Speech communication is gaining in popularity in many different contexts as technology evolves. With the introduction of mobile electronic devices such as cell phones and laptops, and fixed electronic devices such as video and teleconferencing systems, more people are communicating which leads to an increasing demand for new services and better speech quality. Methods to enhance speech recorded by microphones often operate blindly without prior knowledge of the signals. With the addition of multiple microphones to allow for spatial filtering, many blind speech enhancement methods have to operate blindly also in the spatial domain. When attempting to improve the quality of spoken communication it is often necessary to be able to reliably determine the location of the speakers. A dedicated source localization method on top of the speech enhancement methods can assist the speech enhancement method by providing the spatial information about the sources. This thesis addresses the problem of speech-source localization, with a focus on the problem of localization in the presence of multiple concurrent speech sources. The primary work consists of methods to estimate the direction of arrival of multiple concurrent speech sources from an array of sensors and a method to correct the ambiguities when estimating the spatial locations of multiple speech sources from multiple arrays of sensors. The thesis also improves the well-known SRP-based methods with higher-order statistics, and presents an analysis of how the SRP-PHAT performs when the sensor array geometry is not fully calibrated. The thesis is concluded by two envelope-domain-based methods for tonal pattern detection and tonal disturbance detection and cancelation which can be useful to further increase the usability of the proposed localization methods. The main contribution of the thesis is a complete methodology to spatially locate multiple speech sources in enclosed environments. New methods and improvements to the combined solution are presented for the direction-of-arrival estimation, the location estimation and the location ambiguity correction, as well as a sensor array calibration sensitivity analysis.

"This thesis develops a system for synchronizing two wireless transmitters so that they are able to implement a distributed beamformer in several different channel models. This thesis considers a specific implementation of the system and proposes a metric to quantify its performance. The system's performance is investigated in single-path and multi-path time-invariant channel scenarios, as well as in single-path time-varying channel scenarios. Where prior systems have difficulty in implementing a distributed beamformer in multi-path channels and/or mobile scenarios, the results of this thesis show that the Round-Trip Time-Division distributed beamforming system is able to perform as a beamformer in all three of the channel models considered. "

Ingeniero Civil Eléctrico / La utilización de vehículos eléctricos es una forma efectiva de contrarrestar el impacto negativo producido por la humanidad en el medio ambiente del planeta, a través de la reducción de gases nocivos emitidos a la atmósfera. Sin embargo, este tipo de automóviles aún no alcanza el suficiente desarrollo, lo que hace necesario efectuar mejoras para acelerar su industrialización. Un avance tecnológico relevante es la posibilidad de alimentar el vehículo mientras está en movimiento, lo que permitiría eliminar los largos tiempos de carga de las baterías. El presente informe tiene como objetivo estudiar un sistema que permita transferir potencia a un automóvil en movimiento, utilizando la tecnología de direccionamiento de haces. Para esto, es necesario investigar previamente los conceptos de transferencia inalámbrica de energía, beamforming, diseño de antenas e interferencia de ondas. Para estudiar el sistema se crean simulaciones y pruebas de concepto. Se inicia observando la interferencia de las ondas producidas por un arreglo lineal de parlantes a través de una simulación animada. Con el objetivo de definir características del modelo, se analizan los patrones de radiación producidos por un arreglo al variar los parámetros fundamentales, además de analizar el comportamiento del mismo en distintas bandas de frecuencia. Finalmente, se realizan ensayos con sonido para visualizar el comportamiento de beamforming en un escenario con cuatro parlantes. La implementación de la simulación de interferencia consigue visualizar el comportamiento de las ondas en el tiempo y el espacio, conformando una primera aproximación al sistema. Por otro lado, la variación de los parámetros del arreglo permite definir el número de antenas, la distancia entre las mismas y la frecuencia de trabajo que permiten una mayor directividad del modelo. Finalmente, las pruebas de concepto con sonido conforman una aplicación válida de beamforming, logrando direccionar las ondas en un ángulo en particular. El principal alcance del trabajo realizado es su función como puente en la creación de trabajos futuros. Por un lado, el sistema de transferencia de energía a un vehículo puede ser utilizado como base para una aplicación real, a través del diseño de las antenas y creación de prototipos. Por otro, las pruebas de concepto posibilitan la implementación estudios que busquen probar la validez de conceptos utilizados en telecomunicaciones, aplicadas a un grupo de parlantes como sustituto de un arreglo de antenas.

In this study, we consider a relay network, with two transceivers and r relay nodes. We assume that each of relays and the two transceivers have a single antenna. For establishing the connection between these two transceivers, we propose a two-way relaying scheme which takes three phases (time slots) to accomplish the exchange of two information symbols between the two transceivers. In the first and second phases, the transceivers, transmit their signals, toward the relays, one after other. The signals that are received by relays are noisy versions of the original signals. Each relay, multiplies its received signal by a complex beamforming coefficient to adjust the phase and amplitude of the signal. Then in the third phase, each relay transmits the summation of so-obtained signals to both transceivers. Our goal is to find the optimal values of transceivers’ transmit powers and the optimal values of the beamforming coefficients by minimizing the total transmit power subject to quality of service constraints. In our approach, we minimize the total transmit power under two constraints. These two constraints are used to guarantee that the transceivers’ receive Signal-to-Noise Ratios (SNRs) are above given thresholds. To solve the underlying optimization problem, we develop two techniques. The first technique is a combination of a two-dimensional search and Second-Order Convex Cone Programming (SOCP). More specifically, the set of feasible values of transceivers’ transmit powers is quantized into a sufficient fine grid. Then, at each vertice of this grid, an SOCP problem is solved to obtain the beamforming coefficients such that for the given pair of transceivers’ transmit powers, the total transmit power is minimized. The pair of the transceivers’ transmit powers, which result in the smallest possible value of the total transmit power, leads us to the solution of the problem. This approach requires a two-dimensional search and solving an SOCP problem at each point of the corresponding two-dimensional grid. Thus, it can be prohibitively expensive in terms of computational complexity. As a second method, we resort to a gradient based steepest descent technique. Our simulation results show that this second technique performs very close to the optimal two-dimensional search based algorithm. Finally we compare our technique with multi-relay distributed beamforming schemes, previously developed in literature and show that our three-phase two-way relaying scheme requires less total power as compared to the two-phase two-way relaying method. On the other hand, the two-phase two-way relaying achieves higher data rates when compared with three-phase two-way relaying for the same total transmit power. Also, we observe that the three-phase scheme has more degrees of freedom while multi-relay distributed beamforming schemes, previously developed in literature appears to be more bandwidth efficient. / UOIT

This research examines a two-way relay network consisting of two transceivers and multiple parallel relays, which are equipped with single antennas and operate in a halfduplex mode. In this system, the two transceivers prefer to exchange their information via relays. It is assumed that the relays have the full instantaneous channel state information (CSI) and relay the signals using the amplify-and-forward (AF) method. The performance of two AF bi-directional network beamforming schemes, namely multiple access broadcast channel (MABC) strategy and time division broadcast channel (TDBC) protocol, under joint optimal power control and beamforming design are studied and compared. To do so, we first design a TDBC-based bi-directional network beamformers, through minimization of the total power consumed in the whole network subject to quality of service (QoS) constraints, for the case with a direct link between the two transceivers. The corresponding power minimization problem is carried out over the transceiver transmit powers as well as relay beamforming weights, thus resulting in a jointly optimal power allocation and beamforming approach. We devise optimal secondorder cone programming based solutions as well as fast gradient-based solutions to these problems. Then these solutions are exploited to compare the performance of the underlying TDBC-based approach to that of the MABC-based technique developed in [1]. This comparison is important because the TDBC approach appears to have certain advantages which can be exploited towards improving the performance of two-way network beamforming. These advantages include the additional degree of freedom as well as the possibility of benefitting from the availability of a direct link between the two transceivers. Interestingly, in the absence of a direct link between the two transceivers, we show that when the QoS constraints are imposed to meet certain given probabilities of un-coded error (or, equivalently, to meet certain signal-to-noise ratio constraints), these two schemes perform closely in terms of the minimum total transmit power. However, when the QoS iv constraints are used to guarantee certain given rates, the MABC-based scheme outperforms the TDBC counterpart. In the case when a direct link exists between the two transceivers, the TDBC-based approach can outperform the MABC-based method provided that the direct link is strong enough. / UOIT

This thesis proposes a distributed beamforming technique in wireless networks with half-duplex amplify-and-forward relays. With full channel state information, it has been shown that transmit beamforming is able to achieve significant diversity and coding gain. However, it takes large amount of overhead. First, we adopt the Generalized Lloyd Algorithm to design codebooks which minimize average SNR, and reduce the feedback rate by quantizing the channel state information. Furthermore, we utilize the correlation property of time-varying channels to compress the size of feedback message required to accomplish distributed beamforming. We model time-varying channels as a first-order finite-state Markov chain, namely the emph{channel state Markov chain}. Then, we propose two methods to compress the feedback bits according to the property of the transition probabilities among different channel states. One method is to compress the feedback by discarding some channel states which is less likely to be transited given current state. In the other method, we reserve all channel states and adopt Huffman coding to compress the feedback bits based on the transition probabilities. Simulations also show that distributed beamforming with compressed feedback performs closely to the beamforming with infinite feedback.

Estimating S-wave velocities (Vs) from Rayleigh-wave velocities (VR) is widely used in field seismic testing for geotechnical engineering purposes. In this research, two widely used surface-wave methods, the Spectral-Analysis-of-Surface-Waves (SASW) and Multichannel-Analysis-of-Surface-Waves (MASW) methods, are evaluated and compared in field experiments. An experimental parametric study was undertaken of the SASW and MASW methods. Conventional seismic sources in the SASW method are sledge hammers, bulldozers and vibroseises. For MASW testing, sledge hammers and small shakers are usually used as the seismic sources. In this research, MASW testing was performed with traditional and non-traditional sources at a site owned by the City of Austin, Texas. Experimental dispersion curves and Vs profiles from SASW tests are used as references for the field parametric study with the MASW method. The source type, source offset, receiver spacing and number of receivers were varied to evaluate the impact of each variable on the field experimental dispersion curve. Two type of receivers, 1-Hz and 4.5-Hz natural-frequency geophones, were also compared in these tests. A second part of this research involved studying the use of characterizing geotechnical materials based on Vs. This work included two projects. The first project involved basalt on the Big Island of Hawaii. To develop empirical ground motion prediction models for the purpose of earthquake hazard mitigation and seismic design on the Big Island, the subsurface site conditions beneath 22 strong motion stations were investigated by SASW tests. Vs profiling was performed to depths of more than 100 ft. Vs30, the average Vs in the top 30 m, was also calculated to assign NEHRP site classes to different testing locations. Different materials, mainly thought to be stiff basalt, were characterized and grouped based on the Vs values. These groups were then compared with reference curves for sand and gravel (Menq, 2003) to differentiate the groups. The second project dealing with charactering geotechnical materials based on Vs involved of soil/rock profiles at a project site in British Columbia, Canada. The goals in terms of this research were to: (1) compare the Vs profiles from the different test locations to investigate the stiffnesses of different geologic materials, the variability in the material stiffnesses, and the estimated depth to bedrock, and (2) to compare the Vs profiles to existing geological and geotechnical information such as nearby boreholes, cone penetration tests and seismic cone penetration tests. Good agreement between SASW Vs profiles and boring records is expected when lateral variability at the site is low. However, when lateral variability is significant, then the difference between localized measurements, like borings and CPT results, and global measurements, like SASW Vs results, can further contribute to understanding the site conditions as shown at the site in British Columbia, Canada. / text

In many audio applications, the signal of interest is corrupted by acoustic background noise, interference, and reverberation. The presence of these contaminations can significantly degrade the quality and intelligibility of the audio signal. This makes it important to develop signal processing methods that can separate the competing sources and extract a source of interest. The estimated signals may then be either directly listened to, transmitted, or further processed, giving rise to a wide range of applications such as hearing aids, noise-cancelling headphones, human-computer interaction, surveillance, and hands-free telephony. Many of the existing approaches to speech separation/extraction relied on beamforming techniques. These techniques approach the problem from a spatial point of view; a microphone array is used to form a spatial filter which can extract a signal from a specific direction and reduce the contamination of signals from other directions. However, when there are fewer microphones than sources (the underdetermined case), perfect attenuation of all interferers becomes impossible and only partial interference attenuation is possible. In this thesis, we present a framework which extends the use of beamforming techniques to underdetermined speech mixtures. We describe frequency domain non-linear mixture of beamformers that can extract a speech source from a known direction. Our approach models the data in each frequency bin via Gaussian mixture distributions, which can be learned using the expectation maximization algorithm. The model learning is performed using the observed mixture signals only, and no prior training is required. The signal estimator comprises of a set of minimum mean square error (MMSE), minimum variance distortionless response (MVDR), or minimum power distortionless response (MPDR) beamformers. In order to estimate the signal, all beamformers are concurrently applied to the observed signal, and the weighted sum of the beamformers’ outputs is used as the signal estimator, where the weights are the estimated posterior probabilities of the Gaussian mixture states. These weights are specific to each timefrequency point. The resulting non-linear beamformers do not need to know or estimate the number of sources, and can be applied to microphone arrays with two or more microphones with arbitrary array configuration. We test and evaluate the described methods on underdetermined speech mixtures. Experimental results for the non-linear beamformers in underdetermined mixtures with room reverberation confirm their capability to successfully extract speech sources.

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Wireless communication systems, aside from path-loss, also suffer from small scale up-and- down variations in the power of the received signal. These fluctuations in the received signal power, commonly referred to as multi-path fading, result in a significant perfor- mance degradation of the system. One way to combat fading is diversity. The idea behind diversity is to provide the receiver with multiple independent copies of the transmitted signal, either in time, frequency or space dimension. In broadcast networks with underlying slow-faded channels, an appropriate option for exploiting diversity is transmit diversity, which deploys several antennas in the transmitter terminal. Based on the amount of available channel state information on the transmitter side, various transmit diversity schemes have been proposed in the literature. Because of certain limitations of broadcast networks, a practical assumption in these networks is to provide no channel state information for the transmitter. In this dissertation, a new scheme is proposed to exploit transmit diversity for broad- cast networks, assuming no channel state information in the transmitter. The main idea of our proposed method is to virtually impose time variations to the underlying slow-faded channels by multiplying quantized pseudo-random phases to data symbols before trans- mission. Using this method, all necessary signal processing can be transferred to the RF front-end of the transmitter, and therefore, the implementation cost is much less than that of alternative approaches. Under the proposed method, the outage probability of the system is analyzed and the corresponding achievable diversity order is calculated. Simulation results show that the performance of our proposed scheme falls slightly below that of the optimum (Alamouti type) approach in the low outage probability region.