Thesis (Ph. D.)--State University of New York at Binghamton, Thomas J. Watson School of Engineering and Applied Science, Department of Mechanical Engineering, 2008. / Includes bibliographical references.

The thesis focuses on fast acquisition techniques for spread spectrum packet radio communications systems. Matched filters are often used to achieve fast acquisitions. A new synchronizer using multiple acquisition detection is designed to achieve a highly reliable synchronization with a very simple receiver structure. Since PN codes, in practice, cannot be made too long due to the difficulty of manufacturing long matched filters and the limitation on the bandwidth of the frequency spectrum for the system, the reliable synchronization can be only obtained by repeating the transmission of the acquisition code at the beginning of each packet. The verification or coincidence detection is done by means of a marker detection following an acquisition. A hard-limiting synchronizer is also examined combined with the multiple acquisition detection. The hard-limiting synchronizer is simpler to implement and suitable for receiving signals with a large SNR dynamic range, but it cannot work well when multi-user interference and multi-path interference are present. For this reason, a new linear Automatic Threshold Control (ATC) synchronizer is developed for detecting signals with a large amplitude dynamic range while preserving good performance in multi-path and multi-user interference. The idea of the ATC scheme is to adjust the receiver acquisition threshold level according to the SNR of the received signal such that the largest (or the most likely) correlation peak in a short time period is selected for the synchronization alignment. Therefore false acquisitions caused by strong correlation side-lobes during the acquisition can be eliminated. For the more realistic situation where the multi-user interference or near-far effect causes severe performance degradation, we proposed a novel non-linear multi-user detector or multistage detector which is suitable for both the synchronous and asynchronous CDMA systems. This sub-optimal detector is able to achieve the performance of the optimal detector with very small computation complexity. The near-far effect will no longer exist because the interference from the unexpected users is considered to be not always harmful for the detection of a specific users' message. To apply this detection technique to asynchronous CDMA systems, acquisition for each users' PN codes becomes more critical, because during the acquisition, the information from the other users' PN code is usually not available, which means that acquisition still suffers the near-far effect. The proposed acquisition scheme based on interference cancellation technique and the ATC scheme can alleviate the near-far effect significantly, and provide the necessary condition for the appropriate operations of multi-user detectors. / Graduate

Turbulent flow over a rough surface produces sound that radiates outside the near wall region. This noise source is often at a lower level than the noise created by edges and bluff body flows, but for applications with large surface area to perimeter ratios at low Mach number, this noise source can have considerable levels. In the first part of this dissertation, a detailed study is made of the ability of the Glegg & Devenport (2009) scattering theory to predict roughness noise. To this end, comparisons are made with measurements from cuboidal and hemispherical roughness with roughness Reynolds numbers, hu_Ï /ν, ranging from 24 to 197 and roughness height to boundary layer thickness ratios of 5 to 18. Their theory is shown to work very accurately to predict the noise from surfaces with large roughness Reynolds numbers, but for cases with highly inhomogeneous wall pressure fields, differences grow between estimation and measurement. For these surfaces, the absolute levels were underpredicted but the spectral shape of the measurement was correctly determined indicating that the relationship of the radiated noise with the wavenumber wall pressure spectrum and roughness geometry appears to remain relatively unchanged. In the second part of this dissertation, delay and sum beamforming and least-squares analyses were used to examine roughness noise recorded by a 36-sensor linear microphone array. These methods were employed to estimate the variation of source strengths through short fetches of large hemispherical and cuboidal element roughness. The analyses show that the lead rows of the fetches produced the greatest streamwise and spanwise noise radiation. The least-squares analysis confirmed the presence of streamwise and spanwise aligned dipoles emanating from each roughness element as suggested by the LES of Yang & Wang (2011). The least-squares calculated source strengths show that the streamwise aligned dipole is always stronger than that of the spanwise dipole, but the relative magnitude of the difference varies with frequency. / Ph. D.

In this thesis, a computationally efficient method for detecting a whistle and capturing it using a 4 microphone array is proposed. Furthermore, methods are developed to efficiently process the data captured from all the microphones to estimate the direction of the sound source. The accuracy, the shortcoming and the constraints of the method proposed are also discussed. There is an emphasis placed on being computationally efficient so that the methods may be implemented on a low cost microcontroller and be used to provide a heading to an Unmanned Ground Vehicle. / MS / As humans, we rely on our sense of hearing to help us interact with the outside world. It helps us to listen not just to other people but also for sounds that maybe a warning for us. It can often be the first warning we get of an impending danger as we might hear a predator before we see it or we might hear a car brake and slip before we turn to look at it. However, it is not merely the ability to hear a sound that makes hearing so useful. It is the fact that we can tell which direction the sound is coming from that makes it so important. That is what allows us to know which direction to turn towards to respond to someone or from which direction the sound warning us of danger is coming. We may not be able to pinpoint the location of the source with complete accuracy but we can discern the general heading. It was this idea that inspired this research work. We wanted to be capable of estimating where a sound is coming from while being computationally efficient so that it may be implemented in real time with the help of a low cost microcontroller. This would then be used to provide a heading to an Unmanned Ground Vehicle while keeping the costs down.

The concept of replacing threads with flexible wires and sensors in a fabric to provide an underlying platform for integrating electronic components is known as e-textiles. This concept can be used to design applications involving different types of electronic components including sensors, digital signal processors, microcontrollers, color-changing fibers, and power sources. The adaptability of the textiles to the needs of the individual and the functionality of electronics can be integrated to provide unobtrusive, robust, and inexpensive clothing with novel features. This thesis focuses on the design of e-textiles for acoustic signal processing applications. This research examines challenges encountered when developing e-textile applications involving distributed arrays of microphones. A framework for designing such applications is presented. The design process and the performance analysis of two e-textiles, a large-scale beamforming fabric and a speech-processing vest, are presented. / Master of Science

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In recent years, the popularity of 360-degree video paired with 1st order ambisonic audio has seen a rise on different social media platforms online. Due to this increase in popularity, many new ambisonic microphones have been developed and are now available on the market. However, most of the research into this field has almost exclusively been in the form of case studies where microphone manufacturers showcase practical applications of their equipment, and no real comparisons between ambisonic recording methods have been made. This study aims to fill that gap by conducting a listening test that compares four common methods of recording ambisonic audio and to evaluate listeners’ preferences regarding spatial attributes. Due to a relatively small sample size of 15 no definitive conclusions can be made, but the study did find a clear preference towards a combined method of an ambisonic microphone paired with spot microphones.

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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.

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