Sound source localization with microphone arrays has received considerable attention as a means for the automated tracking of individuals in an enclosed space and as a necessary component of any general-purpose speech capture and automated camera pointing system. A novel computationally efficient method compared to traditional source localization techniques is proposed and is both theoretically and experimentally investigated in this research. This thesis first reviews the previous work in this area. The evolution of a new localization algorithm accompanied by an array structure for audio signal localization in three dimensional space is then presented. This method, which has similarities to the structure of the eye, consists of a novel hemispherical microphone array with microphones on the shell and one microphone in the center of the sphere. The hemispherical array provides such benefits as 3D coverage, simple signal processing and low computational complexity. The signal processing scheme utilizes parallel computation of a special and novel closeness function for each microphone direction on the shell. The closeness functions have output values that are linearly proportional to the spatial angular difference between the sound source direction and each of the shell microphone directions. Finally by choosing directions corresponding to the highest closeness function values and implementing linear weighted spatial averaging in those directions we estimate the sound source direction. The experimental tests validate the method with less than 3.10 of error in a small office room. Contrary to traditional algorithmic sound source localization techniques, the proposed method is based on parallel mathematical calculations in the time domain. Consequently, it can be easily implemented on a custom designed integrated circuit.

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Modern satellite, wireless communication, and radar systems often demand wideband performance for multi-channel operation and the ability to steer multiple beams for multiple moving targets. This dissertation covers a variety of topics to design low-cost and wideband antenna systems. The main areas of study are microwave devices controlled piezoelectric transducers (PETs) and wideband baluns and balanced microwave circuits using parallel-strip lines. Some focus has also been given to the design of Rotman lens for multiple beam generation and Vivaldi antenna arrays for wideband two-dimensional scanning. The dielectric perturbation technique controlled by PET is introduced to design a wideband phase shifter and a QPSK modulator, and to tune the resonant frequency of a slot dipole. The designed PET-controlled phase shifters are used for beam steering in a dual beam phased array using a bidirectional feeding scheme and a five-beam phased array using a microstrip Rotman lens. Vivaldi-type antennas are commonly used to achieve wideband performance. Very wideband performance can be achieved using an antipodal tapered slot antenna because of its inherent simple wideband transition from microstrip line to parallel-strip line. An antipodal tapered slot antenna and a phased array are designed to span 10 to 35 GHz. In addition, a 4??4 two-dimensional antenna array is designed using wideband antipodal tapered slot antennas, and two sets of PET-controlled phase shifters for E- and H-plane scanning are fabricated to steer the beam. As a microwave system using wideband antenna array, a new low-cost and wideband phased array radar is developed using a modulated pulse over 8 to 20 GHz band. The double-sided parallel-strip line as a balanced line is presented. The parallelstrip line offers much flexibility for microwave circuit designs. This transmission line makes it possible to realize a low impedance line and allows the design of a compact wideband balun and junction. Wideband transitions (or baluns) from parallel-strip line to microstrip line, a typical unbalanced transmission line, are realized to cover several octave bandwidth. Balanced microwave filters and a hybrid coupler are developed using the parallel-strip line.

Thesis (Ph. D.)--University of Washington, 2001. / Vita. Includes bibliographical references (leaves 174-175).

Thesis (M.S. in Electrical Engineering)--Naval Postgraduate School, March 2003. / Thesis advisor(s): Herschel H. Loomis, Alan A. Ross. Includes bibliographical references (p. 117). Also available online.

Due to their high potential performance and reduced energy and power consumption, field-programmable gate arrays (FPGAs) are widely used as accelerators for today’s computationally intensive applications. These applications use advanced algorithms more sophisticated than ever before. The high design complexity along with fast development process challenges the traditional FPGA design methodology using hardware description languages. High-level synthesis accelerates design implementation by raising the level of design abstraction beyond register transfer level. This dissertation work develops a highly energy-efficient FPGA high-level synthesis tool, ArchSyn, using an application-specific coarse-grain architecture as an intermediate synthesis step. ArchSyn provides rapid and energy-efficient compilation of dataflow graphs (DFGs) on FPGAs by scheduling the dataflow operations on an array of directly connected simple configurable processing elements (CPEs). Each CPE in the array performs primitive compute operations according to a small local sequencer at each cycle. Data are communicated via multi-hop routing within the direct interconnect network. The scheduler schedules each compute operation of the DFG obtained from the high-level design to execute on a particular hardware CPE at a particular cycle. It also determines the communication schedule of the intermediate data among the producing and consuming CPEs, optionally buffering them with distributed memory along the path. As such, the lengthy process of synthesizing a full custom hardware design on FPGA is reduced to a scheduling and mapping process. By restricting the fine-grain programmability into a coarse grain processor network scheduling problem, the compilation time can be improved substantially, thereby improving the overall productivity of the designer. Furthermore, taking advantage of the programmability of FPGAs, the effect of the array interconnect architecture on the energy-efficiency of the resulting system is studied. By altering the array configuration, the data communication scheme among the CPEs must also be changed. This has a net effect on both the energy consumption spent on data movement as well as on the overall compute performance. It is shown that by using array topology that is customized to the input DFG, up to 28% improvement in energy-efficiency could be achieved. An exploratory framework based on a genetic algorithm was developed that allows us to obtain such application-specific connection network. Such degree of customization is possible only with the programmability of FPGAs. Moreover, such topology adaptation can be achieved rapidly as only routings between a fixed set of pre-placed CPEs are required. Implementations using ArchSyn and an existing FPGA compilation tool xPilot were compared. ArchSyn gave a 2X better energy consumption and a 11X better energy-delay product for computation with very regular and simple data dependency. For computation with irregular data dependency, the energy consumption and energy-delay product improvement was 9.6X and 199X. / published_or_final_version / Electrical and Electronic Engineering / Doctoral / Doctor of Philosophy

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Sound source localization with microphone arrays has received considerable attention as a means for the automated tracking of individuals in an enclosed space and as a necessary component of any general-purpose speech capture and automated camera pointing system. A novel computationally efficient method compared to traditional source localization techniques is proposed and is both theoretically and experimentally investigated in this research. This thesis first reviews the previous work in this area. The evolution of a new localization algorithm accompanied by an array structure for audio signal localization in three dimensional space is then presented. This method, which has similarities to the structure of the eye, consists of a novel hemispherical microphone array with microphones on the shell and one microphone in the center of the sphere. The hemispherical array provides such benefits as 3D coverage, simple signal processing and low computational complexity. The signal processing scheme utilizes parallel computation of a special and novel closeness function for each microphone direction on the shell. The closeness functions have output values that are linearly proportional to the spatial angular difference between the sound source direction and each of the shell microphone directions. Finally by choosing directions corresponding to the highest closeness function values and implementing linear weighted spatial averaging in those directions we estimate the sound source direction. The experimental tests validate the method with less than 3.10 of error in a small office room. Contrary to traditional algorithmic sound source localization techniques, the proposed method is based on parallel mathematical calculations in the time domain. Consequently, it can be easily implemented on a custom designed integrated circuit.

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