The Generalized Sidelobe Canceller is an adaptive algorithm for optimally estimating the parameters for beamforming, the signal processing technique of combining data from an array of sensors to improve SNR at a point in space. This work focuses on the algorithm’s application to widely-separated microphone arrays with irregular distributions used for human voice capture. Methods are presented for improving the performance of the algorithm’s blocking matrix, a stage that creates a noise reference for elimination, by proposing a stochastic model for amplitude correction and enhanced use of cross correlation for phase correction and time-difference of arrival estimation via a correlation coefficient threshold. This correlation technique is also applied to a multilateration algorithm for an efficient method of explicit target tracking. In addition, the underlying microphone array geometry is studied with parameters and guidelines for evaluation proposed. Finally, an analysis of the stability of the system is performed with respect to its adaptation parameters.

Thesis (M.S.)--State University of New York at Binghamton, Thomas J. Watson School of Engineering and Applied Science, Department of Electrical and Computer Engineering, 2009. / Includes bibliographical references.

Thesis (M.S.)--State University of New York at Binghamton, Thomas J. Watson School of Engineering and Applied Science, Department of Electrical and Computer Engineering, 2008. / Includes bibliographical references.

This thesis studies how the mechanical design of a surveillance camera affects the acoustic performance, mainly in terms of the frequency response within the human hearing range. During the project, the mechanical characteristics that affect frequency response were investigated by measuring the camera’s audio behavior in an anechoic chamber. A theoretical and adaptable acoustic model was built in COMSOL to simulate the frequency response of the sound path. Measurement and simulation results were compared to identify critical aspects of the mechanical design and adjust accordingly for better acoustic performance.

Avaliou-se a eficácia de um procedimento de teleconsulta baseado na internet para a verificação do aparelho de amplificação sonora individual (AASI) por meio da realização de medidas com microfone sonda em adultos deficientes auditivos. Foram comparados os valores da amplitude da resposta de ressonância da orelha externa (REUR), resposta de ressonância da orelha externa com uso da amplificação (REAR) e o ganho de inserção (REIG) realizados face a face (F) e à distância (D). O erro casual das medidas da REUR, REAR e REIG realizadas pelos dois métodos foi analisado. Participaram do estudo: grupo A: sessenta adultos (média de idade de 67 anos) com deficiência auditiva unilateral (n=15) ou bilateral (n=45), totalizando 105 orelhas; grupo B: 19 adultos (média de idade de 28 anos) sem queixas auditivas (19 orelhas). Os AASIs utilizados foram do tipo mini-retroauricular e não possuíam microfone direcional ou algoritmos de cancelamento de retroalimentação e/ou redução de ruído. Para o grupo B foi feito ajuste do AASI resultando na menor saída possível. Cinco fonoaudiólogos voluntários auxiliaram na coleta dos dados à distância. O procedimento face a face foi realizado pela pesquisadora em todos os participantes, utilizando-se o equipamento Unity PC Probe Mic (Siemens) conectado a um computador pessoal. Para realização das medidas à distância esse computador foi conectado à LAN (10 Mpbs). O participante e o fonoaudiólogo voluntário se posicionaram na sala junto ao equipamento (ambiente de teste). A pesquisadora posicionou-se em outra sala denominada ambiente remoto juntamente com um notebook também conectado à LAN. Webcams e headsets foram utilizados para a captura de audio e video, transmitidas em tempo real pelo software Polycom PVX, o qual também foi utilizado para compartilhamento de aplicativos. Por meio desse compartilhamento a pesquisadora realizava as medidas diretamente no participante, com auxílio do fonoaudiólogo voluntário. Quatro repetições das medidas da REUR, REAR e REIG foram obtidas apenas para o grupo B. Correlações (Pearson) muito fortes e significativas foram obtidas entre as medidas realizadas face a face (F) e à distância (D). O teste t pareado revelou diferenças pequenas, porém significativas entre as medidas F e D para a REUR e REAR. A ANOVA mostrou diferenças significativas entre as repetições realizadas. Diferenças entre os métodos F e D foram obtidas apenas na freqüência de 2000 Hz. Os erros causais obtidos pelas medidas face a face e à distância foram muito semelhantes. As diferenças e variações encontradas entre as medidas face a face e à distância não foram maiores do que a magnitude de variabilidade do próprio procedimento de medidas com microfone sonda. É possível realizar medidas de microfone sonda confiáveis via teleconsulta / telessaúde baseada na internet. Outras investigações são necessárias para validar esse procedimento. / The efficacy of an internet based teleconsultation for hearing aid probe microphone measures in hearing impaired adults was evaluated. The amplitudes of face to face (F) and remote (R) real ear unaided response (REUR), real ear aided response (REAR) and real ear insertion gain (REIG) were compared. The measurement error for repeated measures of REUR, REAR and REIG was analyzed. Participated in this study: group A: 60 adults (mean age: 67 years), with unilateral (n=15) or bilateral (n=45) hearing loss, totalizing 105 ears; group B: 19 normal hearing adults (mean age: 28 years) totalizing 19 ears. Behind the ear hearing aids with no directional microphone, noise reduction and/or feedback cancellation were used. For group B these hearing aids were adjusted as to produce the least amount of gain and output. Five audiologists volunteers helped the data collection. The Unity PC Probe Mic (Siemens) coupled to a personal computer were used by the researcher to carry out face to face measures. For the remote measures this equipment was connected to a local area network (LAN 10 Mpbs). The volunteer and the participant were located in this room (test site). At the remote site the researcher used a notebook connected to the LAN. Webcams and headsets were used for audio and video capture which was transmitted in real time by the Polycom PVX software, which was also used for application sharing. By means of remote controlling of the PC Probe Mic equipment the researcher could perform the remote real ear measurements in the participants. Four repeated measures of REUR, REAR and REIG were obtained for group B only. Strong and significant correlations (Pearson) were obtained between face to face and remote real ear measures. Paired t tests revealed small but significant differences between face to face and remote REUR and REAR. Analysis of variance showed significant differences between repeated measures. Measurement errors for face to face and remote real ear measures were very similar. The differences as well as errors found between face to face and remote measures were never higher than the reported variability for probe microphone measures themselves. It is possible to carry out probe microphone measures by means of teleconsultation / telehealth. Further investigations to validate this procedure are necessary.

För att uppfylla kraven på kvalitetssäkring samt evidensbaserad praktik måste hörapparatanpassning utvärderas objektivt. Syftet med studien är att redovisa nya forskningsresultat gällande objektiv utvärdering av hörapparater med hjälp av hörselgångsmätning, samt att beskriva en modern, evidensbaserad och kliniskt användbar metod för de vanligaste typerna av elektro-akustiska hörapparater med olika typer av insats. Studien består av en allmän litteraturstudie som beskriver tidigare rekommendationer samt en systematisk litteraturstudie där nya rön gällande metodiken vid hörselgångsmätning presenteras. Utifrån hörselgångsmätningens olika delmoment undersöker studien vilka konsekvenser de nya rönen har för utförandet vid hörselgångsmätning. I slutsatsen presenteras en metod för objektiv utvärdering av moderna hörapparater.

A method to synthesize the sound radiation characteristics of musical instruments in a wave field synthesis (WFS) system is proposed and tested. Radiation patterns of a violin are measured with a circular microphone array which consists of 128 pressure receivers. For each critical frequency band one exemplary radiation pattern is decomposed to circular harmonics of order 0 to 64. So the radiation characteristic of the violin is represented by 25 complex radiation patterns. On the reproduction side, these circular harmonics are approximated by 128 densely spaced monopoles by means of 128 broadband impulses. An anechoic violin recording is convolved with these impulses, yielding 128 filtered versions of the recording. These are then synthesized as 128 monopole sources in a WFS system and compared to a virtual monopole playing the unfiltered recording. The study participants perceive the tone color of the recreated virtual violin as being dependent on the listening position and report that the two source types have a different ‘presence’. The test persons rate the virtual violin as less natural, sometimes remarking that the filtering is audible at high frequencies. Further studies with a denser spacing of the virtual monopoles and a presentation in an anechoic room are planned.

Systémy rozpoznávání řeči v dnešní době dosahují poměrně vysoké úspěšnosti. V případě řeči, která je snímána vzdáleným mikrofonem a je tak narušena množstvím šumu a dozvukem (reverberací), je ale přesnost rozpoznávání značně zhoršena. Tento problém je možné zmírnit využitím mikrofonních polí. Tato práce se zabývá technikami, které umožňují kombinovat signály z více mikrofonů tak, aby byla zlepšena kvalita výsledného signálu a tedy i přesnost rozpoznávání. Práce nejprve shrnuje teorii rozpoznávání řeči a uvádí nejpoužívanější algoritmy pro zpracování mikrofonních polí. Následně jsou demonstrovány a analyzovány výsledky použití dvou metod pro beamforming a metody dereverberace vícekanálových signálů. Na závěr je vyzkoušen alternativní způsob beamformingu za použití neuronových sítí.

The aim of the thesis is to study mechanical properties of nanomaterials (multi-layer graphene, silicon, mica) suitable to be used as novel pressure sensors in laser photoacoustic spectroscopy. Membranes (diameter ~ 4 mm, thickness ~ 100 nm) were prepared by mechanical exfoliation method and then attached to a glass window in several slightly different designs. Movement of these membranes was detected using HeNe laser beam reflected from the membrane's surface onto a position sensitive detector. Methanol was used as a model gas and the signal was collected from studied element and microphone simultaneously. Acoustic wave, induced inside a measuring cell by periodic thermal variations, causes the membranes to move. The movement of a membrane is influenced by its mechanical properties, which is possible to determine by fitting the measured data into a mathematical model. Comparison of the output data of all membranes' measurements shows, that the signal intensity is influenced by the method of attaching membrane to a glass window and by volume of free space on a side of a membrane. Metallization of the membrane's surface (~ 70 nm) decreases its springiness thus decreases the sensitivity. Several membranes reached sensitivity comparable with top class microphone.

This thesis project aims for the development of a demonstration platform on mobile devices for testing and demonstrating algorithms for 2D and 3D spatial sound reproduction. The demo system consists of four omnidirectional microphones in a square planar array, an Octo sound card (from Audio Injector), a Raspberry Pi 3B+ (R-Pi) single-board computer and an inertial measurement unit (IMU) located in the center of the array. The microphone array captures sound, which is then digitized, and in turn, transferred to the R-Pi. On the R-Pi, the digitized sound signal is rendered through the directional audio coding (DirAC) algorithm to maintain the spatial properties of the sound. Finally, the digital signal and spatial properties are rendered through Dirac VR to maintain a spatial stereo signal of the recorded environment. The directional audio coding algorithm was initially implemented in Matlab and then ported to C++ since the R-Pi does not support Matlab natively. The ported algorithm was verified on a four-channel in and six-channel out system, processing 400 000 samples at 44 100 kHz. The results show that the C++ DirAC implementation maintained a maximum error of 4.43e-05 or -87 dB compares to the original Matlab implementation. For future research on spatial audio reproduction, a four-microphone smartphone mock-up was constructed based on the same hardware used in the demo system. A software interface was also implemented for transferring the microphone recordings and the orientation of the mock-up to Matlab.