Background noise is particularly damaging to speech intelligibility for people with hearing loss. The problem of reducing noise in hearing aids is one of great importance and great difficulty. Over the years, many solutions and different algorithms have been implemented in order to provide the optimal solution to the problem. Beamforming has been used for a long time and has therefore been extensively researched. Studying the performance of Minimum Variance Distortionless Response (MVDR) beamforming with a three- and four- microphone array compared to the conventional two-microphone array, the aim is to implement a speech signal enhancement and a noise reduction algorithm. By using multiple microphones, it is possible to achieve spatial selectivity, which is the ability to select certain signals based on the angle of incidence, and improve the performance of noise reduction beamformers. This thesis proposes the use of beamforming, an existing technique in order to create a new way to reduce noise transmitted by hearing aids. In order to reduce the complexity of that system, we use hybrid cascades, which are simpler beamformers of two inputs each and connected in series. The configurations that we consider are a three-microphone linear array (monaural beamformer), a three-microphone configuration with a two-microphone linear array and the 3rd microphone in the ear (monaural beamformer), a three-microphone configuration with a two-microphone linear array and the 3rd microphone on contra-lateral ear (binaural beamformer), and finally four-microphone configurations. We also investigate the performance improvement of the beamformer with more than two microphones for the different configurations, against the two-microphone beamformer reference. This can be measured by using objective measurements, such as the amount of noise suppression, target energy loss, output SNR, speech intelligibility index and speech quality evaluation. These objective measurements are good indicators of subjective performance. In this project, we prove that most hybrid structures can perform satisfyingly well compared to the full complexity beamformer. The low complexity beamformer is designed with a fixed target location (azimuth), where its weights are calibrated with respect to a target signal located in front of the listener and for a diffuse noise field. Both second- and third- order beamformers are tested in different acoustic scenarios, such as a car environment, a meeting room, a party occasion and a restaurant place. In those scenarios, the target signal is not arriving at the hearing aid directly from the front side of the listener and the noise field is not always diffuse. We thoroughly investigate what are the performance limitations in that case and how well the different cascades can perform. It is proven that there are some very critical factors, which can affect the performance of the fixed beamformer, concerning all the hybrid structures that were examined. Finally, we show that lower complexity cascades for both second- and third- order beamformers can perform similarly well as the full complexity beamformers when tested for a set of multiple Head Related Transfer Functions (HRTFs) that correspond to a real head shape.

Aeroacoustic emissions were identified as a primary concern in the public acceptance of wind turbines. A review of literature involving sound localization was undertaken and led to the design of two microphone arrays to identify acoustic sources. A small-scale array composed of 27 sensors was produced with the intention of improving the quality of sound measurements over those made by a single microphone in a small, closed-loop wind tunnel. A large-scale array containing 30 microphones was also implemented to allow for measurements of aeroacoustic emissions from airfoils and rotating wind turbines. To minimize cost and pursue alternative sensor technologies, microelectromechanical microphones were selected for the array sensors and assembled into the arrays on printed circuit boards. Characterization of the microphones was completed using a combination of calibration techniques, primarily in a plane wave tube. Array response to known sources was quantified by analyzing source maps with respect to source location accuracy, beamwidth, and root mean square error. Multiple sources and rotating sources were tested to assess array performance. Following validation with known sources, wind tunnel testing of a 600 watt wind turbine was performed at freestream speeds of 2.5 m/s, 3.5 m/s, 4.5 m/s, and to 5.5 m/s. Significant aeroacoustic emissions were noted from the turbine in the 4.5 m/s and 5.5 m/s cases, with an increase of up to 12 dB over background levels. Source maps from the 5.5 m/s tests revealed that the primary location of aeroacoustic emissions was near the outer radii of the rotor, but not at the tip, and generally moved radially outward with increasing frequency. The azimuthal location of the greatest sound pressure levels was typically found to be between 120º and 130º measured counterclockwise from the upward vertical, coinciding with the predicted location of greatest emissions provided by an analytical model based on dipole directivity and convective amplification. Analysis of the acoustic spectra, turbine operating characteristics, and previous literature suggested that the sound emissions emanated from the trailing edge of the blades.

Recent years have brought many new developments in the processing of speech and acoustic signals. Yet, despite this, the process of acquiring signals has gone largely unchanged. Adding spatial diversity to the repertoire of signal acquisition has long been known to offer advantages for processing signals further. The processing capabilities of mobile devices had not previously been able to handle the required computation to handle these previous streams of information. But current processing capabilities are such that the extra workload introduced by the addition of mutiple sensors on a mobile device are not over-burdensome. How these extra data streams can best be handled is still an open question. The present work deals with the examination of one type of spatial processing technique, known as beamforming. A microphone array test platform is constructed and verified through a number of beamforming agorithms. Issues related to speech acquisition through microphones arrays are discussed. The algorithms used for verification are presented in detail and compared to one another.

Aeroacoustic emissions were identified as a primary concern in the public acceptance of wind turbines. A review of literature involving sound localization was undertaken and led to the design of two microphone arrays to identify acoustic sources. A small-scale array composed of 27 sensors was produced with the intention of improving the quality of sound measurements over those made by a single microphone in a small, closed-loop wind tunnel. A large-scale array containing 30 microphones was also implemented to allow for measurements of aeroacoustic emissions from airfoils and rotating wind turbines. To minimize cost and pursue alternative sensor technologies, microelectromechanical microphones were selected for the array sensors and assembled into the arrays on printed circuit boards. Characterization of the microphones was completed using a combination of calibration techniques, primarily in a plane wave tube. Array response to known sources was quantified by analyzing source maps with respect to source location accuracy, beamwidth, and root mean square error. Multiple sources and rotating sources were tested to assess array performance. Following validation with known sources, wind tunnel testing of a 600 watt wind turbine was performed at freestream speeds of 2.5 m/s, 3.5 m/s, 4.5 m/s, and to 5.5 m/s. Significant aeroacoustic emissions were noted from the turbine in the 4.5 m/s and 5.5 m/s cases, with an increase of up to 12 dB over background levels. Source maps from the 5.5 m/s tests revealed that the primary location of aeroacoustic emissions was near the outer radii of the rotor, but not at the tip, and generally moved radially outward with increasing frequency. The azimuthal location of the greatest sound pressure levels was typically found to be between 120º and 130º measured counterclockwise from the upward vertical, coinciding with the predicted location of greatest emissions provided by an analytical model based on dipole directivity and convective amplification. Analysis of the acoustic spectra, turbine operating characteristics, and previous literature suggested that the sound emissions emanated from the trailing edge of the blades.

Thesis (M.S.)--University of Alabama at Birmingham, 2006. / Description based on contents viewed Jan. 29, 2007; title from title screen. Includes bibliographical references.

Automatic speech recognition has become a standard feature on many consumer electronics and automotive products, and the accuracy of the decoded speech has improved dramatically over time. Often, designers of these products achieve accuracy by employing microphone arrays and beamforming algorithms to reduce interference. However, beamforming microphone arrays are too large for small form factor products such as smart watches. Yet these small form factor products, which have precious little space for tactile user input (i.e. knobs, buttons and touch screens), would benefit immensely from a user interface based on reliably accurate automatic speech recognition. This thesis proposes a solution for interference mitigation that employs blind source separation with a compact array of commercially available unidirectional microphone elements. Such an array provides adequate spatial diversity to enable blind source separation and would easily fit in a smart watch or similar small form factor product. The solution is characterized using publicly available speech audio clips recorded for the purpose of testing automatic speech recognition algorithms. The proposal is modelled in different interference environments and the efficacy of the solution is evaluated. Factors affecting the performance of the solution are identified and their influence quantified. An expectation is presented for the quality of separation as well as the resulting improvement in word error rate that can be achieved from decoding the separated speech estimate versus the mixture obtained from a single unidirectional microphone element. Finally, directions for future work are proposed, which have the potential to improve the performance of the solution thereby making it a commercially viable product.

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.

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.

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.

Many animal species, including many species of bats, exhibit collective behavior where groups of individuals coordinate their motion. Most bats are unique among these animals in that they use the active sensing mechanism of echolocation as their primary means of navigation. Due to their use of echolocation in large groups, bats run the risk of signal interference from sonar jamming. However, several species of bats have developed various strategies to prevent interference which may lead to different behavior when flying with conspecifics than when flying alone. This thesis seeks to explore the role of this sensing on the behavior of bat pairs flying together. Field data from a maternity colony of gray bats (Myotis grisescens) were collected using an array of cameras and microphones. These data were analyzed using the information theoretic measure of transfer entropy in order to quantify the interaction between pairs of bats and to determine the effect echolocation calls have on this interaction. Results show that there is evidence of information transfer in both the speed of the bats and their turning behavior, and that such evidence is absent when we consider their heading directions. Unidirectional information transfer was found in some subsets of the data which could be evidence of a leader-follower interaction. / Master of Science / Manyanimalspeciesexhibitcollectivebehaviorwheregroupsofanimalscoordinatetheir motion, as in flocking or schooling. Many species of bats also demonstrate this behavior. Bats are unique among these animals in that they use echolocation as their primary means of navigation. Bats produce ultrasonic pulses or calls and listen to the returning echo to "visualize" their environment. Bats using echolocation in large groups run the risk of other bat calls interfering with their ability to hear their own calls. They have developed various waystopreventinterferencewhichmayleadtodifferentbehaviorwhenflyingwithotherbats thanwhenflyingalone. Fielddatafromamaternitycolonyofgraybatswerecollectedusing a system of cameras and microphones. These data were analyzed to quantify the interaction between pairs of bats and to determine the effect echolocation calls have on this interaction. Results show that there is evidence of information transfer about both the speed of the bats and their turning behavior. There was also evidence of a possible leader-follower interaction in some subsets of the data.