This thesis presents a comprehensive methodology for predicting and minimising the noise radiated from a constant speed gearbox assembly by means of attaching optimally placed stiffening ribs on the casing. The procedure involves building an FE model of the gearbox, which is updated using modal parameters extracted from a modal test. This is followed by synthesis of the required FRFs with respect to the forcing degrees-of-freedom. The forces, which are assumed to act only at the bearings are identified from these FRFs and the measured operational velocities of the casing. The identified forces are then used to excite the updated FE model to re-calculate the vibration velocities. A boundary element method is used to calculate the final radiated sound power to be compared with that measured. The same forces are used later to excite the modified gearbox casing to determine the improvement given by optimised modification. The optimisation study minimises the vibration energy of the casing in 10% bands around critical frequencies, in this case the first two harmonics of the gearmesh frequency. To allow for errors in the model, the excitation is by white noise, so as to produce wide stop bands, rather than minimising the response at particular frequencies. The vibration energy is weighted for radiation efficiency, A-weighting, and relative source strength in the bands. The final optimal stiffener layout is validated through a final vibration and acoustic calculation on the updated gearbox model using the forces identified in the earlier steps. The study of one particular gearbox concludes that i) the proposed hybrid optimisation scheme produces a theoretical effective noise reduction of 3 dBA for the total sound power. ii) Because the gearmesh harmonics were targeted, a further 5 dB improvement was effectively gained by eliminating the tonal penalty which otherwise applied. iii) From plate studies it was demonstrated that the stiffening ribs could be attached using epoxy cement (to avoid welding) and that the properties of the cemented joint could be determined by model updating after attaching one rib, so as to obtain a better prediction of the final optimised result.

Local active noise control systems aim to create zones of quiet at specific locations within a sound field. The created zones of quiet generally tend to be small, especially for higher frequencies, and are usually centred at the error sensors. For an observer to experience significant reductions in the noise, the error sensors therefore have to be placed relatively close to an observer’s ears, which is not always feasible or convenient. Virtual sensing methods have been proposed to overcome these problems that have limited the scope of successful local active noise control applications. These methods require non-intrusive sensors that are placed remotely from the desired locations of maximum attenuation. These non-intrusive sensors are used to provide an estimate of the sound pressures at these locations, which can then be minimised by a local active noise control system. This effectively moves the zones of quiet away from the physical locations of the transducers to the desired locations of maximum attenuation, such as a person’s ears. A number of virtual sensing algorithms have been proposed previously. The difference between these algorithms is the structure that is assumed to compute an estimate of the virtual error signals. The question now arises as to whether there is an optimal structure that can be used to solve the virtual sensing problem, which amounts to a linear estimation problem. It is well-known that the Kalman filter provides an optimal structure for solving such problems. An optimal solution to the virtual sensing problem is therefore derived in this thesis using Kalman filtering theory. The proposed algorithm is implemented on an acoustic duct arrangement to demonstrate its effectiveness. The presented experimental results indicate that the zone of quiet was effectively moved away from the physical sensor towards the desired location of maximum attenuation. The previously proposed virtual sensing algorithms have been developed with the aim to create zones of quiet at virtual locations that are assumed spatially fixed within the sound field. Because an observer is very likely to move their head, the desiredlocations of the zones of quiet are generally moving through the sound field rather than being spatially fixed. For effective control, a local active noise control system incorporating a virtual sensing method thus has to be able to create moving zones of quiet that track the observer’s ears. A moving virtual sensing method is therefore developed in this thesis that can be used to estimate the error signals at virtual locations that are moving through the sound field. It is shown that an optimal solution to the moving virtual sensing problem can be derived using Kalman filtering theory. A practical implementation of the developed algorithm is combined with an adaptive feedforward control algorithm and implemented on an acoustic duct arrangement. The presented experimental results illustrate that a narrowband moving zone of quiet that tracks the desired location of maximum attenuation has successfully been created. / http://proxy.library.adelaide.edu.au/login?url= http://library.adelaide.edu.au/cgi-bin/Pwebrecon.cgi?BBID=1291123 / Thesis (Ph.D.) -- University of Adelaide, School of Mechanical Engineering, 2007.

"28th September 2001" / Bibliography: p. 285-310. / xlvi, 350 p. : ill. (some col.), plates (col.) ; 30 cm. / Title page, contents and abstract only. The complete thesis in print form is available from the University Library. / This thesis describes by theory and experiment, the effectiveness of an alternate error sensing strategy suitable for the control of general free field sound radiation problems, namely active intensity. In this research, active intensity error sensing has been rigorously analysed: 1. in the near and far field of the disturbance source; 2. to determine its ability to lead to global control via sound power attenuation; 3. in a real control system. The performance of active intensity error sensors were evaluated both analytically and experimentally in progressively more complex environments to identify their capabilities and limitations. / Thesis (Ph.D.)--University of Adelaide, Dept. of Mechanical Engineering, 2001

Manchester, in the first half of the nineteenth century, held great fascination to many as an example of a town being remade and transformed by the technological developments of the Industrial Revolution. The transformations experienced included the rapid physical expansion of the town and its population and the reordering of its society from a traditional, rigid class structure, involving a powerful aristocracy and working-class citizens, to a new social hierarchy with a numerous and influential middle class—an emerging ‘urban aristocracy’ of people involved in manufacturing, commerce and other professions. As these changes took place, the rest of England (and the Western world) looked on—horrified, shocked, awed, fascinated. It was a different world, and the rapid changes taking place in society created in observers a sense of urgency in describing the effects of those changes—particularly the social problems, which had been shaped by industrial life. / A trend in describing Manchester’s residents as ‘philistines’ and the town, generally, as a ‘cultural wasteland’ took hold at this time and has been perpetuated until fairly recently. This thesis explores this trend—Manchester’s nineteenth-century image—and the impact of contemporary opinion on constructions of social hierarchies and cultural reputations. It also aims to show that there was more to Manchester’s cultural life in the nineteenth century than is widely acknowledged, either by contemporaries from the period or by some scholars today, and that the pursuit and experience of music and certain noises was genuinely wanted by the manufacturing class for a number of key reasons. / Delving further, into studies of soundscapes and ‘noise,’ it becomes ever clearer that sound, and how individuals and societies interact with it and interpret it, acts as an important—though frequently overlooked—signifier of class relations and civic identity. Finally, this thesis aims to reconstruct how Manchester sounded in three principal regions of the town to show how a study of soundscapes helps to articulate how the town was psychologically constructed in the minds of inhabitants and visitors, and how it was sensed and experienced.

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<p>This thesis investigates the possibilities of interleaving multiple Digital to Analog converters in a high speed environment. Algorithms for interleaving and noise shaping as well as filters are tailored for high frequency operation.In the first part of the thesis, algorithms are evaluated and models to simulate errors are created. It was concluded that DAC interleaving is feasible to reach high sample rates. Interleaving or parallelization of the $\Sigma\Delta$ noise shaper proved to not be feasible for the specific application due low oversampling and high speed operation.The second part of the thesis consists of measurements on a custom SP Devices development board. These tests confirm that interleaving of DACs works as intended and that it is possible to increase the output bandwidth beyond the one of a single DAC.</p>

The quality and intelligibility of single channel speech degraded by additive noise remains a challenging problem when only the noisy speech is available. An accurate estimation of the noise spectrum is important for the effective performance of speech enhancement algorithms, especially in nonstationary noise environments. This thesis addresses both two issues. First, a speech enhancement algorithm using harmonic features is introduced. A spectral weighting function is derived by constrained optimization to suppress noise in the frequency domain. Two design parameters are included in the suppression gain, namely the frequency-dependent noise-flooring parameter (FDNFP) and the gain factor. The FDNFP controls the level of admissible residual noise in the enhanced speech, while further enhancement is achieved by adaptive comb filtering using the gain factor with a peak-picking algorithm. Second, a noise estimation algorithm is proposed for nonstationary noise conditions. The speech presence probability is updated by introducing a time-frequency dependent threshold. The frequency dependent smoothing factor for noise estimation is computed based on the estimated speech presence probability in each frequency bin. This algorithm adapts quickly to nonstationary noise environments and preserves more information on weak speech phoneme. The performance of the proposed speech enhancement algorithm is evaluated in terms of Perceptual Evaluation of Speech Quality (ITU-PESQ) scores and Modified Bark Spectral Distortion (MBSD) measures, composite objective measures and listening tests. Our listening tests indicate that 16 listeners on average preferred our harmonic enhanced speech over any of three other approaches about 73% of the time. The performance of the proposed noise estimation algorithm combined with the proposed speech enhancement method in nonstionary noise environments is also tested in terms of ITU-PESQ scores and MBSD measures. Experimental results indicate that the proposed noise estimation algorithm when integrated with the harmonic enhancement method outperforms spectral subtraction, signal subspace method, a perceptually-based enhancement method with a constant noise-flooring parameter, and our original harmonic speech enhancement method in highly nonstationary noise environments.

Recently Kalman filter has been well applied to the problems of image restoration. In this thesis, we apply Kalman filter to estimate the optical transfer function for an imaging system. The signal model is the optical transfer function obtained from a ratio of the defected and clean pictures in frequency domain. There is thermal noise involved in sampling the optical image signal. We model this thermal noise as the additive measurement noise. We remove the thermal noise by Winner filtering. This filtered image is finally restored by the above estimated the optical transfer function. The experiments are set up by the instruments including the video camera, capture card, and personal computer. Experimental results, including the estimation of gamma and noise power, have demonstrated that the estimated optical transfer function is useful for image restoration.

Low-frequency noise is a health concern for workers in industrial workshops; rooms of highly varying size and dimensions, usually containing obstacles (the ‘fittings’). Low-frequency noise can be generated from sources such as reciprocating or rotating machinery, or ventilation systems. As the exposure time to the noise lengthens, workers are increasingly at risk to harmful effects such as hearing loss, communication difficulty, personal discomfort, and even nausea from induced body vibrations. Passive methods of noise control, such as absorption or barriers, generally perform better at high frequencies, but are inadequate at low frequencies. A proposed solution is active noise control, which relies on destructive interference of sound waves to reduce noise levels. However, this depends on phase, and how it is affected when sound waves encounter diffracting obstacles. In addition, the geometrical configuration of the active-control system must be optimized, which can be done using a prediction model. Sound-prediction models can also estimate the decibel level of sound within a given room configuration created by a source and the attenuation provided by the control system. Therefore, it is of interest to develop a model that predicts sound propagation in fitted rooms with phase. In this thesis, sound-pressure fields were investigated in rooms containing parallelepiped obstacles at low frequencies for which the wavelength is comparable to the obstacle dimensions. The geometric theory of diffraction (GTD) was used to model edge diffraction from an obstacle and, thus, the pressure field in shadow regions. A ray-tracing prediction model was improved to consider both the amplitude and phase of sound fields, and also the effects of edge diffraction. To validate the prediction model, experiments were performed in an anechoic chamber where a source and diffracting objects were located. In collaboration with Dr Valeau at the Université de Poitiers in France, a second model based on the finite element method (FEM) was used to compare prediction results. It was found that the phase depends mostly on the direct unblocked source-to-receiver distance. The FEM and experimental results showed that occluding objects cause phase shifts. The implementation of first-order diffraction into the ray-tracing program was successful in predicting shadow zones, thus producing a better prediction of realistic sound fields in rooms with obstacles.

This thesis investigates the possibilities of interleaving multiple Digital to Analog converters in a high speed environment. Algorithms for interleaving and noise shaping as well as filters are tailored for high frequency operation.In the first part of the thesis, algorithms are evaluated and models to simulate errors are created. It was concluded that DAC interleaving is feasible to reach high sample rates. Interleaving or parallelization of the $\Sigma\Delta$ noise shaper proved to not be feasible for the specific application due low oversampling and high speed operation.The second part of the thesis consists of measurements on a custom SP Devices development board. These tests confirm that interleaving of DACs works as intended and that it is possible to increase the output bandwidth beyond the one of a single DAC.