Measuring acoustic pressure gradients is critical in many applications such as directional microphones for hearing aids and sound intensity probes. This measurement is especially challenging with decreasing microphone size, which reduces the sensitivity due to small spacing between the pressure ports. Novel, micromachined biomimetic microphone diaphragms are shown to provide high sensitivity to pressure gradients on one side of the diaphragm with low thermal mechanical noise. These structures have a dominant mode shape with see-saw like motion in the audio band, responding to pressure gradients as well as spurious higher order modes sensitive to pressure. In this dissertation, integration of a diffraction based optical detection method with these novel diaphragm structures to implement a low noise optical pressure gradient microphone is described and experimental characterization results are presented, showing 36 dBA noise level with 1mm port spacing, nearly an order of magnitude better than the current gradient microphones. The optical detection scheme also provides electrostatic actuation capability from both sides of the diaphragm separately which can be used for active force feedback. A 4-port electromechanical equivalent circuit model of this microphone with optical readout is developed to predict the overall response of the device to different acoustic and electrostatic excitations. The model includes the damping due to complex motion of air around the microphone diaphragm, and it calculates the detected optical signal on each side of the diaphragm as a combination of two separate dominant vibration modes. This equivalent circuit model is verified by experiments and used to predict the microphone response with different force feedback schemes. Single sided force feedback is used for active damping to improve the linearity and the frequency response of the microphone. Furthermore, it is shown that using two sided force feedback one can significantly suppress or enhance the desired vibration modes of the diaphragm. This approach provides an electronic means to tailor the directional response of the microphones, with significant implications in device performance for various applications. As an example, the use of this device as a particle velocity sensor for sound intensity and sound power measurements is investigated. Without force feedback, the gradient microphone provides accurate particle velocity measurement for frequencies below 2 kHz, after which the pressure response of the second order mode becomes significant. With two-sided force feedback, the calculations show that this upper frequency limit may be increased to 10 kHz. This improves the pressure residual intensity index by more than 15 dB in the 50 Hz-10 kHz range, matching the Class I requirements of IEC 1043 standards for intensity probes without any need for multiple spacers.

MEMS microphones have been a research topic for the last two and half decades. The state-of-the-art comprises surface mount MEMS microphones in laptops, mobile phones and tablets, etc. The popularity and the commercial success of MEMS microphones is largely due to the steep cost reduction in manufacturing afforded by the mass scale production with microfabrication technology. The current MEMS microphones are de-signed along the lines of traditional microphones that use capacitive transduction with or without permanent charge (electret type microphones use permanent charge of their sensor element). These microphones offer high sensitivity, stability and reasonably at frequency response while reducing the overall size and energy consumption by exploiting MEMS technology. Conceptually, microphones are simple transducers that use a membrane or diaphragm as a mechanical structure which deflects elastically in response to the incident acoustic pressure. This dynamic deflection is converted into an electrical signal using an appropriate transduction technique. The most popular transduction technique used for this application is capacitive, where an elastic diaphragm forms one of the two parallel plates of a capacitor, the fixed substrate or the base plate being the other one. Thus, there are basically two main elements in a microphone { the elastic membrane as a mechanical element, and the transduction technique as the electrical element. In this thesis, we propose and study novel design for both these elements. In the mechanical element, we propose a simple topological change by introducing slits in the membrane along its periphery to enhance the mechanical sensitivity. This simple change, however, has significant impact on the microphone design, performance and its eventual cost. Introduction of slits in the membrane makes the geometry of the structural element non-trivial for response analysis. We devote considerable effort in devising appropriate modeling techniques for deriving lumped parameters that are then used for simulating the system response. For transduction, we propose and study an FET (Field Effect Transistor) coupled micro-phone design where the elastic diaphragm is used as the moving (suspended) gate of an FET and the gate deflection modulated drain current is used in the subthreshold regime of operation as the output signal of the microphone. This design is explored in detail with respect to various design parameters in order to enhance the electrical sensitivity. Both proposed changes in the microphone design are motivated by the possibilities that the microfabrication technology offers. In fact, the design proposed here requires further developments in MEMS technology for reliably creating gaps of 50-100 nm between the substrate and a large 2D structure of the order of a few hundred microns in diameter. In the First part of the thesis, we present detailed simulations of acoustic and squeeze lm domain to understand the effect slits could bring upon the behaviour of the device as a microphone. Since the geometry is nontrivial, we resort to Finite element simulations using commercial packages such as COMSOL Multiphysics and ANSYS in the structural, acoustic and Fluid-structure domains to analyze the behaviour of a microphone which has top plate with nontrivial geometry. On the simulated Finite element data, we conduct low and high frequency limit analysis to extract expressions for the lumped parameters. This technique is well known in acoustics. We borrow this technique of curve Fitting from the acoustics domain and apply it in modified form into the squeeze lm domain. The dynamic behaviour of the entire device is then simulated using the extracted parameters. This helps to simulate the microphone behaviour either as a receiver or as a transmitter. The designed device is fabricated using MEMSCAP PolyMUMPS process (a foundry Polysilicon surface micromachining process). We conduct vibrometer (electrostatic ex-citation) and acoustic characterization. We also study the feasibility of a microphone with slits and the issues involved. The effect of the two dissipation modes (acoustic and squeeze lm ) are quantified with the experimentally determined quality factor. The experimentally measured values are: Resonance is 488 kHz (experimentally determined), low frequency roll-off is 796 Hz (theoretical value) and is 780 Hz as obtained by electrical characterization. The first part of this thesis focusses on developing a comprehensive understanding of the effect of slits on the performance of a MEMS microphone. The presence of slits near the circumference of the clamped plate cause reduction in its rigidity. This leads to an increase in the sensitivity of the device. Slits also cause pressure equalization between the top and bottom of the diaphragm if the incoming sound is at relatively low frequencies. At this frequency, also known as the lower cutoff frequency, the microphone's response starts dropping. The presence of slits also changes the radiation impedance of the plate as well as the squeeze lm damping below the plate. The useful bandwidth of the microphone changes as a consequence. The cavity formed between the top plate and the bottom fixed substrate increases the stiffness of the device significantly due to compression of the trapped air. This effect is more pronounced here because unlike the existing capacitive MEMS microphones, there is no backchamber in the device fabricated here. In the second part of the thesis, we present a novel subthreshold biased FET based MEMS microphone. This biasing of the transistor in the subthreshold region (also called as the OFF-region) offers higher sensitivity as compared to the above threshold region (also called as the ON-region) biasing. This is due to the exponentially varying current with change in the bias voltage in the OFF-region as compared to the quadratic variation in the ON-region. Detailed simulations are done to predict the behaviour of the device. A lumped parameter model of the mechanical domain is coupled with the drain current equations to predict the device behaviour in response to the deflection of the moving gate. From the simulations, we predict that the proposed biasing offers a device sensitive to even sub-nanometer deflection of the flexible gate. As a proof of concept, we fabricate fixed-fixed beams which utilize CMOS-MEMS fabrication. The process involves six lithography steps which involve two CMOS and the remaining MEMS fabrication. The fabricated beams are mechanically characterized for resonance. Further, we carry out electrical characterization for I-V (current-voltage) characteristics. The second part of the thesis focusses on a novel biasing method which circumvents the need of signal conditioning circuitry needed in a capacitive based transduction due to inbuilt amplification. Extensive simulations with equivalent circuit has been carried out to determine the increased sensitivity and the role of various design variables.

Le bruit, en particulier celui dû au trafic routier, est cité par de nombreuses études comme une source de préoccupation sociétale majeure. Jusqu'à présent les réponses des pouvoirs publics ne se basent que sur une quantification énergétique de l'exposition sonore, souvent via la mesure ou l'estimation du LA ou du Lden, et des prises de décisions relatives à la diminution du niveau sonore. Or des études psychoacoustiques ont montré que le niveau sonore n'expliquait qu'une faible part de la gêne sonore ressentie. Il est donc intéressant d'avoir plus d'information sur la source de bruit et de ne pas la réduire à un simple niveau sonore. Dans cette thèse, nous proposons de concevoir un outil permettant d'estimer la gêne sonore associée à chaque véhicule du trafic routier via l'utilisation de son signal audio et de modèles de gêne sonore. Pour ce faire, le signal audio du véhicule est isolé de l'ensemble du trafic routier urbain grâce à l'utilisation de méthodes inverses et de grands réseaux de microphones ainsi que du traitement d'images pour obtenir sa trajectoire. Grâce à la connaissance de la trajectoire ainsi que du signal, le véhicule est classifié par une méthode de machine learning suivant la taxonomie de Morel et al. Une fois sa catégorie obtenue, la gêne spécifique du véhicule est estimée grâce à un modèle de gêne sonore utilisant des indices psychoacoustiques et énergétiques. Cela permet l'estimation des gênes sonores spécifiques à chaque véhicule au sein du trafic routier. L'application de cette méthode est faite lors d'une journée de mesure sur une grande artère parisienne. / Noise, especially road traffic noise, is cited by many studies as a source of major societal concern. So far, public responses are based only on energy quantification of sound exposure, often by measuring or estimating LA or Lden, and sound-level reduction related decision are taken. Nevertheless, psychoacoustic studies have shown that the sound level explains only a small part of the perceived noise annoyance. It is interesting to have more information about the source of noise and not to reduce the information to its sound level. In this thesis a tool is proposed for estimating the noise annoyance induced by each road vehicle using its audio signal and noise annoyance models. To do so, the audio signal of the vehicle is isolated by using inverse methods, large microphones arrays and image processing to obtain its trajectory. The knowledge of the trajectory and of the signal allows the vehicle to be classified by a machine learning method according to Morel et al. taxonomy. Once its category obtained, the specific annoyance of the vehicle is estimated thanks to a noise annoyance model using psychoacoustic and energetic indices. This allows the estimation of specific noise annoyance for each vehicle within the road traffic. The application of this method is made during a measurement day on a large Parisian artery.

Ce travail de thèse traite du développement d’une antenne microphonique compacte et d’une chaîne de traitement du signal dédiée, pour la reconnaissance et la localisation angulaire de cibles aériennes. L’approche globale proposée consiste en une détection initiale de cible potentielle, la localisation et le suivi de la cible, et une détection affinée par un filtrage spatial adaptatif informé par la localisation de la cible. Un algorithme original de localisation goniométrique est proposé. Il utilise l’algorithme RANSAC sur des données pression-vitesse large bande [100 Hz - 10 kHz], estimées en temps réel, dans le domaine temporel, par des différences et sommes finies avec des doublets de microphones à espacements inter-microphoniques adaptés à la fréquence. L’extension de la bande passante de l’antenne en hautes fréquences est rendue possible par l’utilisation de différences finies d’ordre élevé, ou de variantes de la méthode PAGE (Phase and Amplitude Gradient Estimation) adaptées à l’antenne développée. L’antenne acoustique compacte ainsi développée utilise 32 microphones MEMS numériques répartis dans le plan horizontal sur une zone de 7.5 centimètres, selon une géométrie d’antenne adaptée aux l’algorithmes de localisation et de filtrage spatial employés. Des essais expérimentaux de localisation et de suivi de trajectoire contrôlée par une sphère de spatialisation dans le domaine ambisonique ont montré une erreur de localisation moyenne de 4 degrés. Une base de données de signatures acoustiques de drones en vol a été créée, avec connaissance de la position du drone par rapport à l’antenne microphonique apportée par des mesures GPS. L’augmentation des données par bruitage artificiel, et la sélection dedescripteurs acoustiques par des algorithmes évolutionnistes, ont permis de détecter un drone inconnu dans un environnement sonore inconnu jusqu’à 200 mètres avec le classifieur JRip. Afin de faciliter la détection et d’en augmenter la portée, l’étape de détection initiale est précédée d’une formation de voies différentielle dans 4 directions principales (nord, sud, est, ouest), et l’étape de détection affinée est précédée d’une formation de voies de Capon informée par la localisation et le suivi de la cible à identifier. / This thesis deals with the development of a compact microphone array and a dedicated signal processing chain for aerialtarget recognition and direction of arrival (DOA) estimation. The suggested global approach consists in an initial detection ofa potential target, followed by a DOA estimation and tracking process, along with a refined detection, facilitated by adaptivespatial filtering. An original DOA estimation algorithm is proposed. It uses the RANSAC algorithm on real-time time-domainbroadband [100 Hz - 10 kHz] pressure and particle velocity data which are estimated using finite differences and sums ofsignals of microphone pairs with frequency-dependent inter-microphone spacings. The use of higher order finite differences, or variants of the Phase and Amplitude Gradient Estimation (PAGE) method adapted to the designed antenna, can extend its bandwidth at high frequencies. The designed compact microphone array uses 32 digital MEMS microphones, horizontally disposed over an area of 7.5 centimeters. This array geometry is suitable to the implemented algorithms for DOA estimation and spatial filtering. DOA estimation and tracking of a trajectory controlled by a spatialization sphere in the Ambisonic domain have shown an average DOA estimation error of 4 degrees. A database of flying drones acoustic signatures has been set up, with the knowledge of the drone’s position in relation to the microphone array set out by GPS measurements. Adding artificial noise to the data, and selecting acoustic features with evolutionary programming have enabled the detection of an unknown drone in an unknown soundscape within 200 meters with the JRip classifier. In order to facilitate the detection and extend its range, the initial detection stage is preceded by differential beamforming in four main directions (north, south, east, west), and the refined detection stage is preceded by MVDR beamforming informed by the target’s DOA.

In this thesis, nondestructive surface wave measurements are presented for characterization of dynamic modulus and layer thickness on different pavements and cement concrete slabs. Air-coupled microphones enable rapid data acquisition without physical contact with the pavement surface. Quality control of asphalt concrete pavements is crucial to verify the specified properties and to prevent premature failure. Testing today is primarily based on destructive testing and the evaluation of core samples to verify the degree of compaction through determination of density and air void content. However, mechanical properties are generally not evaluated since conventional testing is time-consuming, expensive, and complicated to perform. Recent developments demonstrate the ability to accurately determine the complex modulus as a function of loading time (frequency) and temperature using seismic laboratory testing. Therefore, there is an increasing interest for faster, continuous field data evaluation methods that can be linked to the results obtained in the laboratory, for future quality control of pavements based on mechanical properties. Surface wave data acquisition using accelerometers has successfully been used to determine dynamic modulus and thickness of the top asphalt concrete layer in the field. However, accelerometers require a new setup for each individual measurement and are therefore slow when testing is performed in multiple positions. Non-contact sensors, such as air-coupled microphones, are in this thesis established to enable faster surface wave testing performed on-the-fly. For this project, a new data acquisition system is designed and built to enable rapid surface wave measurements while rolling a data acquisition trolley. A series of 48 air-coupled micro-electro-mechanical sensor (MEMS) microphones are mounted on a straight array to realize instant collection of multichannel data records from a single impact. The data acquisition and evaluation is shown to provide robust, high resolution results comparable to conventional accelerometer measurements. The importance of a perfect alignment between the tested structure’s surface and the microphone array is investigated by numerical analyses. Evaluated multichannel measurements collected in the field are compared to resonance testing on core specimens extracted from the same positions, indicating small differences. Rolling surface wave measurements obtained in the field at different temperatures also demonstrate the strong temperature dependency of asphalt concrete. A new innovative method is also presented to determine the thickness of plate like structures. The Impact Echo (IE) method, commonly applied to determine thickness of cement concrete slabs using an accelerometer, is not ideal when air-coupled microphones are employed due to low signal-to-noise ratio. Instead, it is established how non-contact receivers are able to identify the frequency of propagating waves with counter-directed phase velocity and group velocity, directly linked to the IE thickness resonance frequency. The presented non-contact surface wave testing indicates good potential for future rolling quality control of asphalt concrete pavements. / <p>QC 20170209</p>

Time-Frequency (TF) masking is an audio processing technique useful for isolating an audio source from interfering sources. TF masking has been applied and studied in monaural and binaural applications, but has only recently been applied to distributed microphone arrays. This work focuses on evaluating the TF masking technique's ability to isolate human speech and improve speech intelligibility in an immersive "cocktail party" environment. In particular, an upper-bound on TF masking performance is established and compared to the traditional delay-sum and general sidelobe canceler (GSC) beamformers. Additionally, the novel technique of combining the GSC with TF masking is investigated and its performance evaluated. This work presents a resource-efficient method for studying the performance of these isolation techniques and evaluates their performance using both virtually simulated data and data recorded in a real-life acoustical environment. Further, methods are presented to analyze speech intelligibility post-processing, and automated objective intelligibility measurements are applied alongside informal subjective assessments to evaluate the performance of these processing techniques. Finally, the causes for subjective/objective intelligibility measurement disagreements are discussed, and it was shown that TF masking did enhance intelligibility beyond delay-sum beamforming and that the utilization of adaptive beamforming can be beneficial.

Este trabalho apresenta a implementação prática de uma rede de microfones para ser utilizada em tempo real. A solução proposta envolve o uso de hardware de baixo custo e software de código aberto. Em termos de hardware, a rede de microfones utilizou dispositivos de áudio USB conectados diretamente a um computador pessoal (PC). Em termos de software, foram utilizados a biblioteca de código aberto Advanced Linux Sound Architecture (ALSA) e o sistema operacional Linux. Algumas implementações foram realizadas na biblioteca ALSA para que fosse possível a utilização da rede de microfones dentro do Linux. Os algoritmos implementados na biblioteca ALSA foram o Delay and Sum, Generalized Sidelobe Canceller (GSC) e o Post-Filtering. Os aspectos teóricos dos principais algoritmos empregados nas redes de microfones foram abordados de forma extensa. Os resultados teóricos e práticos desta implementação são apresentados no final deste trabalho. Todo o desenvolvimento de software foi publicado na Internet para que sirva de base para futuros trabalhos. / This work presents the practical implementation of a microphone array to be used in real time. The proposed solution involves the use of low-cost hardware and open source software. In terms of hardware, the microphone array used USB audio devices connected directly to a personal computer (PC). In terms of software, it was used the open-source library Advanced Linux Sound Architecture (ALSA) and Linux operating system. Some implementations were carried out in ALSA library to make it possible to use the microphone array within Linux. The algorithms implemented in ALSA library were the Delay and Sum, Generalized Sidelobe Canceller (GSC) and Post-Filtering. The theory of the main algorithms used in microphone array were discussed extensively. The results for the theoretical and practical implementation are presented at the end of this work. All software development was published on the Internet to serve as a basis for future works.

Ovarian cancer is one of the five most common causes of cancer death in women in the USA and UK. It is usually diagnosed when it is well established beyond the ovary in the peritoneum. Intravenous injection of cisplatin is a common palliative therapy for ovarian cancer patients. Intraperitoneal therapy has been shown to improve survival for patients. Poly(lactide-co-glycolide) (PLGA) is a biodegradable polyester which has been proven safe for medical implantation. PLGA microspheres or fibres have been considered in this work as depots for delivering intraperitoneal cisplatin directly to the tumour site. The aims of this work were (1) to develop microsphere depot formulations with improved drug release profiles compared to previous work; (2) Novel cisplatin containing solid and hollow fibres were to be developed and investigated as alternative structures for depot devices; (3) The drug release profiles were to be examined using mathematical models to allow rational comparison of the devices. It was found that cisplatin containing PLGA 65:35 solid and hollow fibres represent a novel, reproducible formulation for encapsulating higher amounts of cisplatin for an equivalent mass of excipient than other polymer formulations. The fibres developed in this study were able to maintain elevated concentrations of unbound cisplatin in the presence of a biological matrix for approximately 100 hours in vitro.

Este trabalho apresenta a implementação prática de uma rede de microfones para ser utilizada em tempo real. A solução proposta envolve o uso de hardware de baixo custo e software de código aberto. Em termos de hardware, a rede de microfones utilizou dispositivos de áudio USB conectados diretamente a um computador pessoal (PC). Em termos de software, foram utilizados a biblioteca de código aberto Advanced Linux Sound Architecture (ALSA) e o sistema operacional Linux. Algumas implementações foram realizadas na biblioteca ALSA para que fosse possível a utilização da rede de microfones dentro do Linux. Os algoritmos implementados na biblioteca ALSA foram o Delay and Sum, Generalized Sidelobe Canceller (GSC) e o Post-Filtering. Os aspectos teóricos dos principais algoritmos empregados nas redes de microfones foram abordados de forma extensa. Os resultados teóricos e práticos desta implementação são apresentados no final deste trabalho. Todo o desenvolvimento de software foi publicado na Internet para que sirva de base para futuros trabalhos. / This work presents the practical implementation of a microphone array to be used in real time. The proposed solution involves the use of low-cost hardware and open source software. In terms of hardware, the microphone array used USB audio devices connected directly to a personal computer (PC). In terms of software, it was used the open-source library Advanced Linux Sound Architecture (ALSA) and Linux operating system. Some implementations were carried out in ALSA library to make it possible to use the microphone array within Linux. The algorithms implemented in ALSA library were the Delay and Sum, Generalized Sidelobe Canceller (GSC) and Post-Filtering. The theory of the main algorithms used in microphone array were discussed extensively. The results for the theoretical and practical implementation are presented at the end of this work. All software development was published on the Internet to serve as a basis for future works.

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