Thesis (M.E.)--University of Canterbury, 2005. / Typescript (photocopy). Includes bibliographical references (p. [109]-116). Also available via the World Wide Web.

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Several adaptive filters were investigated to enhance speech auditory brainstem responses (speech ABR). The objective was to shorten the long recording time currently needed by the standard coherent averaging method to obtain acceptable performance, which has limited the clinical adoption of speech ABR. Five algorithms were implemented: Wiener Filter (WF), Steepest Descent (SD), Adaptive Noise Cancellation (ANC) based on Least-Mean-Square error (LMS) and normalized LMS error (nLMS), and a multi-adaptive cascade combination of SD and LMS. The performance of the adaptive filters was assessed on speech ABR data gathered from several subjects and compared with coherent averaging using the overall Signal-to-Noise Ratio (SNR), the local SNR around the fundamental frequency and the first formant, and Mean-Square-Error (MSE) in the time and frequency domains. The adaptive filters could reduce the time needed, by at least one order of magnitude, for obtaining comparable signal quality as that obtained with coherent averaging.

Thesis (M.S.)--University of Missouri--Rolla, 2007. / Vita. The entire thesis text is included in file. Title from title screen of thesis/dissertation PDF file (viewed December 3, 2007) Includes bibliographical references (p. 65-66).

Increasing densities on magnetic data storage devices leads to problems of severe intersymbol interference (ISI), additive noise and non-linearities. Advanced detection strategies for magnetic recording channels fall into two categories: partial response equalization with maximum likelihood decoding and decision feedback equalization. This study focuses on doing an adaptive all-pass forward filter for the decision feedback channel. The decision feedback channel can be equalized by a low-order continuous-time filter, and does not require a transversal filter with high-precision multiplication. This results in considerable savings in both power consumption and chip die area. One problem that has yet to be addressed is how to adaptively set the coefficients of the all-pass filter. This thesis examines the design and performance of an adaptive all-pass filter. The performances in terms of the mean-squared error (MSE) of a first- and second-order all-pass are evaluated. They are compared to a conventional FIR filter design of various lengths. An adaptive algorithm based on the least mean-squared (LMS) error is developed and characterized over a range of storage densities. Since this does not require sampling of the filter input or any states of the forward filter, the system could be realized in continuous-time up to the decision device. Numerical simulations for various data densities and noise variances are done to verify the theoretically expected performance and the adaptation behavior of the all-pass. / Graduation date: 1997

Several adaptive filters were investigated to enhance speech auditory brainstem responses (speech ABR). The objective was to shorten the long recording time currently needed by the standard coherent averaging method to obtain acceptable performance, which has limited the clinical adoption of speech ABR. Five algorithms were implemented: Wiener Filter (WF), Steepest Descent (SD), Adaptive Noise Cancellation (ANC) based on Least-Mean-Square error (LMS) and normalized LMS error (nLMS), and a multi-adaptive cascade combination of SD and LMS. The performance of the adaptive filters was assessed on speech ABR data gathered from several subjects and compared with coherent averaging using the overall Signal-to-Noise Ratio (SNR), the local SNR around the fundamental frequency and the first formant, and Mean-Square-Error (MSE) in the time and frequency domains. The adaptive filters could reduce the time needed, by at least one order of magnitude, for obtaining comparable signal quality as that obtained with coherent averaging.

Active Noise Control (ANC) is an old concept which has generated increased interest over the past 10-15 years. Using the principle of destructive interference of waves, an inverse pressure wave - "anti-sound wave" is generated in order to attenuate the undesired noise. To achieve substantial cancellation of sound, performance of the cancelling sources must be accurately monitored and controlled. This has only become possible with the rapid development of digital signal processing theory and hardware. Most of the early work in the area of ANC has been done in duct silencing using single channel feed forward and feedback control arrangements. Providing that the sound wavelength is large enough (Le. frequency low enough) in comparison with the cross-sectional dimensions of the duct, spherical sound waves can be adequately approximated with plane waves. The problem is then reduced from three to two dimensions, which provides the possibilities for better understanding of the basic mechanisms of active noise cancellation and study of various adaptive control algorithms. The aim of the present work is to systematically investigate ANC methods for outdoor applications, through the development of Electronically Controlled Acoustic Shadow (ECAS) systems. In this work, the problem is fully three-dimensional. Multichannel ANC methods are proposed to be used, to reduce the noise emitted by large vibrating structures, such as power transformers, in the open air. The adopted approach is to design an active sound wall to create a controlled "anti-sound" shadow. In this way unwanted sound can be reduced in the direction of a complaint area. The potential applications for outdoor ANC systems are considerable. There is need to reduce low frequency sound, which is very hard to reduce using conventional methods - very heavy and expensive structures are required. This opens up the whole field of reducing noise from heavy rotating machinery, such as large generators/motors, factory machinery and mills (many of which have to operate 24 hours per day to remain competitive - which in turn causes noise problems). This work is divided into two main parts. First part considers computer modelling, simulations and theoretical investigation of Electronically Controlled Acoustic Shadows (ECAS) systems. It is demonstrated, that these shadows can be superior to acoustic shadows generated naturally by solid barriers. Detailed analysis predicts that deep shadows (> 1 00 dB) are po.ssible, indicating that practical shadows (>20 dB) are potentially achievable. The object of second part of the work is to investigate practical ECAS systems and establish their performance. In Chapters 2 and 3 (PART 1) the system performance at the fundamental, 100Hz frequency of transformer noise is analysed. To investigate the influence of a large number of parameters on the active wall performance, computer modelling of the primary and secondary (cancelling) sources is developed. The acoustic radiation from this primary source distribution is computed in the far field over a given control angle (both azimuthal and elevation angles). Angles between the 150 and 600 in azimuth and 150 to 300 in elevation are co~'sidered. Phase and amplitude of the secondary sources are than computed through the matrix algebra using exact solution of the least squares problem to minimise the sound at the sensor array. Using this modelling important properties of the acoustic shadows generated by active walls are established, and the basic theory to explain these shadows is formulated. No such theory existed previously. The concept of generating an acoustic shadow in the direction of the complaint area, has resulted in the acoustic properties of a 15°xI5° reference shadow being established in detail. It appears that any arbitrary shadow at this frequency can then be constructed by an addition of the~~ reference shadows, the shadow depth depending on the density of the cancellers per unit angle. Deep shadows in access of 100 dB are predicted, making practical shadows from real sources a possibility. It is now feasible to predict and optimise the future performance of proposed active wall configurations using the computer modelling and developed theory. Further. in the first part of the document (Chapter 4). acoustic interference across high frequency finite Source distributions is studied. The basic theory of non compact sources is considered and the possibility of continuous source representation with a finite number of discrete sources is discussed. The concept of non discreteness or poor discrete representation is established. Here, the .~coustic wavelength is considered small compared to the separation distance between discrete sources. The extent of the near field from these discrete source arrays is also established. where the simplified far field radiation equation breaks down. Finally, in Chapter 4. the optimisation and performance of cancelling arrays to create acoustic shadows from non compact. discrete representation of finite source distributions is investigated.

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