Phased array receivers are a set of antenna elements that are capable of forming multiple simultaneous beams over a field of view. In radio astronomy, the study of deep space radio sources, a phased array feed (PAF) is placed at the focus of a large dish telescope that spatially samples the focal plane. PAFs provide an increase in the field of view as compared to the traditional single pixel horn feed, thereby increasing survey speed while maintaining low sensitivity. Phased arrays are also capable of radio frequency interference (RFI) mitigation which is useful in both radio astronomy and wireless communications when detecting signals in the presence of interferers such as satellites. Digital signal processing algorithms are used to process and analyze data provided by phased array receivers. During the commissioning of the Focal-plane L-band Array feed for the Green Bank telescope (FLAG), sensitivity consistent with an equivalent system temperature below 18 K was measured. To demonstrate the astronomical capability of the receiver, a pulsar (PSR B2011+38) was detected, and an HI source (NGC4258) was mapped with the real-time beamformer and fine channel correlator, respectively. This work also details improvements made to the software of the FLAG digital backend such as the design and implementation of an algorithm to remove scalloping ripple from the spectrum of two cascading polyphase filter banks (PFB). This work will also provide a brief introduction to a model-based beam interpolation algorithm capable of increasing spatial resolution of radio source maps as well as reducing time spent performing calibration. The development of a phased array receiver digital back end for the Office of Naval Research (ONR) is also detailed. This broadband system will be capable of communication in hostile RFI-rich environments with the aid of a real-time RFI mitigation algorithm currently implemented in software. This algorithm will be compatible with other PAF receiver systems and will enable RFI mitigation in other applications such as radio astronomy. This work will provide details on the implementation of this algorithm, the development and modification of other system software as well as full system tests of the 150 MHz bandwidth receiver have been conducted and will be shown in this document.

Binaural beamforming technology, which is based on the auditory perception of both ears, uses a wireless data connection to exchange data between the right-side and the left-side hearing aids. Over the years, several multichannel speech enhancement algorithms have been used in the hearing aid industry. For example, beamforming algorithms work by keeping a target signal undistorted while attenuating the noise fields (such as diffuse noise or white noise) and the interferers from different directions. Fixed and adaptive algorithms of this nature have been under active investigation by the hearing aid industry. Although binaural beamforming hearing aids designs have shown better performance than single-channel based hearing aids or bilateral hearing aids, the performance of binaural beamforming still suffers from errors in the direction of arrival estimates, i.e., errors which occur when the right set of steering vectors is used in a beamformer design but the target signal source is not located at the direction considered in the design. Therefore, this thesis is devoted to find and propose structures showing more robustness to errors in the direction of arrival estimates. The focus is mainly on the Generalized Sidelobe Canceller (GSC) structure and several binaural beamforming algorithms and configurations are proposed in this thesis as alternatives for the fixed beamformer and blocking matrix units of the GSC. The proposed algorithms show promise of providing wider notch and/or wider beam possibilities, as well as providing greater noise reduction and superior adaptive null positioning capabilities. The algorithms proposed in this thesis were simulated in MATLAB using recorded signals and data provided by a hearing aid firm, to assess their utility for improving hearing aid performance. The results demonstrated a superiority over algorithms currently in use in industry.

This thesis describes a real-time software-defined-radio implementation of a two source distributed beamformer. The technique in this thesis can be used to synchronize the carriers of two single antenna wireless transmitters (i.e. ``sources") with independent local clocks so that their bandpass transmissions arrive in-phase at an intended receiver (i.e. ``destination"). Synchronization is achieved via: (i) an unmodulated beacon transmitted by the destination to the sources and (ii) a pair of secondary unmodulated beacons between the sources. No explicit channel state information is exchanged between the sources and/or the destination. Using this method, it is possible to realize a two-source distributed beamformer that provides a reduction in overall transmit energy and increased security due to the directionality of the transmitted signal. System characterization results are provided along with experimental results for both time-invariant and time-varying channels. The experimental results in this thesis confirm the theoretical predictions and also provide explicit guidelines for a real-time implementation of a two-source distributed beamforming system.

This thesis considers the design of a beamformer which can enhance desired signals in an environment consisting of broadband nearfield and/or farfield sources. The thesis contains: a formulation of a set of analysis tools which can provide insight into the intrinsic structure of array processing problems; a methodology for nearfield beamforming; theory and design of a general broadband beamformer; and a consideration of a coherent nearfield broadband adaptive beamforming problem. To a lesser extent, the source localization problem and background noise modeling are also treated. ¶: A set of analysis tools called modal analysis techniques which can be used to a solve wider class of array signal processing problems, is first formulated. The solution to the classical wave equation is studied in detail and exploited in order to develop these techniques. ¶: Three novel methods of designing a beamformer having a desired nearfield broadband beampattern are presented. The first method uses the modal analysis techniques to transform the desired nearfield beampattern to an equivalent farfield beampattern. A farfield beamformer is then designed for a transformed farfield beampattern which, if achieved, gives the desired nearfield pattern exactly. The second method establishes an asymptotic equivalence, up to complex conjugation, of two problems: (i) determining the nearfield performance of a farfield beampattern specification, and (ii) determining the equivalent farfield beampattern corresponding to a nearfield beampattern specification. Using this reciprocity relationship a computationally simple nearfield beamforming procedure is developed. The third method uses the modal analysis techniques to find a linear transformation between the array weights required to have the desired beampattern for farfield and nearfield, respectively. ¶: An efficient parameterization for the general broadband beamforming problem is introduced with a single parameter to focus the beamformer to a desired operating radius and another set of parameters to control the actual broadband beampattern shape. This parameterization is derived using the modal analysis techniques and the concept of the theoretical continuous aperture. ¶: A design of an adaptive beamformer to operate in a signal environment consisting of broadband nearfield sources, where some of interfering signals may be correlated with desired signal is also considered. Application of modal analysis techniques to noise modeling and broadband coherent source localization conclude the thesis.

The thesis examines the problem of joint receive and transmit beamforming for a wireless network which consists of one relay node equipped with multiple antennas. The transmitter and the receiver are single antenna systems. The communication system consists of two phases. In the first phase the transmitter sends the information symbol to the relay while in the second phase, the relay re-transmits a linearly transformed version of the vector of its received signals. The concept of general-rank beamforming is applied to this communication scheme for the case of the uplink (transmitter-relay) and downlink (relay-receiver) channel vectors being statistically independent and statistically dependent. In the general-rank beamforming approach, the multi-antenna relay multiplies the received signal vector with a general-rank complex weight matrix and re-transmits each entry of the output vector on the corresponding antenna. The thesis presents a closed form solution to the general-rank beamforming power minimization problem with proof that for statistically independent uplink and downlink channels, the general-rank beamforming approach results in a rank-one solution for the beamforming matrix. The simulation results have shown that when the generalrank beamformer is applied to the case of statistically dependent uplink and downlink channels, the general-rank beamforming technique significantly outperforms the separable receive and transmit beamforming method. / UOIT

Les implants cochléaires sont des appareils destinés à la réhabilitationdes surdités profondes et totales. Ils assurent la stimulation du nerf auditif en plaçant des électrodes dans la cochlée. Différentes études ont été établis afin d'améliorer l'intelligibilité de la parole dans le bruit chez le patientporteur de cet appareil. Les techniques bilatérales et binaurales permettent dereproduire une audition binaurale, car les deux oreilles sont simulées (commepour les personnes normo-entendantes). Ainsi la localisation et la perceptiondes sons environnants sont améliorées par rapport à une implantationmonaurale. Toutefois, les capacit'es de reconnaissances des mots sont trèsvite limitées en pr'esence de bruits. Nous avons d'evelopp'es des techniquesde r'eduction de bruit afin d'augmenter les performances de reconnaissance.Des améliorations de 10% à 15% suivant les conditions ont été observées. Néanmoins, si la perception est améliorée par les algorithmes, ils focalisent sur une direction, et ainsi, la localisation est alors réduite à l'angle delocalisation. Une seconde étude a alors été effectuée pour mesurer l'effetdes algorithmes sur la localisation. Ainsi, le beamformer donne les meilleurs résultats de compréhension mais la moins bonne localisation. La ré-injectiond'un pourcentage du signal d'entrée sur la sortie a permis de compenser laperte de la localisation sans détériorer l'intelligibilité. Le résultat de ces deux expériences montre qu'il faut un compromis entre laperception et la localisation des sons pour obtenir les meilleures performances.

Bilateral implantation of a cochlear implant (CI) after a >2 year period of unilateral hearing with a second implant has been shown to result in altered latencies in brainstem responses in children with congenital deafness. In this thesis, a neural source localization method was developed to investigate the effects of unilateral CI use on cortical development after the implantation of a 2nd CI. The electroencephalography (EEG) source localization method is based on the linearly constrained minimum variance (LCMV) vector beamformer and utilizes null constraints to minimize the electrical artifact produced by the CI. The accuracy of the method was assessed and optimized through simulations and comparisons to beamforming with magnetoencephalography (MEG) data. After using cluster analyses to ensure that sources compared across subjects originate from the same neural generators, a study was done to examine the effects of unilateral CI hearing on hemispheric lateralization to monaural responses. It was found that a >2 year period of unilateral hearing results in expanded projections from the 1st implanted ear to the contralateral auditory area that is not reversed by implantation of a 2nd CI. A subsequent study was performed to examine the effects of unilateral CI hearing on the contributions of the 1st and 2nd implanted ears to the binaural response. It was found that in children with > 2 years of unilateral hearing, the binaural response is dominated by the 1st implanted ear. Together, these results suggest that the delay between the 1st and 2nd CI should be minimized in bilateral implantation to avoid dominance of auditory pathways from the 1st implanted ear. This dominance limits developmental competition from the 2nd CI and potentially contributes to poorer performance in speech detection in noise tasks.

Bilateral implantation of a cochlear implant (CI) after a >2 year period of unilateral hearing with a second implant has been shown to result in altered latencies in brainstem responses in children with congenital deafness. In this thesis, a neural source localization method was developed to investigate the effects of unilateral CI use on cortical development after the implantation of a 2nd CI. The electroencephalography (EEG) source localization method is based on the linearly constrained minimum variance (LCMV) vector beamformer and utilizes null constraints to minimize the electrical artifact produced by the CI. The accuracy of the method was assessed and optimized through simulations and comparisons to beamforming with magnetoencephalography (MEG) data. After using cluster analyses to ensure that sources compared across subjects originate from the same neural generators, a study was done to examine the effects of unilateral CI hearing on hemispheric lateralization to monaural responses. It was found that a >2 year period of unilateral hearing results in expanded projections from the 1st implanted ear to the contralateral auditory area that is not reversed by implantation of a 2nd CI. A subsequent study was performed to examine the effects of unilateral CI hearing on the contributions of the 1st and 2nd implanted ears to the binaural response. It was found that in children with > 2 years of unilateral hearing, the binaural response is dominated by the 1st implanted ear. Together, these results suggest that the delay between the 1st and 2nd CI should be minimized in bilateral implantation to avoid dominance of auditory pathways from the 1st implanted ear. This dominance limits developmental competition from the 2nd CI and potentially contributes to poorer performance in speech detection in noise tasks.

The capacity of a cellular system is limited by two different phenomena, namely multipath fading and multiple access interference (MAI). A Two Dimensional (2-D) receiver combats both of these by processing the signal both in the spatial and temporal domain. An ideal 2-D receiver would perform joint space-time processing, but at the price of high computational complexity. In this dissertation we investigate computationally simpler technique termed as a Beamformer-Rake. In a Beamformer-Rake, the output of a beamformer is fed into a succeeding temporal processor to take advantage of both the beamformer and Rake receiver. Wireless service providers throughout the world are working to introduce the third generation (3G) cellular service that will provide higher data rates and better spectral efficiency. Wideband CDMA (WCDMA) has been widely accepted as one of the air interfaces for 3G. A Beamformer-Rake receiver can be an effective solution to provide the receivers enhanced capabilities needed to achieve the required performance of a WCDMA system. This dissertation investigates different Beamformer-Rake receiver structures suitable for the WCDMA system and compares their performance under different operating conditions. This work develops Beamformer-Rake receivers for WCDMA uplink that employ Eigen-Beamforming techniques based on the Maximum Signal to Noise Ratio (MSNR) and Maximum Signal to Interference and Noise Ratio (MSINR) criteria. Both the structures employ Maximal Ratio Combining (MRC) to exploit temporal diversity. MSNR based Eigen-Beamforming leads to a Simple Eigenvalue problem (SE). This work investigates several algorithms that can be employed to solve the SE and compare the algorithms in terms of their computational complexity and their performance. MSINR based Eigen-Beamforming results in a Generalized Eigenvalue problem (GE). The dissertation describes several techniques to form the GE and algorithms to solve it. We propose a new low-complexity algorithm, termed as the Adaptive Matrix Inversion (AMI), to solve the GE. We compare the performance of the AMI to other existing algorithms. Comparison between different techniques to form the GE is also compared. The MSINR based beamforming is demonstrated to be superior to the MSNR based beamforming in the presence of strong interference. There are Pilot Symbol Assisted (PSA) beamforming techniques that exploit the Minimum Mean Squared Error (MMSE) criterion. We compare the MSINR based Beamformer-Rake with the same that utilizes Direct Matrix Inversion (DMI) to perform MMSE based beamforming in terms of Bit Error Rate (BER). In a wireless system where the number of co-channel interferers is larger than the number of elements of a practical antenna array, we can not perform explicit null-steering. As a result the advantage of beamforming is partially lost. In this scenario it is better to attain diversity gain at the cost of spatial aliasing. We demonstrate this with the aid of simulation. Orthogonal Frequency Division Multiplexing (OFDM) is a multi-carrier technique that has recently received considerable attention for high speed wireless communication. OFDM has been accepted as the standard for Digital Audio Broadcast (DAB) and Digital Video Broadcast (DVB) in Europe. It has also been established as one of the modulation formats for the IEEE 802.11a wireless LAN standard. OFDM has emerged as one of the primary candidates for the Fourth Generation (4G) wireless communication systems and high speed ad hoc wireless networks. We propose a simple pilot symbol assisted frequency domain beamforming technique for OFDM receiver and demonstrate the concept of sub-band beamforming. Vector channel models measured with the MPRG Viper test-bed is also employed to investigate the performance of the beamforming scheme. / Ph. D.

Time delay estimation (TDE)-based algorithms for estimation of direction of arrival (DOA) have been most popular for use with speech signals. This is due to their simplicity and low computational requirements. Though other algorithms, like the steered response power with phase transform (SRP-PHAT), are available that perform better than TDE based algorithms, the huge computational load required for this algorithm makes it unsuitable for applications that require fast refresh rates using short frames. In addition, the estimation errors that do occur with SRP-PHAT tend to be large. This kind of performance is unsuitable for an application such as video camera steering, which is much less tolerant to large errors than it is to small errors. We propose an improved TDE-based DOA estimation algorithm called time delay selection (TIDES) based on either minimizing the weighted least squares error (MWLSE) or minimizing the time delay separation (MWTDS). In the TIDES algorithm, we consider not only the maximum likelihood (ML) TDEs for each pair of microphones, but also other secondary delays corresponding to smaller peaks in the generalized cross-correlation (GCC). From these multiple candidate delays for each microphone pair, we form all possible combinations of time delay sets. From among these we pick one set based on one of the two criteria mentioned above and perform least squares DOA estimation using the selected set of time delays. The MWLSE criterion selects that set of time delays that minimizes the least squares error. The MWTDS criterion selects that set of time delays that has minimum distance from a statistically averaged set of time delays from previously selected time delays. Both TIDES algorithms are shown to out-perform the ML-TDE algorithm in moderate signal to reverberation ratios. In fact, TIDES-MWTDS gives fewer large errors than even the SRP-PHAT algorithm, which makes it very suitable for video camera steering applications. Under small signal to reverberation ratio environments, TIDES-MWTDS breaks down, but TIDES-MWLSE is still shown to out-perform the algorithm based on ML-TDE. / Master of Science