Background noise is particularly damaging to speech intelligibility for people with hearing loss. The problem of reducing noise in hearing aids is one of great importance and great difficulty. Over the years, many solutions and different algorithms have been implemented in order to provide the optimal solution to the problem. Beamforming has been used for a long time and has therefore been extensively researched. Studying the performance of Minimum Variance Distortionless Response (MVDR) beamforming with a three- and four- microphone array compared to the conventional two-microphone array, the aim is to implement a speech signal enhancement and a noise reduction algorithm. By using multiple microphones, it is possible to achieve spatial selectivity, which is the ability to select certain signals based on the angle of incidence, and improve the performance of noise reduction beamformers. This thesis proposes the use of beamforming, an existing technique in order to create a new way to reduce noise transmitted by hearing aids. In order to reduce the complexity of that system, we use hybrid cascades, which are simpler beamformers of two inputs each and connected in series. The configurations that we consider are a three-microphone linear array (monaural beamformer), a three-microphone configuration with a two-microphone linear array and the 3rd microphone in the ear (monaural beamformer), a three-microphone configuration with a two-microphone linear array and the 3rd microphone on contra-lateral ear (binaural beamformer), and finally four-microphone configurations. We also investigate the performance improvement of the beamformer with more than two microphones for the different configurations, against the two-microphone beamformer reference. This can be measured by using objective measurements, such as the amount of noise suppression, target energy loss, output SNR, speech intelligibility index and speech quality evaluation. These objective measurements are good indicators of subjective performance. In this project, we prove that most hybrid structures can perform satisfyingly well compared to the full complexity beamformer. The low complexity beamformer is designed with a fixed target location (azimuth), where its weights are calibrated with respect to a target signal located in front of the listener and for a diffuse noise field. Both second- and third- order beamformers are tested in different acoustic scenarios, such as a car environment, a meeting room, a party occasion and a restaurant place. In those scenarios, the target signal is not arriving at the hearing aid directly from the front side of the listener and the noise field is not always diffuse. We thoroughly investigate what are the performance limitations in that case and how well the different cascades can perform. It is proven that there are some very critical factors, which can affect the performance of the fixed beamformer, concerning all the hybrid structures that were examined. Finally, we show that lower complexity cascades for both second- and third- order beamformers can perform similarly well as the full complexity beamformers when tested for a set of multiple Head Related Transfer Functions (HRTFs) that correspond to a real head shape.

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This thesis describes a blind beamforming technique for GPS receivers. It improves the performance of a GPS receiver by mitigating interference and enhancing GPS signals separately and has a three-stage structure. The technique is based on a linear antenna array and integrates the eigendecomposition based subspace and multiple independent beamforming techniques. A signal model is carefully constructed. Particular emphasis is placed upon the projection matrix derived from the subspace technique. The effect of interference and phase error on this technique is discussed. This technique is tested and compared to null steering and MMSE technique using simulated data for a number of interference environments. Furthermore, the proposed technique is applied to real data and shows several advantages over simple null steering. / Thesis (M.Eng.Sc.) - University of Adelaide, School of Electrical and Electronic Engineering, 2008

In this thesis, multidimensional signal processing techniques to mitigate disturbances in synthetic aperture systems such as radio telescopes are investigated. Here, two computationally efficient three-dimensional (3D) spatio-temporal (ST) finite impulse response (FIR) cone filter bank structures are proposed. Furthermore, a strategy is proposed to design 3D ST FIR frustum filter banks, having double-frustum-shaped passbands oriented along the temporal axis, derived from appropriate 3D ST FIR cone filter banks. Both types of cone and frustum filter banks are almost alias free and provide near-perfect reconstruction. In the proposed cone and frustum filter banks, both temporal and spatial filtering operations can be carried out at a significantly lower rate compared to previously reported 3D ST FIR cone filter banks implying lower power consumption. Furthermore, the proposed cone and frustum filter banks require a significantly lower computational complexity than previously reported 3D ST FIR cone and frustum filter banks. Importantly, this is achieved without deteriorating the improvement in signal-to-interference-plus-noise ratio. A theoretical analysis of brightness distribution (BD) errors caused by parameter perturbations and mismatches among the transfer functions of receivers employed in synthetic aperture systems is presented. First, the BD errors caused by perturbations in the transfer functions of low noise amplifiers (LNAs) and anti-aliasing filters (AAFs) are considered, and the characteristics of the additive BD error and its effects on synthesized BDs are thoroughly analyzed. Second, the conditions that should be satisfied by the transfer functions of digital beamformers to eliminate the BD errors caused by their phase responses are examined. The sufficient condition to eliminate the BD errors is that the transfer functions are matched, and, interestingly, the phase responses are not necessary to be linear. Furthermore, the BD errors caused by typical tolerances of passive L and C elements used to implement the AAFs and those caused by the random variations of gain from LNA to LNA are quantified through numerical simulations. The simulations indicate that substantial BD errors are observed at frequencies that are close to the passband edge of the AAFs. / Graduate

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