Optimum filter-and-sum beamforming is useful for array systems that suffer from spatially correlated noise and interference over large bandwidth. The set of finite impulse response (FIR) filter coefficients used to implement the optimum filter-and-sum beamformer are selected to optimize signal-to-noise ratio (SNR) and reduce interference from the certain directions. However, these array systems may also be vulnerable to dispersion caused by physical components such as antennas and cables, especially when the dispersion is unequal between sensors. The unequal responses can be equalized by using FIR filters. Although the problems of optimum-SNR beamforming, interference mitigation, and per-sensor dispersion have previously been individually investigated, their combined effects and strategies for mitigating their combined effects do not seem to have been considered. In this dissertation, combination strategies for optimum filter-and-sum beamforming and sensor dispersion correction are investigated. Our objective is to simultaneously implement optimum filter-and-sum beamforming and per-sensor dispersion correction using a single FIR filter per sensor. A contribution is to reduce overall filter length, possibly also resulting in a significant reduction in implementation complexity, power consumption, and cost. Expressions for optimum filter-and-sum beamforming weights and per-sensor dedispersion filter coefficients are derived. One solution is found via minimax optimization. To assess feasibility, the cost is analyzed in terms of filter length. These designs are considered in the context of LWA1, the first ``station'' of the Long Wavelength Array (LWA) radio telescope, consisting of 512 bowtie-type antennas and operating at frequencies between 10 MHz and 88 MHz. However, this work is applicable to a variety of systems which suffer from non-white spatial noise and directional interference and are vulnerable to sensor dispersion; e.g., sonar arrays, HF/VHF-band riometers, radar arrays, and other radio telescopes. / Ph. D.

A new technique to de-Dopplerize microphone signals from moving sources of sound is derived. Currently available time domain de-Dopplerization techniques require oversampling and interpolation of the microphone time data. In contrast, the technique presented in this dissertation performs the de-Dopplerization entirely in the frequency domain eliminating the need for oversampling and interpolation of the microphone data. As a consequence, the new de-Dopplerization technique is computationally more efficient. The new de-Dopplerization technique is then implemented into a frequency domain beamforming algorithm to locate moving sources of sound. The mathematical formulation for the implementation of the new de-Dopplerization technique is presented for sources moving along a linear trajectory and for sources moving along a circular trajectory, i.e. rotating sources. The resulting frequency domain beamforming method to locate moving sound sources is then validated using numerical simulations for various source configurations (e.g. emission angle, emission frequency, and source velocity), and different processing parameters (e.g. time window length). Numerical datasets for sources with linear motion as well as for rotating sources were simulated. For comparison purposes, selected datasets were also processed using traditional time domain beamforming. The results from the numerical simulations show that the frequency domain beamforming method is at least 10 times faster than the traditional time domain beamforming method with the same performance. Furthermore, the results show that as the number of microphones and/or grid points increase, the processing time for the traditional time domain beamforming method increases at a rate 20 times larger than the rate of increase in processing time of the new frequency domain beamforming method. / Ph. D.

The huge advancement in the textiles industry and the accurate control on the mechanization process coupled with cost-effective manufacturing offer an innovative environment for new electronic systems, namely electronic textiles. The abundance of fabrics in our regular life offers immense possibilities for electronic integration both in wearable and large-scale applications. Augmenting this technology with a set of precepts and a simulation environment creates a new software/hardware architecture with widely useful implementations in wearable and large-area computational systems. The software environment acts as a functional modeling and testing platform, providing estimates of design metrics such as power consumption. The construction of an electronic textile (e-textile) hardware prototype, a large-scale acoustic beamformer, provides a basis for the simulator and offers experience in building these systems. The contributions of this research focus on defining the electronic textile architecture, creating a simulation environment, defining a networking scheme, and implementing hardware prototypes. / Ph. D.

The Eight-meter-wavelength Transient Array (ETA) system is a unique implementation of an array-based radio telescope. The instrument is designed to further astronomy by detecting and characterizing dispersed pulses received between 29–47 MHz. To aid data processing of radio signals received through 24 antennas, the ETA system performs real-time stream processing as data is passed from antennas to hard disk storage. The processing includes digital sampling, downconversion, filtering, Fast Fourier Transforms, and beamforming operations and is performed by 28 commercial-off-the-shelf (COTS) FPGA boards. Sixteen of the FPGA boards constitute the reconfigurable computing cluster (RCC) which performs the FFT and beamforming operations and is the focus of this thesis. The FPGA-based architecture allows the RCC to provide the high computational and communication throughput required by the ETA system. In addition, the FPGA design allows for a custom processing data path, parallel processing, global synchronization, and rapid development at a low cost. / Master of Science

The usage of Unmanned Aerial Systems (UASs), such as quadcopters and hexacopters, has steadily increased over the past few years in both recreational and commercial use. This increased availability to purchase such systems has also given rise to many safety and security concerns. A common concern is that the misuse of a UAS can cause damage to airplanes and helicopters in and around airports. Another growing concern is the use of UASs for terrorist intentions such as using the UAS as a remote controlled bomb. There is clearly a need to be able to detect the presence of unwanted UASs in restricted areas. This thesis work presents the beginning work towards a method to detect the presence of these UASs using the blade pass frequency (BPF) of the motors and rotors of a home made quadcopter. A low cost uniform linear microphone array is first used to perform a simple delay-and-sum beamformer to spatially filter out noise sources. The beamformer output is then divided into sub-bands using three bandpass filters centered on the expected location of the fundamental BPF and its 2nd and 3rd harmonics. For each sub-band, an adaptive filter called an adaptive line enhancer is used to extract and enhance the narrowband signals. The response of the adaptive filters are then used to detect the quadcopter by looking for the presence of the 2nd and 3rd harmonics of the fundamental BPF. Static tests of the quadcopter out in a field showed promising results for this method with the ability to detect up to the 3rd harmonic 90ft away and the 2nd harmonic 130 ft away. / Master of Science / The usage of Unmanned Aerial Systems (UASs), such as quadcopters and hexacopters, has steadily increased over the past few years in both recreational and commercial use. This increased availability to purchase such systems has also given rise to many safety and security concerns. A common concern is that the misuse of a UAS can cause damage to airplanes and helicopters in and around airports. Another growing concern is the use of UASs for terrorist intentions such as using the UAS as a remote controlled bomb. There is clearly a need to be able to detect the presence of unwanted UASs in restricted areas. This thesis work presents the beginning work towards a method to detect the presence of a home made quadcopter based on the sound it produces. A series of microphone are first used to remove surrounding sounds that could interfere with the quadcopter’s sound. The output of this processes is then divided into smaller sections using three filters centered on the expected location of the most important and information rich parts of the quadcopter’s sound. For each section, a final filter is used to extract and enhance the signals of interest produced by the quadcopter. The response of these filters are then used to detect whether the quadcopter is present or not. Static tests of the quadcopter out in a field showed promising results for this method with the ability to detect the quadcopter 90 to 130 ft away.

The blind source separation problem has received a great deal of attention in previous years. The aim of this problem is to estimate a set of original source signals from a set of linearly mixed signals through any number of signal processing techniques. While many methods exist that attempt to solve the blind source separation problem, a new technique is being used that uniquely separates audio sources as they are received from a microphone array. In this thesis a new algorithm is proposed that that utilizes the ICA algorithm in conjunction with a filtering technique that separates source signals and then removes sources of interference so that a signal of interest can be accurately tracked. Experimental results will compare a common blind source separation technique to the new algorithm and show that the new algorithm can detect a signal of interest and accurately track it as it moves through an anechoic environment. / Master of Science

This thesis presents simulation data on array operation in wideband communication systems. It is shown that array structures with closer inter-element spacing outperform structures with much larger inter-element spacing. It is also shown that circular structures outperform linear structures. This performance difference between the classifications of arrays is due largely to the circular array's ability to handle high levels of interference. Even though a diversity combining scheme (MRC) was used in the simulator, the arrays provided interference rejection capabilities due to the closely spaced antenna elements. Though diversity does provide a gain in received signal, relative to the faded signal, realized diversity gain only comes about once interference has been mitigated. This thesis work showed that in an environment with a lot of interferers, the rejection of those interferers by an array is of utmost importance, even more than fading mitigation. / Master of Science

Orthogonal Frequency Division Multiplexing (OFDM) has recently gained a lot of attention and is a potential candidate for Fourth Generation (4G) wireless systems because it promises data rates up to 10Mbps. A variation of OFDM is Multi-Carrier CDMA (MC-CDMA) which is an OFDM technique where the individual data symbols are spread using a spreading code in the frequency domain. The spreading code associated with MC-CDMA provides multiple access technique as well as interference suppression. Often times in cellular and military environments the desired signal can be buried below interference. In such conditions, the processing gain associated with the spreading cannot provide the needed interference suppression. This research work investigates multi-antenna receivers for OFDM and MC-CDMA systems; specifically this works investigates adaptive antenna algorithms for MC-CDMA for very different channel conditions. Frequency domain beamforming is studied in this research predominantly through simulation. As an alternative a time domain beamforming is also studied. Time variations in the channel can disrupt the orthogonality between subcarriers. Minimum Mean Square Error (MMSE) detection coupled with MMSE beamforming is proposed for time varying channels. Semi-analytic results are derived to study the Bit Error Rate (BER) performance. These results show significant performance improvement in the presence of interference. Joint MMSE weights in space and frequency is also investigated and semi-analytic results are derived to study their BER performance. / Master of Science

This thesis describes the design and implementation of a 4-channel microphone array that is an adaptive beamformer used for hands-free telephony in a noisy environment. The microphone signals are amplified, then sent to an A/D converter. The microprocessor board takes the data from the 4 channels and utilizes digital signal processing to determine the direction-of-arrival of the sources and create an output which 'steers' the microphone array to the desired look direction while trying to minimize the energy of interference sources and noise. All of the processing for this thesis will be done on a computer using MATLAB. The MUSIC algorithm is used for direction finding in this thesis. It is shown to be effective in estimating direction-of-arrival for 1 speech source and 2 speech sources that are spaced fairly apart, with significant results down to a -5 dB SNR even. The MUSIC algorithm requires knowledge of the number of sources a priori, requiring an estimator for the number of sources. Though proposed estimators for the number of sources were examined, an effective estimator was not encountered for the case where there are multiple speech sources. Beamforming methods are examined which utilize knowledge of the source direction-of-arrival from the MUSIC algorithm. The input is split into 6 subbands such that phase-steered beamforming would be possible. Two methods of phase-steered beamforming are compared in both narrowband and wideband scenarios, and it is shown that phase-steering the array to the desired source direction-of-arrival has about 0.3 dB better beamforming performance than the simple time-delay steered beamformer using no subbands. As the beamforming solution is inadequate to achieve desired results, a generalized sidelobe canceler (GSC) is developed which incorporates a beamformer. The sidelobe canceler is evaluated using both NLMS and RLS adaptation. The RLS algorithm inherently gives better results than the NLMS algorithm, though the computational complexity renders the solution impractical for implementation with today's technology. A testing setup is presented which involves a linear 4-microphone array connected to a DSP chip that collects the data. Tests were done using 1 speech source and a model of the car noise environment. The sidelobe canceler's performance using 6 subbands (phase-delay GSC) and using 1 band (time-delay GSC) with NLMS updating are compared. The overall SNR improvement is determined from the signal and noise input and output powers, with signal-only as the input and noise-only as the input to the GSC. The phase-delay GSC gives on average 7.4 dB SNR improvement while the time-delay GSC gives on average 6.2 dB SNR improvement. / Master of Science

With usable spectrum becoming increasingly more congested, the need for robust, adaptive communications to take advantage of spatially-separated signal sources is apparent. Traditional phased array beamforming techniques used for interference removal rely on perfect calibration between elements and precise knowledge of the array configuration; however, if the exact array configuration is not known (unknown or imperfect assumption of element locations, unknown mutual coupling between elements, etc.), these traditional beamforming techniques are not viable, so a blind beamforming approach is required. A novel blind beamforming approach is proposed to address complex narrow-band interference environments where the precise array configuration is unknown. The received signal is decomposed into orthogonal narrow-band partitions using a polyphase filter-bank channelizer, and a rank-reduced version of the received matrix on each sub-channel is computed through reconstruction by retaining a subset of its singular values. The wideband spectrum is synthesized through a near-perfect polyphase reconstruction filter, and a composite wideband spectrum is obtained from the maximum eigenvector of the resulting covariance matrix.The resulting process is shown to suppress numerous interference sources (in special cases even with more than the degrees of freedom of the array), all without any knowledge of the primary signal of interest. Results are validated with both simulation and wireless laboratory over-the-air experimentation. / M.S. / As the number of devices using wireless communications increase, the amount of usable radio frequency spectrum becomes increasingly congested. As a result, the need for robust, adaptive communications to improve spectral efficiency and ensure reliable communication in the presence of interference is apparent. One solution is using beamforming techniques on digital phased array receivers to maximize the energy in a desired direction and steer nulls to remove interference. However, traditional phased array beamforming techniques used for interference removal rely on perfect calibration between antenna elements and precise knowledge of the array configuration. Consequently, if the exact array configuration is not known (unknown or imperfect assumption of element locations, unknown mutual coupling between elements, etc.), these traditional beamforming techniques are not viable, so a beamforming approach with relaxed requirements (blind beamforming) is required. This thesis proposes a novel blind beamforming approach to address complex narrow-band interference in spectrally congested environments where the precise array configuration is unknown. The resulting process is shown to suppress numerous interference sources, all without any knowledge of the primary signal of interest. Results are validated with both simulation and wireless laboratory experimentation conducted with a two-element array, verifying that proposed beamforming approach achieves a similar performance to the theoretical performance bound of receiving packets in AWGN with no interference present.