Bioacoustics is a relatively new field of research focused on studying the acoustic signals of vocal animal species. The field has been a topic of interest for many years due its passive approach and avoidance of species-level limitations, such as tracking rare or nocturnal species. It has been used to locate birds in terrestrial environments; however, localization in urban environments remains unstudied. This research aims to fill the gap by attempting to estimate the location of 30 discrete calls in eight unique, urban environments. Sites represented two distinct traffic scenarios: moderate traffic and high traffic. Three system arrays of three different sizes utilizing the Song Meter SM2+ units were tested at each site to determine the effect of array size on call visibility and location estimation. An American robin (Turdus migratorius) distress call was played through a loudspeaker at the thirty locations for each array. The spectrogram of each of these calls was examined to determine the number of channels with a visible call signature. If the file contained at least one visible call per song meter (36% of our sound files), cross correlation was used to determine the differences in the time of arrival of calls at all the microphones in the array, called lag values, which were used to calculate the origin location of the call. However, resulting lag values in this study were too large to produce reliable location estimates. This was likely due to imprecise synchronization in the field or poorly defined calls within the spectrograms. Our overall low visibility is likely a result of the high signal to noise ratio common in urban environments. Further research is necessary to continue to test the viability of acoustic localization in urban environments.

This thesis considers the design of a beamformer which can enhance desired signals in an environment consisting of broadband nearfield and/or farfield sources. The thesis contains: a formulation of a set of analysis tools which can provide insight into the intrinsic structure of array processing problems; a methodology for nearfield beamforming; theory and design of a general broadband beamformer; and a consideration of a coherent nearfield broadband adaptive beamforming problem. To a lesser extent, the source localization problem and background noise modeling are also treated. ¶: A set of analysis tools called modal analysis techniques which can be used to a solve wider class of array signal processing problems, is first formulated. The solution to the classical wave equation is studied in detail and exploited in order to develop these techniques. ¶: Three novel methods of designing a beamformer having a desired nearfield broadband beampattern are presented. The first method uses the modal analysis techniques to transform the desired nearfield beampattern to an equivalent farfield beampattern. A farfield beamformer is then designed for a transformed farfield beampattern which, if achieved, gives the desired nearfield pattern exactly. The second method establishes an asymptotic equivalence, up to complex conjugation, of two problems: (i) determining the nearfield performance of a farfield beampattern specification, and (ii) determining the equivalent farfield beampattern corresponding to a nearfield beampattern specification. Using this reciprocity relationship a computationally simple nearfield beamforming procedure is developed. The third method uses the modal analysis techniques to find a linear transformation between the array weights required to have the desired beampattern for farfield and nearfield, respectively. ¶: An efficient parameterization for the general broadband beamforming problem is introduced with a single parameter to focus the beamformer to a desired operating radius and another set of parameters to control the actual broadband beampattern shape. This parameterization is derived using the modal analysis techniques and the concept of the theoretical continuous aperture. ¶: A design of an adaptive beamformer to operate in a signal environment consisting of broadband nearfield sources, where some of interfering signals may be correlated with desired signal is also considered. Application of modal analysis techniques to noise modeling and broadband coherent source localization conclude the thesis.

L'étalonnage absolu des microphones de mesure en acoustique repose sur l'obtention d'étalons primaires, eux-mêmes étalonnés suivant un protocole sophistiqué (méthode de réciprocité) et conformément aux normes en vigueur ; ces normes ont fait l'objet d'améliorations au cours des dernières décennies mais laissent toujours des zones d'ombre. Parallèlement, la caractérisation appropriée des oreilles artificielles, nécessaire au réglage des audiomètres et par suite à leur étalonnage, fait aujourd'hui défaut.<br /><br />Ce propos met en cause la précision de l'étalonnage des étalons de mesure de pressions acoustiques et l'insuffisance des réglages d'appareils médicaux largement utilisés. Les enjeux pratiques, techniques et scientifiques ont donc leur importance et les études à mener comportent des exigences qui nécessitent des recherches approfondies. C'est ainsi que les thèmes abordés font appel ici à la méthode de la réciprocité en cavité et en champ libre.<br />Dans la première partie du travail, l'objectif recherché est d'adapter et d'améliorer la méthode de réciprocité en cavité. L'adaptation de cette méthode conduit à une technique de mesure d'impédances d'entrée de petits éléments acoustiques, tels que des tubes, fentes, cavités (utilisés dans l'oreille artificielle). L'amélioration des incertitudes de mesure des efficacités recherchée pour les hautes fréquences a conduit à proposer une modélisation améliorée d'un microphone ainsi que du dispositif d'étalonnage dans sa globalité de manière à étudier l'influence des modes radiaux dans la cavité sur les résultats de l'étalonnage.<br /><br />La deuxième partie de ce travail trouve son origine dans une comparaison clé à l'échelle internationale portant sur les techniques d'étalonnage des microphones en champ libre. Cette comparaison clé a nécessité une refonte complète du dispositif expérimental du LNE, des techniques d'acquisitions et des méthodes de filtrage des perturbations liées aux faibles niveaux acoustiques mis en jeu. Ce travail a conduit à entreprendre des études plus approfondies sur les plans analytique et expérimental du concept de centre acoustique d'un microphone.<br /><br />Certains résultats obtenus posent les bases des travaux futurs qui devraient permettre de poursuivre la modélisation pour réduire les incertitudes mais également pour prévoir la mise en oeuvre des méthodes adaptées à la métrologie des capteurs du futur qui seront fabriqués par des procédés relevant des microtechnologies.

Since the micro-electro mechanical system microphone, MEMS microphone, has the advantages of superior sound quality, low power consumption, higher temperature resistance and anti-noise ability in used. The researchers therefore have studied the functions of MEMS microphone since 1980s. The MEMS microphones is applied as the part of 3G mobile phone in the market. Though the functions of microphone are improved by manufacturing process technique and new material designed, this study tends to provide a new, low-cost and rapid design idea to gain the performance in chamber of microphone. Taguchi method and BEASY software, which is boundary element method, are combined to evaluate the results of the design in sound field. Taguchi method is a famous method in industrial design to find out relations between system parameters and chamber size. BEASY is a tool for sound field analysis in the research. The result from Taguchi method appears the sound pressure level gain about 2.2 dB to 2.4 dB due to the change of microphone chamber size only. It is also interested in studying the optimization design for position of microphone. It is displayed that the location of port is closer to the boundary of chip will also increase about 0.3 dB to 0.6dB sound pressure level in sound field. The higher frequency of sound source will also create larger sound pressure level at two corners on the port.

No description available.

To achieve the miniaturization and high performance of the mobile phone, notebook, hearing aid and personal digital assistant (PDA), many researchers focus on the developing a new-type microphone with very small dimension, high quality and low manufacturing cost utilizing MEMS technology. By using the surface and bulk micromachining technologies, this thesis designed and fabricated a capacitive MEMS microphone with a polyimide bcakplate microstructure. The main processing steps adopted in this study include five photolithoghaphies and seven thin-film depositions. A MEMS-based microphone with an only 2¡Ñ2 mm2 sensing area of the floating Si3N4/Poly-Si/Si3N4 membrane and a 2 £gm-height gap distance between the top and bottom electrodes was implemented and characterized. Measured in a special isolated-box and under 1 kHz audio frequency, a -60.3 dB/Pa sensitivity (deducted the 22.6 dB output gain of the pre-amplifier) and a 51 dB signal to noise ratio (SNR) of the implemented MEMS microphone can be obtained as the biasing voltage only about 3 volts. The very low driving voltage, moderate SNR and sensitivity demonstrated in this work keep abreast with the results of many outstanding research laboratories in the world.

The focus of this study is the optoelectronic integration of a micro-optical displacement detection architecture with a biomimetic MEMS microphone membrane based on the directional hearing mechanism of the parasitic fly Ormia Ochracea. The micromachined microphones feature optical interferometric displacement detection achieved using a commercially available Vertical Cavity Surface Emitting Laser (VCSEL) coupled with a custom designed silicon photodiode array. This design is shown to have significant advantages over conventional hearing aid microphones, which employ capacitive detection. A Multi-Chip Module (MCM) optoelectronic package is designed to integrate the biomimetic membrane with the optical displacement detection electronics in order to produce a fully integrated acoustic sensor. The modular package components, which are fabricated using high resolution stereolithography apparatus (SLA) equipment, provide accurate optical alignment of the optoelectronic components and allow complete device integration in a package with a total volume under 0.5cc. Characterization of the integrated microphones is described in detail, including measurements of sensitivity, noise floor and directivity. A displacement resolution of 3.5x10⁻¹³ m/√Hz was measured between 4kHz and 16kHz in an anechoic test chamber, corresponding to a dynamic range of 115dB for the optical detection architecture. The total noise SPL of the device is 35.9dBA. Unlike capacitive microphones with similar noise levels, the device developed in this work exhibits first order dipole directivity patterns between 250Hz-1kHz, with an ideal Directivity Index of 4.8dB @ 1kHz and directional attenuation exceeding 25dB. With these results the optoelectronic package presented in this work demonstrates the viability of the integrated optical biomimetic microphones in compact, low power applications, specifically directional hearing aids.

Thesis (Ph.D)--Electrical and Computer Engineering, Georgia Institute of Technology, 2009. / Committee Chair: Hasler, Paul; Committee Co-Chair: Degertekin, Levent; Committee Member: Anderson, David; Committee Member: Ayazi, Farrokh; Committee Member: Brand, Oliver; Committee Member: Hesketh, Peter. Part of the SMARTech Electronic Thesis and Dissertation Collection.

The main purpose of this research is to discuss the specificity of the microphone as a gestural tool through which the sequence organization of media two-party interviews is accomplished. The study focuses on the practical communicative problems of microphone operations in a media setting where the parties have alternating turns, and addresses the question of who of the participants speaks next and for how long. It is particularly concerned with investigating what the participating parties can do with the microphone, what it accomplishes, and how it is used as a tool for interaction with an audience. We particularly focus on how microphone moves can make use see how how people orient to emergent content structures in talk, and how microphone performs as a device for confirmation of verbal turns.

The detection of sound sources with microphone arrays can be enhanced through processing individual microphone signals prior to the delay and sum operation. One method in particular, the Phase Transform (PHAT) has demonstrated improvement in sound source location images, especially in reverberant and noisy environments. Recent work proposed a modification to the PHAT transform that allows varying degrees of spectral whitening through a single parameter, andamp;acirc;, which has shown positive improvement in target detection in simulation results. This work focuses on experimental evaluation of the modified SRP-PHAT algorithm. Performance results are computed from actual experimental setup of an 8-element perimeter array with a receiver operating characteristic (ROC) analysis for detecting sound sources. The results verified simulation results of PHAT- andamp;acirc; in improving target detection probabilities. The ROC analysis demonstrated the relationships between various target types (narrowband and broadband), room reverberation levels (high and low) and noise levels (different SNR) with respect to optimal andamp;acirc;. Results from experiment strongly agree with those of simulations on the effect of PHAT in significantly improving detection performance for narrowband and broadband signals especially at low SNR and in the presence of high levels of reverberation.