Le Haut-Parleur Digital Matriciel (HPDM) est un moyen de transduction électroacoustique qui reçoit comme entrée un signal numérique et qui effectue la conversion vers l'analogique directement dans l'air. Il est constitué de plusieurs éléments rayonnants disposés au sein d'une matrice. Ces éléments seront désignés par le terme «speaklet» lorsqu'ils sont de tailles réduites. Le rayonnement acoustique du HPDM est en effet très sensible à la taille de la matrice, ce qui le rend tout particulièrement adapté à la technologie MEMS. Cette thèse porte sur l'étude de HPDM MEMS piézoélectriques. Après une introduction qui débute par certaines généralités jusqu'à se focaliser sur le sujet, la thèse aborde les modélisations multiphysiques des HPDM étudiés, le dimensionnement des speaklets puis les tests expérimentaux. Des modèles analytiques ainsi que des simulations numériques et par éléments finis sont mis en place et permettent de prédire le comportement mécanique des speaklets présentés, les pressions rayonnées par les HPDM et les puissances électriques consommées. Les speaklets sont ensuite dimensionnés à partir de l'empilement technologique afin de maximiser le niveau de pression qu'ils génèrent. Des tests expérimentaux permettent alors de valider la majorité des modèles ou au contraire de revenir sur certains d'entre eux pour les optimiser ou montrer leurs limitations. Les résultats ont en effet montré l'importance de la prise en compte des contraintes résiduelles, qui provoquent une déformée initiale des speaklets et modifient leurs fréquences propres, rendant alors l'utilisation de grands rayons inefficaces. En accord avec les modèles, les speaklets possèdent un comportement dynamique linéaire, ce qui permet de les caractériser à l'aide de fonctions de transfert. La théorie et les enregistrements sonores montrent alors qu'un HPDM composé de tels speaklets permet dans le meilleur des cas de produire une pression identique à celle générée par la même matrice pilotée en analogique. Dans notre cas, des taux de distorsions supérieurs ont été obtenus lors des reconstructions digitales, à cause des réponses non uniformes des speaklets, dues à des résistances d'accès différentes. Le HPDM présenté possède cependant d'autres avantages, le plus important étant la très faible consommation électrique qu'il est théoriquement possible d'atteindre en utilisant les méthodes de charges et de décharges adiabatiques. Le HPDM piézoélectrique MEMS apparait donc comme étant une technologie prometteuse. L'optimisation de notre premier prototype à l'aide des outils développés doit en effet conduire à un HPDM capable de générer une pression équivalente à celle obtenue en mode analogique, mais avec un rendement électroacoustique beaucoup plus important. Les futurs travaux devront ensuite se concentrer sur la conception de speaklets non-linéaires et sur la forme du pulse de pression qu'ils génèrent, afin de gagner en niveau sonore. / The Digital Loudspeaker Array (DLA) is an electroacoustic transducer which receives as input a digital signal and performs the analog conversion directly into the air. It consists of a plurality of radiating elements arranged in a matrix. These elements will be designated by the term “speaklet” when they are reduced in size. The acoustic radiation of a DLA is indeed very sensitive to the size of the matrix due to differences in path length, which makes it especially suitable for MEMS technology. This thesis is on the study of a piezoelectric MEMS DLA. After an introduction that is increasingly focused on the subject, the thesis addresses the multiphysics modeling of the DLA, dimensioning of the speaklets and experimental tests. Analytical formulas, numerical simulations and finite element models are developed and used to predict the mechanical behavior of the presented speaklets, the pressure radiated by the DLA and the electrical power consumption. The speaklet are then dimensioned from the technological stack (set in advance) in order to maximize the pressure level. Experimental tests involving the use of an anechoic chamber, an optical interferometer, a vibrometer and an impedancemeter validate most of the models. Otherwise, these tests are usefull for improving some of them or for showing their limitations. The results have shown the importance of the residual stresses, which cause an initial deformation of the speaklets and modify their resonance frequencies, thus rendering ineffective the use of large radii. In accordance with the models, the static deflection of the speaklets is nonlinear but their dynamic behavior is linear. This enables characterizations using transfer functions. Theory and sound recordings show that a DLA made of such speaklets can produce in the best case the same pressure to that generated by the same matrix driven in an analog way. In our case, more distortions were obtained in digital reconstructions because of non-uniform responses of the speaklets, due to different access resistances. However, the presented DLA has other advantages, the most important being the very low power consumption it is theoretically possible to achieve using the adiabatic charge principle. The piezoelectric MEMS DLA thus appears as a promising technology. The optimization of our first prototype using the developed tools should indeed lead to a DLA able to generate an equivalent presure to that obtained with analog control, but with a far greater electroacoustic efficiency. Future work should then focus on the design of nonlinear speaklets and on the shaping of the pulse of pressure they generate, in order to increase the total pressure level.

Dielectric elastomer actuators (DEAs) have been used for artificial muscles for years. Recently the DEA-based deformable surfaces have demonstrated controllable microscale roughness, ease of operation, fast response, and possibilities for programmable control. DEA muscles used in bioinspired robotic arms for large deformation and strong force also become desirable for their efficiency, low manufacturing cost, high force-to-weight ratio, and noiseless operation. The DEA-based responsive surfaces in microscale roughness control, however, exhibit limited durability due to irreversible dielectric breakdown. Lowering device voltage to avoid this issue is hindered by an inadequate understanding of the electrically-induced wrinkling deformation as a function of the deformable dielectric film thickness. Also, the programmable control and geometric analysis of the structured surface deformation have not yet been fully explored. Current methods to generate anisotropic wrinkles rely on mechanical pre-loading such as stretching or bending, which complicates the fabrication and operation of the devices. With a fixed mechanical pre-loading, the device can only switch between the flat state and the preset wrinkling state. In this thesis, we overcome these shortcomings by demonstrating a simple method for fabricating fault-tolerant electro-responsive surfaces and for controlling surface wrinkling patterns. The DEA-based system can produce different reversible surface topographies (craters, irregular wrinkles, structured wrinkles) upon the geometrical design of electrode and application of voltage. It remains functional due to its ability to self-insulate breakdown faults even after multiple high voltage breakdowns, and the induced breakdown punctures can be used for amplification of local electric fields for wrinkle formation at lower applied voltages. We enhance fundamental understanding of the system by using different analytical models combined with numerical simulation to discuss the mechanism and critical conditions for wrinkle formation, and compare it with the experimental results from surface topography, critical field to induce wrinkles in films of different thickness, and wrinkling patterns quantitatively analysed by different disorder metrics. Based on the results, we demonstrate its wide applicability in adjustable transparency films, dynamic light-grating filter, molding for static surface patterns, and multi-stable mirror-diffusor-diffraction grating device. For DEAs used for macroscopic-scale deformation in robotic arms, the main issue that undermines the performance of DEA muscles is the trade-off between strong force and large displacement, which limits the durability and range of potential robotic and automation applications of DEA-driven devices. In this thesis, this challenge is tackled by using DEAs in loudspeaker configuration for independent scaling-up of force and displacement, developing a theoretical prediction to optimise the operation of such DEAs in bioinspired antagonistic system to maximise speed and power of the robotic arm, and designing a clutch-gear-shaft mechanical system collaborating with the muscles to decouple the displacement and output force. Therefore, the trade-off between force and displacement in traditional DEA muscles can be resolved. The mechanical system can also convert the short linear spurt to an unlimited rotary motion. Combining these advantages, continuous movement with high output force can be accomplished.

We investigated two design strategies that would allow us to efficiently process audio signals on embedded systems such as mobile phones and portable electronics. In the first strategy, we exploit properties of the human auditory system to process audio signals. We designed a sound enhancement algorithm to make piezoelectric loudspeakers sound "richer" and "fuller," using a combination of bass extension and dynamic range compression. We also developed an audio energy reduction algorithm for loudspeaker power management by suppressing signal energy below the masking threshold. In the second strategy, we use low-power analog circuits to process the signal before digitizing it. We designed an analog front-end for sound detection and implemented it on a field programmable analog array (FPAA). The sound classifier front-end can be used in a wide range of applications because programmable floating-gate transistors are employed to store classifier weights. Moreover, we incorporated a feature selection algorithm to simplify the analog front-end. A machine learning algorithm AdaBoost is used to select the most relevant features for a particular sound detection application. We also designed the circuits to implement the AdaBoost-based analog classifier.

Submitted by Luciana Ferreira (lucgeral@gmail.com) on 2017-06-14T11:09:20Z No. of bitstreams: 2 license_rdf: 0 bytes, checksum: d41d8cd98f00b204e9800998ecf8427e (MD5) Dissertação - Luiz Henrique Reis de Jesus - 2017.pdf: 3311759 bytes, checksum: 20e4044376c2666f102131892f7b3fc2 (MD5) / Approved for entry into archive by Luciana Ferreira (lucgeral@gmail.com) on 2017-06-14T11:09:35Z (GMT) No. of bitstreams: 2 license_rdf: 0 bytes, checksum: d41d8cd98f00b204e9800998ecf8427e (MD5) Dissertação - Luiz Henrique Reis de Jesus - 2017.pdf: 3311759 bytes, checksum: 20e4044376c2666f102131892f7b3fc2 (MD5) / Made available in DSpace on 2017-06-14T11:09:35Z (GMT). No. of bitstreams: 2 license_rdf: 0 bytes, checksum: d41d8cd98f00b204e9800998ecf8427e (MD5) Dissertação - Luiz Henrique Reis de Jesus - 2017.pdf: 3311759 bytes, checksum: 20e4044376c2666f102131892f7b3fc2 (MD5) Previous issue date: 2017-05-15 / Conselho Nacional de Pesquisa e Desenvolvimento Científico e Tecnológico - CNPq / This dissertation of master degree describes an approach of the interactive optimization process associate to the Evolution Strategies method for the evaluation of the loudspeaker optimization project, with the objective to present the advantages achieved after the user interventions throughout the optimization process. Its development is based on the methodology of the Evolution Strategies characterized by the concept of natural selection, which uses combination and mutation methods to generate new individuals. However, for greater efficiency in the responses of the objective function, as well as the reduction in its convergence time, the optimization process requires interventions in stabilization of local minima and maxima. In the interventions made available to the user, will be considered manipulations in the operators of mutation and combination, mutation of the parameters of self-adaptation, as well as the change of objective and the variation of their respective restriction. As a differential, an interface was developed to make feasible the user interventions applied to the optimization process, as well as the monitoring of the entire process. In this work, also evaluated optimization test functions with the objective of validating the proposed methodology. / Esta dissertação de mestrado descreve uma abordagem do processo de otimização interativa associado ao método das Estratégias de Evolução para a avaliação do projeto de otimização do alto-falante, com o objetivo de apresentar as vantagens alcançadas após as intervenções do usuário ao longo do processo de otimização. Seu desenvolvimento é baseado na metodologia das Estratégias de Evolução caracterizada pelo conceito de seleção natural, o qual utiliza de métodos de combinação e mutação para a geração de novos indivíduos. No entanto, para maior eficiência nas respostas a função objetivo, bem como a redução em seu tempo de convergência, o processo de otimização necessita de intervenções em estabilizações de mínimos e máximos locais. Nas intervenções disponibilizadas ao usuário, serão consideradas manipulações nos operadores de mutação e combinação, mutação dos parâmetros de auto-adaptação, bem como a mudança de objetivo e a variação de sua respectiva restrição. Como diferencial, foi desenvolvida uma interface para viabilizar as intervenções do usuário aplicadas ao processo de otimização, bem como o acompanhamento de todo o processo. Neste trabalho, também foram avaliadas funções de teste de otimização com o objetivo de validar a metodologia proposta.

This thesis deals with the measurement and description of properties of the two-way variable modular loudspeaker system marked as OM-3, especially the influence of the mutual position of drivers on its characteristics. The thesis examines individual factors taking their contribution on characteristics of the loudspeaker system in total. Further it measures and evaluates options of compensations of mutual position changes of drivers using an all-pass filter. Within this thesis is also included laboratory task preparation which is focused on this problematics.

The present final thesis discusses the transmission of a narrowly directional parametric sound beam through an amplitude-modulated ultrasonic wave, utilizing the effect of auto-demodulation in a nonlinear medium and ensuring the subsequent processing of the input signal for the parametric sound transmitter. Emphasis is placed on the mathematical tools that relate to parametric sound transmission in a nonlinear medium. The basic part of the thesis describes a parametric speaker and the associated amplitude modulation techniques, which constitute a major prerequisite for the processing of the transmitted signal. In the following section, the author then analyzes the computational intensity of these techniques, considering applicable hardware approaches. Finally, the fabrication and practical use of the proposed solution are discussed, including the measurement of typical parameters such as the spatial radiation characteristics, total harmonic distortion, and transmission channel bandwidth.

The master´s thesis deals with theoretical research and practical application of the principle of spreading acoustic waves through nonlinear medium. The theoretical part describes five different types of amplitude modulation and their mathematic representation, which were verified by the program Matlab. Next, the impedance characteristic of ultrasound PZT transducer was measured and the design of PWM modulator and class D amplifier was described. The practical part of the master´s thesis contains the realization of the parametric loudspeaker assembled from the class D amplifier and the PWM modulator. In conclusion, the designed parametric loudspeaker was measured and the individual types of amplitude modulation were compared with respect to the total harmonic distortion of transmitted useful signal.

Monitoring is hugely important for musicians in any type of live situation for them to provide a good musical performance, and In-ear Monitoring (IEM) is increasingly popular among both musicians and audio engineers. Despite this, there has been little research on the topic and musicians and audio engineers would benefit from knowing more about the system, to utilize it in the best way possible. Because of the potentially added issues with structure borne sound, the study has chosen to focus on vocalists that use IEM. Through interviews with both Less experienced IEM users and Experienced IEM users, this study aims to find out what issues IEM users experience in live situations. Other than perceived issues, the study investigates how issues evolve with increased experience, comparing the two user groups, what solutions can be found and how an IEM mix should be for a vocalist. Through a Thematic analysis, the study reaches the conclusion that the main issue with the IEM system for vocalists is isolation from the surrounding world, and that the isolation leads to other perceived issues such as worsening the live feeling and obstructing contact with the audience. The study also concludes that there is a clear connection between monitoring and musical performance with poor monitoring leading to worse performance. The main solution observed is habit with the system, good communication between the audio engineers and musicians and an understanding for both sides workflow to increase efficiency in mixing. With this knowledge, further research can be made to increase the understanding for musicians and audio engineers respectively, and audio engineers and musicians can already focus on increasing cooperation.

La caractérisation des sources sonores dues aux vibrations est un défi dans le domaine du bruit et des vibrations. Dans cette thèse, une approche expérimentale pour caractériser la propagation du son d’une machine complexe a été étudiée. Pour caractériser de manière appropriée la source sonore placée dans un environnement quelconque, il a été indispensable de prendre en compte les phénomènes de rayonnement et de diffraction. Cela permet de prédire une pression acoustique. Une technique particulière, appelée source mannequin, a été développée pour répondre à cette problématique. Le mannequin est une enceinte fermée de taille similaire mais qui a une forme simplifiée par rapport à la machine complexe, et sert de modèle de diffraction sonore. Le mannequin est équipé d’une série de haut-parleurs alignés dans le prolongement de la surface de l’enceinte. La superposition du champ acoustique créé par chaque haut-parleur modélise le rayonnement acoustique de la machine complexe. Cette thèse introduit donc le concept de source mannequin et traite de trois problèmes émanant de la mise en pratique de celui-ci : (1) l’estimation du transfert d’impédance dans l’espace (fonction de Green), (2) les spécifications de l’enceinte et de la série de haut-parleurs, et (3) l’estimation des sources équivalentes en termes de débit volumique. L’approche est étudiée au travers de cas d’études expérimentaux et numériques. / The characterisation of vibrating sound sources is a challenge in noise and vibration engineering. In this thesis, an experimental approach to the characterisation of air-borne sound from a complex machine is investigated. A proper characterisation has to account for both radiation and diffraction phenomena in order to describe the sound source when inserted into an arbitrary space which enables prediction of sound pressure. A particular technique — a dummy source — has been conceived to deal with this problem. The dummy is a closed cabinet of similar size but much simpler shape than the complex machine, and it serves as a model of sound diffraction. The dummy is equipped with a flush-mounted array of loudspeaker drivers. The superposition of sound fields created by the individual drivers models sound radiation of the complex machine. This thesis introduces the concept of a dummy source and discusses three problems that need to be addressed for its practical application: (1) estimation of the transfer impedance of the space (the Green’s function), (2) the specification of the cabinet and the driver array, and (3) the estimation of the equivalent source strengths in terms of volume velocity. The approach is investigated via experimental and numerical case studies.