorithme ne nécessite pas une connaissance globale du réseau, mais utilise des données locales. Les résultats sont validés en étudiant le temps au bout duquel plus de nœuds reçoivent l'information et en comparant avec d'autres algorithmes de la littérature. Cependant, dans les réseaux réels, les changements de structure ne sont pas dus qu'à la mobilité, mais également à des changements de la densité des nœuds à un moment donné. Pour tenir compte de l'influence de tels événements sur la diffusion des informations concernant la sécurité publique, l'auteur utilise les concepts de modèle de métapopulation, épidémiologiques, “beamforming” et mobilité géographique obtenu à partir de données D4D. L'auteur propose la création de trois états latents qu'il ajoute au modèle épidémiologique connu: SIR. L'auteur étudie les états transitoires en analysant l'évolution du nombre de postes ayant reçu les informations et compare les résultats concernant ce nombre dans les différents cas. L'auteur démontre ainsi que le scenario qu'il propose permet d'améliorer le processus de diffusion des informations. Il montre aussi les effets de différents paramètres comme le nombre de sources, le nombre de paquets, les paramètres de mobilité et ceux qui caractérisent les antennes sur la diffusion des informations / In public warning message systems, information dissemination across the network is a critical aspect that has to be addressed. Dissemination of warning messages should be such that it reaches as many nodes in the network in a short time. In communication networks those based on device to device interactions, dissemination of the information has lately picked up lot of interest and the need for self organization of the network has been brought up. Self organization leads to local behaviors and interactions that have global effects and helps in addressing scaling issues. The use of self organized features allows autonomous behavior with low memory usage. Some examples of self organization phenomenon that are observed in nature are Lateral Inhibition and Flocking. In order to provide self organized features to communication networks, insights from such naturally occurring phenomenon is used. Achieving small world properties is an attractive way to enhance information dissemination across the network. In small world model rewiring of links in the network is performed by altering the length and the direction of the existing links. In an autonomous wireless environment such organization can be achieved using self organized phenomenon like Lateral inhibition and Flocking and beamforming (a concept in communication). Towards this, we first use Lateral Inhibition, analogy to Flocking behavior and beamforming to show how dissemination of information can be enhanced. Lateral Inhibition is used to create virtual regions in the network. Then using the analogy of Flocking rules, beam properties of the nodes in the regions are set. We then prove that small world properties are achieved using average path length metric. However, the proposed algorithm is applicable to static networks and Flocking and Lateral Inhibition concepts, if used in a mobile scenario, will be highly complex in terms of computation and memory. In a mobile scenario such as human mobility aided networks, the network structure changes frequently. In such conditions dissemination of information is highly impacted as new connections are made and old ones are broken. We thus use stability concept in mobile networks with beamforming to show how information dissemination process can be enhanced. In the algorithm, we first predict the stability of a node in the mobile network using locally available information and then uses it to identify beamforming nodes. In the algorithm, the low stability nodes are allowed to beamform towards the nodes with high stability. The difference between high and low stability nodes is based on threshold value. The algorithm is developed such that it does not require any global knowledge about the network and works using only local information. The results are validated using how quickly more number of nodes receive the information and different state of the art algorithms. We also show the effect of various parameters such as number of sources, number of packets, mobility parameters and antenna parameters etc. on the information dissemination process in the network. In realistic scenarios however, the dynamicity in the network is not only related to mobility. Dynamic conditions also arise due to change in density of nodes at a given time. To address effect of such scenario on the dissemination of information related to public safety in a metapopulation, we use the concepts of epidemic model, beamforming and the countrywide mobility pattern extracted from the $D4D$ dataset. Here, we also propose the addition of three latent states to the existing epidemic model ($SIR$ model). We study the transient states towards the evolution of the number of devices having the information and the difference in the number of devices having the information when compared with different cases to evaluate the results. Through the results we show that enhancements in the dissemination process can be achieved in the addressed scenario

Vu que la modulation multi-porteuses est largement utilisée dans les communications sans fil et la radio cognitive (CR pour “Cognitive Radio”) améliore l’utilisation des ressources radio et du spectre, nous nous concentrons sur les réseaux radio cognitifs (CR) pour faire progresser l’allocation des ressources, le routage, et l’ajustement de la puissance d’émission vers les récepteurs (synthèse de faisceaux ou beamforming) dans cette thèse. Nous étudions deux types de modulations multi-porteuses :Orthogonal Frequency-Division Multiplexing (OFDM) à base d’ondelettes (WOFDM pourWavelet OFDM) et OFDM dans sa forme classique ou traditionnelle (OFDM s’appuyant sur la transformation de Fourier pour partager les ressources). WOFDM adopte Wavelet Packet Modulation (WPM) pour obtenir des lobes secondaires beaucoup plus faibles dans la densité spectrale de puissance du signal transmis en comparaison à OFDM. WPM permet de surcroit à WOFDM de s’affranchir du Préfixe Cyclique (indispensable à OFDM) et d’exploiter l’égalisation pour combattre l’Interférence entre Symboles (ISI). Nous évaluons la performance de WOFDM sous différentes conditions du canal radio. Nous comparons la performance de WOFDM, qui s’appuie sur l’égalisation dans le domaine temporel, à celle de OFDM, qui requiert l’utilisation du Préfixe Cyclique et opère dans le domaine fréquentiel / In view of the wide usage of multicarrier modulation in wireless communications and the prominent contribution of Cognitive Radio (CR) to deal with critical shortage of spectrum resource, we focus on multicarrier based cognitive radio networks to investigate general resource allocation issues: subcarrier allocation, power allocation, routing, and beamforming in this thesis. We investigate two types of multicarrier modulation: Wavelet-based Orthogonal Frequency Division Multiplexing (WOFDM) and Fourier-based Orthogonal Frequency Division Multiplexing (OFDM). WOFDM adopts Wavelet Packet Modulation (WPM). Compared with fourier-based OFDM, wavelet-based OFDM achieves much lower side lobe in the transmitted signal. Wavelet-based OFDM excludes Cyclic Prefix (CP) which is used in fourier-based OFDM systems. Wavelet-based OFDM turns to exploit equalization to combat Inter-Symbol Interference (ISI). We evaluate the performance of WOFDM under different channel conditions. We compare the performance of wavelet-based OFDM using equalization in the time domain to that of fourier-based OFDM with CP and the equalization in the frequency domain

This research delves into wireless communication systems, with a particular focus on initial access processes, channel modeling, and beamforming strategies. The study involves meticulous channel data collection across diverse urban, suburban, and rural terrains, each presenting unique propagation challenges. The research also simulates a typical communication network with four base stations, adjusting their configurations to suit the varied terrains. A central focus is the implementation of the cell search methodology, including the exploration of random beamforming at both system and cell levels. The findings indicate that the cell-level system configurations do not yield significant performance improvements over the baseline configuration. Furthermore, potential increased costs associated with this strategy are noted. However, it is essential to highlight that this project serves as a critical exploration of the potential benefits of random beamforming at the cell level within non-public network scenarios. While the improvements observed are minimal, the insights gained from this research are poised to guide future research endeavors and contribute to the elimination of uncertainties in the field of wireless communication. / Denna forskning fördjupar sig i trådlösa kommunikationssystem, med särskilt fokus på initiala åtkomstprocesser, kanalmodellering och strålformningsstrategier. Studien involverade noggrann kanaldatainsamling över olika urbana, förorts- och landsbygdsterränger, var och en med unika spridningsutmaningar. Forskningen simulerade också ett typiskt kommunikationsnätverk med fyra basstationer som justerade deras konfigurationer för att passa de varierande terrängerna. Ett centralt fokus var implementeringen av cellsökningsmetoden, inklusive utforskning av slumpmässig strålformning på både system- och cellnivå. Resultaten indikerade att systemkonfigurationerna på cellnivå inte gav signifikanta prestandaförbättringar jämfört med baslinjekonfigurationen. Dessutom noterades potentiella ökade kostnader förknippade med denna strategi. Det är dock viktigt att betona att detta projekt fungerade som en kritisk utforskning av de potentiella fördelarna med slumpmässig strålformning på cellnivå inom icke-offentliga nätverksscenarier. Även om de observerade förbättringarna var minimala, är insikterna från denna forskning redo att vägleda framtida forskningsinsatser och bidra till att eliminera osäkerheter inom området trådlös kommunikation.

Automotive radars have gained considerable interest in recent years for applications of road safety for vulnerable road users. The use of multipleinput and multiple-output (MIMO) technology in automotive radar has helped in realising a virtual aperture greater than the physical aperture of the antenna which has reduced the size of the overall radar module. But increasing the number of MIMO channels for greater angular resolution can introduce increased computational complexity, processing time and latency. A new type of radar using the multiple input multiple steered output (MIMSO) radar can alleviate these concerns by replacing the angle-FFT with beamforming by means of a lens. A double-layer lens with a beamforming layer and a radiating layer with a radiating aperture on the 2-D footprint of the lens is proposed as an antenna system for this new radar technique. This work focuses on the radiating aperture which has been realised as a travelling wave planar slotted array in gap waveguide technology due to its benefit of low losses and ease of manufacturing. A ridged gap waveguide is chosen for the reduction of the waveguide size and to avoid the appearance of grating lobes in the visible range for large scan angles. The planar slotted array is synthesised in the travelling wave configuration and reflection cancelling notches are used in the ridge to cancel the reflections from the slots. The aperture is chosen to be of a circular shape for a compact design and to maximise aperture efficiency. The planar array is verified with a full-wave simulation with a bandwidth of 76 to 81 GHz and a realised gain of 27.7 dBi at the centre frequency. The array can be scanned up to ±50◦ with a scan loss of 2.4 dBi. / Fordonsradarer har fått stort intresse under de senaste åren för tillämpningar av trafiksäkerhet för utsatta trafikanter. Användningen av MIMO-teknik (multipleinput och multiple-output) i bilradar har hjälpt till att realisera en virtuell bländaröppning som är större än antennens fysiska bländaröppning, vilket har minskat storleken på den totala radarmodulen. Men att öka antalet MIMOkanaler för större vinkelupplösning kan introducera ökad beräkningskomplexitet, bearbetningstid och latens. En ny typ av radar som använder MIMSO-radarn (multiple input multiple steered output) kan lindra dessa problem genom att ersätta vinkel-FFT med strålformning med hjälp av en lins. En dubbelskiktslins med ett strålformande skikt och ett strålande skikt med en strålande bländare på linsens 2D-fotavtryck föreslås som ett antennsystem för denna nya radarteknik. Detta arbete fokuserar på strålningsöppningen som har realiserats som en plan slitsad array i gap-vågledarteknologi på grund av dess fördel med låga förluster och enkel tillverkning. En vågledare med räfflade gap väljs för att minska vågledarstorleken och för att undvika uppkomsten av gitterlober i det synliga området för stora avsökningsvinklar. Den plana uppsättningen syntetiseras i den vandringsvågkonfigurationen och reflektionsupphävande skåror används i åsen för att eliminera reflektionerna från slitsarna. Bländaren är vald för att ha en cirkulär form för en kompakt design och för att maximera bländareffektiviteten. Planar arrayen verifieras med en helvågssimulering med en bandbredd på 76 till 81 GHz och en realiserad vinst på 27, 7 dBi vid mittfrekvensen. Arrayen kan skannas upp till ±50◦ med en skanningsförlust på 2, 4 dBi.

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[ES] En la actualidad, cada vez existen más y más tareas que necesitamos exportar y automatizar en dispositivos portables de bajo consumo que se alimentan de baterías, en los cuales es imprescindible realizar un uso "optimo" de la energía disponible con la finalidad de no drenarlas rápidamente.En la sección primera de esta tesis, "Filtros de señales de audio digital", "optimizamos" las implementaciones de diferentes filtros, tanto generales como específicos, para aplicaciones de sonido digital diseñados e implantados en plataformas basadas en las arquitecturas ARM®. Como filtros generales, trabajamos con los filtros FIR, IIR y Parallel IIR, siendo este tipo de filtros implementados a bajo nivel con instrucciones vectoriales NEON®. Finalmente, se implementa un filtro de separación de señales conocido como "Beamforming", el cual plantea después de su estudio, la problemática de realizar una factorización QR de una matriz relativamente grande en tiempo real, lo cual nos lleva a desarrollar diferentes técnicas de "aceleración" de los cálculos de la misma. En la segunda parte, "Rellenado de mapa de profundidad de una escena", describimos el proceso de rellenado de un mapa de profundidad de una escena capturada a partir del uso de la imagen RGB y de un mapa de profundidad disperso donde únicamente tenemos valores de profundidad en los bordes de los objetos que componen la escena. Estos algoritmos de "rellenado" del mapa de profundidad, también han sido diseñados e implantados en dispositivos basados en la arquitectura ARM®. / [CA] Actualment, cada vegada existixen més i més tasques que tenen la necessitat d'exportar i automatitzar a dispositius portables de baix consum que s'alimenten amb bateríes, als quals es imprescindible realitzar un ús "óptim" de l'energia disponible amb la finalitat de no drenar-les ràpidament. Part I: Filtres de senyals d'àudio digital En aquesta secció "optimitzarem" les implementacions de diferents filtres, tant generals com específics, empreats a aplicacions de so digital disenyats e implantats a plataformes basades a les arquitectures ARM®. Com a filtres generals, treballem amb els filtres FIR, IIR y Parallel IIR, sent aquests tipus de filtres implementats a baix nivell amb instruccions vectorials NEON®. Finalment, s'implementa un filtro de separació de senyals conegut com "Beamforming", el qual planteja després del seu estudi, la problem`atica de realitzar una factorizació QR d'una matriu relativament gran en temps real, i açó ens porta a desenvolupar diferents tècniques "d'acceleració" dels càlculs de la mateixa. Part II: Emplenat del mapa de profunditat d'una escena A la secció d'image per computador, descrivim el procés d'emplenat d'un mapa de profunditat d'una escena capturada fent servir l'image RGB i un mapa de profunditat dispers on únicament tenim valors de profunditat als bordes dels objetes que composen l'escena. Aquests algoritmes "d'emplenat" del mapa de profunditat, també han sigut disenyats e implantats a dispositius basats en l'arquitectura ARM®. / [EN] Currently, there are more and more tasks that we need to export and automate in low-consumption mobile devices that are powered by batteries, in which it is essential to make an "optimum" use of the available energy in order to do not drain them quickly. Part I: Filters of digital audio signals In this section we "optimize" the implementations of different filters, both general and specific, for digital sound applications designed and implemented on platforms based on the ARM®. As general filters, we work with the FIR, IIR and Parallel IIR filters, these types of filters being implemented at a low level with NEON®vector instructions. Finally, a signal separation filter known as "Beamforming" is implemented, which set out after its study, the problem of performing a QR factorization of a relatively large matrix in real time, which leads us to develop different techniques of "acceleration" of the calculations of it. Part II: Filling the depth map of a scene In the computer image section, we describe the process of filling in a depth map of a captured scene using RGB image and a sparse depth map where we only have depth values at the edges of the objects that make up the scene. These depth map "filling" algorithms have also been designed and implemented in devices based on the ARM® architecture. / Alventosa Rueda, FJ. (2022). Algoritmos de imagen y sonido digital con restricciones de tiempo real [Tesis doctoral]. Universitat Politècnica de València. https://doi.org/10.4995/Thesis/10251/181573

Abstract : This thesis considers the discrimination of inlet / exhaust noise of aero-engines in free-field static tests using far-field microphone arrays. Various techniques are compared for this problem, including classical beamforming (CB), regularized inverse method (Tikhonov regularization), LI - generalized inverse beamforming (LI-GIB), clean-PSF, clean-SC and two novel methods which are called hybrid method and clean-hybrid. The classical beamforming method is disadvantaged due to its need for a high number of measurement microphones in accordance with the requirements. Similarly, the inverse method is disadvantaged due to their need of having a priori source information. The classical Tikhonov regularization provides improvements in solution stability, however continues to be disadvantaged due to its requirement of imposing a stronger penalty for undetected source positions. Coherent and incoherent sources are resolved by LI-generalized inverse beamforming (L1-GIB). This algorithm can distinguish the multipole sources as well as the monopoles sources. However, source identification by LI-generalized inverse beamforming takes much time and requires a PC with high memory. The hybrid method is a new regularization method which involves the use of an a priori beamforming measurement to define a data-dependent discrete smoothing norm for the regularization of the inverse problem. Compared to the classical beamforming and the inverse modeling, the hybrid (beamforming regularization) approach provides improved source strength maps without substantial added complexity. Although the hybrid method rather solves the disadvantage of the former methods, the application of this method for identification of weaker sources in the presence of the strong sources isn't satisfactory. This can be explained by the large penalization being applied to the weaker source in the hybrid method, which results in underestimation of source strength for this source. To overcome this defect, the clean-SC method and the proposed clean-hybrid method, which is a combination of the hybrid method and the clean-SC, are applied. These methods remove the effect of the strong sources in source power maps to identify the weaker sources. The proposed methods which represent the main contribution of this thesis show promising results and opens new research avenues. Theoretical study of all approaches is performed for various sources and configurations of array. In order to validate the theoretical study, several laboratory experiments are conducted at Universito de Sherbrooke. The proposed methods have further been applied to the measured noise data from a Pratt & Whitney Canada turbo-fan engine and have been observed to provide better spatial resolution and solution robustness with a limited number of measurement microphones compared to the existing methods. / Résumé : La présente thèse étudie la discrimination du bruit d'entrée / de sortie des moteurs d'avion dans des tests statiques en champ libre en utilisant des antennes de microphones en champ lointain. Diverses techniques sont comparées pour ce problème, dont la formation de voie classique (CB), la méthode inverse régularisée (régularisation de Tikhonov), la formation de voies généralisée inverse (L1-GIB), Clean-PSF, Clean-SC et deux méthodes proposées qui s'appellent la méthode hybride et la méthode Clean-hybride. La méthode la formation de voie classique est désavantagée en raison de son besoin de nombreux microphones de mesure. De même, la méthode inverse est désavantagée en raison du besoin d'information a priori sur les sources. La régularisation Tikhonov classique fournit des améliorations dans. la stabilité de la solution; cependant elle reste désavantageuse en raison de son exigence d'imposer une pénalité plus forte pour des positions de source non détectées. Des sources cohérentes et incohérentes peuvent être résolues par la formation de voies généralisée inverse (L1-GIB). Cet algorithme peut identifier les sources multi- polaires aussi bien que les sources monopolaires. Cependant, l'identification de source par la formation de voies généralisée inverse prend beaucoup de temps et exige un ordinateur avec une capacité de mémoire élevée. La méthode hybride est une nouvelle méthode de régularisation qui implique l'utilisation d'un traitement par formation de voie a priori pour définir une norme discrète et dépendante des données pour la régularisation du problème inverse. En comparaison avec la formation de voie classique et la méthode inverse, l'approche hybride (régularisation par formation de voie) fournit des cartographies améliorées d'amplitudes de sources sans aucune complexité supplémentaire substantielle. Bien que la méthode hybride lève les limitations des méthodes classiques, l'application de cette méthode pour l'identification de sources de faible puissance en présence de sources de forte puissance n'est pas satisfaisante. On peut expliquer ceci par la plus grande pénalisation appliquée à la source plus faible dans la méthode hybride, qui aboutit à la sous-estimation de l'amplitude de cette source. Pour surmonter ce défaut, la méthode Clean-SC et la méthode Clean-hybrides proposée qui est une combinaison de la méthode hybride et de Clean-SC sont appliquées. Ces méthodes éliminent l'effet des sources fortes dans les cartographies de puissance de sources pour identifier les sources plus faibles. Les méthodes proposées qui représentent la contribution principale de cette thèse conduisent à des résultats fiables et ouvrent des nouvelles voies de recherche. L'étude théorique de toutes les approches est menée pour divers types de sources et de configurations microphoniques. Pour valider l'étude théorique, plusieurs expériences en laboratoire sont réalisées à Université de Sherbrooke. Les méthodes proposées ont été appliquées aux données de bruit mesurées d'une turbo-soufflante Pratt & Whitney Canada pour fournir une meilleure résolution spatiale des sources acoustique et une solution robuste avec un nombre limité des microphones de mesure comparé aux méthodes existantes.

In teleconferencing the audio quality is key to conducting successful meetings. The conference room setting imposes various challenges on the speech signal processing, such as noise and interfering signals, reverberation, or participants positioned far from the telephone unit. This work aims at improving the received speech signal of a conference telephone by implementing sound source localization and beamforming. The implemented microphone array signal processing techniques are compared to the performance of an existing multi-microphone solution and evaluated under various conditions using a planar uniform circular array. Recordings of test-sequences for the evaluation were performed using a custom-built array mockup. The implemented algorithms did not show good enough performance to motivate the increased computational complexity compared to the existing solution. Moreover, an increase in number of microphones used was concluded to have little or no effect on the performance of the methods. The type of microphone used was, however, concluded to have impact on the performance and a subjective listening evaluation indicated a preference for omnidirectional microphones which is recommended to investigate further. / God ljudkvalitet är en grundsten för lyckade telefonmöten. Miljön i ett konferens-rum medför ett flertal olika utmaningar för behandlingen av mikrofonsignalerna: det kan t.ex. vara brus och störningar, eller att den som talar befinner sig långt från telefonen. Målet med detta arbete är att förbättra den talsignal som tas upp av en konferenstelefon genom att implementera lösningar för lokalisering av talaren och riktad ljudupptagning med hjälp av ett flertal mikrofoner. De implementerade metoderna jämförs med en befintlig lösning och utvärderas under olika brusscenarion för en likformig cirkulär mikrofonkonstellation. För utvärderingen användes testsignaler som spelades in med en specialbyggd enhet. De implementerade algoritmerna kunde inte uppvisa en tillräcklig förbättring i jämförelse med den befintliga lösningen för att motivera den ökade beräkningskomplexitet de skulle medföra. Dessutom konstaterades att en fördubbling av antalet mikrofoner gav liten eller ingen förbättring på metoderna. Vilken typ av mikrofon som användes konstaterades däremot påverka resultatet och en subjektiv utvärdering indikerade en preferens för de rundupptagande mikrofonerna, en skillnad som föreslås undersökas vidare.

Nas últimas décadas, a poluição sonora tornou-se um grande problema para a sociedade. É por esta razão que a indústria tem aumentado seus esforços para reduzir a emissão de ruído. Para fazer isso, é importante localizar quais partes das fontes sonoras são as que emitem maior energia acústica. Conhecer os pontos de emissão é necessário para ter o controle das mesmas e assim poder reduzir o impacto acústico-ambiental. Técnicas como \"beamforming\" e \"Near-Field Acoustic Holography\" (NAH) permitem a obtenção de imagens acústicas. Essas imagens são obtidas usando um arranjo de microfones localizado a uma distância relativa de uma fonte emissora de ruído. Uma vez adquiridos os dados experimentais pode-se obter a localização e magnitude dos principais pontos de emissão de ruído. Do mesmo modo, ajudam a localizar fontes aeroacústicas e vibro acústicas porque são ferramentas de propósito geral. Usualmente, estes tipos de fontes trabalham em diferentes faixas de frequência de emissão. Recentemente, foi desenvolvida a transformada de Kronecker para arranjos de microfones, a qual fornece uma redução significativa do custo computacional quando aplicada a diversos métodos de reconstrução de imagens, desde que os microfones estejam distribuídos em um arranjo separável. Este trabalho de mestrado propõe realizar medições com sinais reais, usando diversos algoritmos desenvolvidos anteriormente em uma tese de doutorado, quanto à qualidade do resultado obtido e à complexidade computacional, e o desenvolvimento de alternativas para tratamento de dados quando alguns microfones do arranjo apresentarem defeito. Para reduzir o impacto de falhas em microfones e manter a condição de que o arranjo seja separável, foi desenvolvida uma alternativa para utilizar os algoritmos rápidos, eliminando-se apenas os microfones com defeito, de maneira que os resultados finais serão obtidos levando-se em conta todos os microfones do arranjo. / In recent decades, noise pollution has become a major problem for society. It is for this reason that the industry has increased its efforts to reduce the emission of noise. To do this, it is important to find out which parts of the sound sources are emitting greater acoustic energy. Knowing the emission points is required to keep track of them and thus be able to reduce acoustic and environmental impact. Techniques such as \"beamforming\" and \"Near-Field Acoustic Holography\" (NAH) allow obtaining acoustic images. These images are obtained using an array of microphones located at a relative distance from a noise source. Once acquired experimental data, one can obtain the location and the magnitude of the main points of noise emission. Similarly, we can find aeroacoustic and vibroacoustic sources, because they are general-purpose tools. Usually, these types of sources work in different frequency bands of the emission. Recently, a Kronecker Array Transform (KAT) was developed to microphone arrays, which provides a significant reduction in computational cost when applied to various image reconstruction methods, provided that the microphones are distributed in a separable array. This thesis proposes perform measurements with real signals using different algorithms previously developed in a dissertation about the quality of the obtained results and computational complexity, and the development of alternatives for data processing when some microphones of the array are defective. To reduce the impact of failures in microphones and maintain the condition that the array is separable, one alternative has been developed to use the fast algorithms by removing only the defective microphones, so that the final results will be obtained taking into account every microphone of the array.

Nas últimas décadas, a poluição sonora tornou-se um grande problema para a sociedade. É por esta razão que a indústria tem aumentado seus esforços para reduzir a emissão de ruído. Para fazer isso, é importante localizar quais partes das fontes sonoras são as que emitem maior energia acústica. Conhecer os pontos de emissão é necessário para ter o controle das mesmas e assim poder reduzir o impacto acústico-ambiental. Técnicas como \"beamforming\" e \"Near-Field Acoustic Holography\" (NAH) permitem a obtenção de imagens acústicas. Essas imagens são obtidas usando um arranjo de microfones localizado a uma distância relativa de uma fonte emissora de ruído. Uma vez adquiridos os dados experimentais pode-se obter a localização e magnitude dos principais pontos de emissão de ruído. Do mesmo modo, ajudam a localizar fontes aeroacústicas e vibro acústicas porque são ferramentas de propósito geral. Usualmente, estes tipos de fontes trabalham em diferentes faixas de frequência de emissão. Recentemente, foi desenvolvida a transformada de Kronecker para arranjos de microfones, a qual fornece uma redução significativa do custo computacional quando aplicada a diversos métodos de reconstrução de imagens, desde que os microfones estejam distribuídos em um arranjo separável. Este trabalho de mestrado propõe realizar medições com sinais reais, usando diversos algoritmos desenvolvidos anteriormente em uma tese de doutorado, quanto à qualidade do resultado obtido e à complexidade computacional, e o desenvolvimento de alternativas para tratamento de dados quando alguns microfones do arranjo apresentarem defeito. Para reduzir o impacto de falhas em microfones e manter a condição de que o arranjo seja separável, foi desenvolvida uma alternativa para utilizar os algoritmos rápidos, eliminando-se apenas os microfones com defeito, de maneira que os resultados finais serão obtidos levando-se em conta todos os microfones do arranjo. / In recent decades, noise pollution has become a major problem for society. It is for this reason that the industry has increased its efforts to reduce the emission of noise. To do this, it is important to find out which parts of the sound sources are emitting greater acoustic energy. Knowing the emission points is required to keep track of them and thus be able to reduce acoustic and environmental impact. Techniques such as \"beamforming\" and \"Near-Field Acoustic Holography\" (NAH) allow obtaining acoustic images. These images are obtained using an array of microphones located at a relative distance from a noise source. Once acquired experimental data, one can obtain the location and the magnitude of the main points of noise emission. Similarly, we can find aeroacoustic and vibroacoustic sources, because they are general-purpose tools. Usually, these types of sources work in different frequency bands of the emission. Recently, a Kronecker Array Transform (KAT) was developed to microphone arrays, which provides a significant reduction in computational cost when applied to various image reconstruction methods, provided that the microphones are distributed in a separable array. This thesis proposes perform measurements with real signals using different algorithms previously developed in a dissertation about the quality of the obtained results and computational complexity, and the development of alternatives for data processing when some microphones of the array are defective. To reduce the impact of failures in microphones and maintain the condition that the array is separable, one alternative has been developed to use the fast algorithms by removing only the defective microphones, so that the final results will be obtained taking into account every microphone of the array.