Thesis (M. Sc.)--University of Hong Kong, 2000. / Includes bibliographical references (leaves 40-45).

Thesis (M.Phil.)--Hong Kong University of Science and Technology, 2009. / Includes bibliographical references.

Thesis (Ph. D.)--State University of New York at Binghamton, Department of Mechanical Engineering, 2007. / Includes bibliographical references (leaves 177-179).

This thesis considers the domain of acoustic source localisation and tracking in an indoor environment. Acoustic tracking has applications in security, human-computer interaction, and the diarisation of meetings. Source localisation and tracking is typically a computationally expensive task, making it hard to process on-line, especially as the number of speakers to track increases. Much of the literature considers single-source localisation, however a practical system must be able to cope with multiple speakers, possibly active simultaneously, without knowing beforehand how many speakers are present. Techniques are explored for reducing the computational requirements of an acoustic localisation system. Techniques to localise and track multiple active sources are also explored, and developed to be more computationally efficient than the current state of the art algorithms, whilst being able to track more speakers. The first contribution is the modification of a recent single-speaker source localisation technique, which improves the localisation speed. This is achieved by formalising the implicit assumption by the modified algorithm that speaker height is uniformly distributed on the vertical axis. Estimating height information effectively reduces the search space where speakers have previously been detected, but who may have moved over the horizontal-plane, and are unlikely to have significantly changed height. This is developed to allow multiple non-simultaneously active sources to be located. This is applicable when the system is given information from a secondary source such as a set of cameras allowing the efficient identification of active speakers rather than just the locations of people in the environment. The next contribution of the thesis is the application of a particle swarm technique to significantly further decrease the computational cost of localising a single source in an indoor environment, compared the state of the art. Several variants of the particle swarm technique are explored, including novel variants designed specifically for localising acoustic sources. Each method is characterised in terms of its computational complexity as well as the average localisation error. The techniques’ responses to acoustic noise are also considered, and they are found to be robust. A further contribution is made by using multi-optima swarm techniques to localise multiple simultaneously active sources. This makes use of techniques which extend the single-source particle swarm techniques to finding multiple optima of the acoustic objective function. Several techniques are investigated and their performance in terms of localisation accuracy and computational complexity is characterised. Consideration is also given to how these metrics change when an increasing number of active speakers are to be localised. Finally, the application of the multi-optima localisation methods as an input to a multi-target tracking system is presented. Tracking multiple speakers is a more complex task than tracking single acoustic source, as observations of audio activity must be associated in some way with distinct speakers. The tracker used is known to be a relatively efficient technique, and the nature of the multi-optima output format is modified to allow the application of this technique to the task of speaker tracking.

Microphone windscreens are typically used to reduce the noise associated with wind flowing over a microphone diaphragm by reducing the velocity of the airflow. While most windscreens are effective at reducing this noise, they do not protect the microphone from many natural elements, such as moisture, sand, and other small particles. The focus of this research was to design a windscreen that protects an array of five microphones located around a 4.5-inch diameter cylinder from these natural elements. The design goals were to have a wind noise attenuation of at least 8 dB, an insertion loss of less than 1 dB from 5-1000 Hz, and a phase shift error of less than 3% over the same range. Computer simulations and experimental testing were used to select two basic designs. Four experimental tests consisting of wind noise attenuation, sand entrapment, insertion loss, and phase change measurements were used to optimize the geometry of these designs. The wind noise attenuation was tested by spinning the microphone array on a long boom and by setting the array in front of a fan. Sand was blown at the windscreen in order to test how well the windscreen protects the microphone array from small particles in the velocity stream. The insertion loss of the windscreen was tested by comparing an incoming signal traveling through the windscreen to the same signal without the windscreen. Finally, the phase shift between microphones was measured using a single frequency and comparing the microphone measurements with and without the windscreen. These four tests were performed on two designs. The first design consists of two foam filled concentric cones set around the microphone array. The second design consists of tubes that project outward from each microphone diaphragm, and then curve downwards. Both final windscreen designs meet the desired requirements. They both reduce wind noise attenuation by approximately 9 dB in a 13 mph wind and over 16 dB in a 20 mph wind. They also have negligible insertion loss, have a phase shift error of less than 3%, and are very efficient at blocking particles from entering the windscreen.

Thesis (MPhil (Music))--University of Stellenbosch, 2011. / ENGLISH ABSTRACT: This thesis investigates the role of natural or realistic audio recordings in popular music in the context of the different nature of popular music where the goal is not necessarily the recreation of the original acoustic space. Traditional microphone techniques are investigated from the perspective of the identifiable characteristics of popular music to establish the role of microphone techniques to obtain a desired outcome. / AFRIKAANSE OPSOMMING: Hierdie tesis ondersoek die rol van natuurlike of realistiese klankopnames in populêre musiek in die lig daarvan dat die aard van populêre opnames verskil van reproduksie wat poog om die opnameruimte akoesties te herskep. Tradisionele mikrofoontegnieke word bestudeer vanuit die hoek van die identifiseerbare eienskappe van populêre musiek om te bepaal watter rol mikrofoontegniek kan speel om ’n verlangde uitkoms te bewerkstellig.

The last few years have seen the start of a unique change in microphones for consumer devices such as smartphones or tablets. Almost all analogue capacitive microphones are being replaced by digital silicon microphones or MEMS microphones. MEMS microphones perform differently to conventional analogue microphones. Their greatest disadvantage is significantly increased self-noise or decreased SNR, while their most significant benefits are ease of design and manufacturing and improved sensitivity matching. This thesis presents research on speech processing, comparing conventional analogue microphones with the newly available digital MEMS microphones. Specifically, voice activity detection, speaker diarisation (who spoke when), speech separation and speech recognition are looked at in detail. In order to carry out this research different microphone arrays were built using digital MEMS microphones and corpora were recorded to test existing algorithms and devise new ones. Some corpora that were created for the purpose of this research will be released to the public in 2013. It was found that the most commonly used VAD algorithm in current state-of-theart diarisation systems is not the best-performing one, i.e. MLP-based voice activity detection consistently outperforms the more frequently used GMM-HMM-based VAD schemes. In addition, an algorithm was derived that can determine the number of active speakers in a meeting recording given audio data from a microphone array of known geometry, leading to improved diarisation results. Finally, speech separation experiments were carried out using different post-filtering algorithms, matching or exceeding current state-of-the art results. The performance of the algorithms and methods presented in this thesis was verified by comparing their output using speech recognition tools and simple MLLR adaptation and the results are presented as word error rates, an easily comprehensible scale. To summarise, using speech recognition and speech separation experiments, this thesis demonstrates that the significantly reduced SNR of the MEMS microphone can be compensated for with well established adaptation techniques such as MLLR. MEMS microphones do not affect voice activity detection and speaker diarisation performance.

No description available.

Directional microphones, which suppress noise coming from unwanted directions while preserving sound signals arriving from a desired direction, are essential to hearing aid technology. The device presented in this paper abandons the principles of standard pressure sensor microphones, dual port microphones, and multi-chip array systems and instead employs a new method of operation. The proposed device uses a lightweight silicon micromachined structure that becomes “entrained” in the oscillatory motion of air vibrations, and thus maintains the vector component of the air velocity. The mechanical structures are made as compliant as possible so that the motion of the diaphragm directly replicates the motion of the sound wave as it travels through air. The microphone discussed in this paper achieves the bi-directionality seen in a ribbon microphone but is built using standard semiconductor fabrication techniques and utilizes piezoelectric readout of a circular diaphragm suspended on compliant silicon springs. Finite element analysis and lumped element modeling have been performed to aid in structural design and device verification. The proposed microphone was successfully fabricated in a cleanroom facility at The University of Texas at Austin. Testing procedures verified that the resonant frequency of the microphone, as expected, was much lower than in traditional microphones. This report discusses the theory, modeling, fabrication and testing of the microphone. / text

Thesis (M.S.)--State University of New York at Binghamton, Dept. of Mechanical Engineering, 2007. / Includes bibliographical references.