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For several decades the propagation characteristics of acoustic pulses (attenuation and sound speed) have heen inverted in attempts to measure the size distributions of gas bubbles in liquids, Primarily this has heen attempted in the ocean for defence and environmental purposes, however there are a growing number of biomedical and industrial applications. In order to simplify the inversion, previous investigators have assumed that the bubbles are undergoing linear, steady-state monochromatic pulsations in a free field, without interacting. These assumptions are always contravened to some extent. This study examines the validity of the assumptions and iclentifies the need for a new time-dependent nonlinear method of determining a bubble's extinction cross section. Such a model is developed and employed in an experiment to estimate the bubble population in the surf-zone, an important but seldom measured region of the ocean, where large populations of bubbles are generated by breaking waves. The necessary theoretical framework to exploit this new model (based on the current state-of-the-art technique) is developed and employs a new method of determining the optimal regularisation parameter for use in the inversion process. A series of laboratory tests and surf-zone sea trials are described that result in a set of bubble populations calculated using linear and, for the first time, nonlinear techniques.

This thesis describes work performed in the analysis and development of positioning algorithms for self-noise of a known kind; it also describes the development of short base-line systems capable of positioning the sources. Many studies of wild cetaceans rely on tracking the movement of wild animals, often in hostile conditions and with limited contact with these animals. Advanced technology exists for satellite or radio tracking of marine wildlife, but this relies on an animal being first caught and tagged. In situations where random interactions with marine wildlife are to be analysed, it is not practicable to attach devices to an animal, so it is appropriate to use passive techniques, in which the animal's self noise is located and tracked. Present passive systems usually include a long base-line array, which can be difficult to deploy. The problem may be overcome by reducing the array in size, but it results in an increase in positioning errors. This study attempts to quantifY these errors and looks into the practicability of short base-line passive arrays. Two systems are described here, both for tracking impulsive sounds in real time. The first is for use on pelagic trawl nets, the other as a prototype high-speed system to prove the different algorithms developed before and during this study. The prototype systems, each having a minimum of four receivers positioned in various configurations, have been tested in a tank with a controlled sound source. The source is a 'pinger', which allows the systems also to be adopted for diver positioning and tracking. A survey of unclassified literature has shown characteristics of cetacean acoustic signatures, which have been utilised in the optimisation of the systems. The physiology of cetaceans has also been reviewed to help understand the physical limitations of the systems presented.

The underwater acoustic channel is characterized by time-varying multipath propagation with large delay spreads of up to hundreds of milliseconds, which introduces severe intersymbol interference (ISI) in digital communication system. Many of the existing channel estimation and equalization techniques used in radio frequency wireless communication systems might be practically inapplicable to underwater acoustic communication due to their high computational complexity. The recursive least squares (RLS)-dichotomous coordinate descent (DCD) algorithm has been recently proposed and shown to perform closely to the classical RLS algorithm while having a significantly lower complexity. It is therefore a highly promising channel estimation algorithm for underwater acoustic communications. However, predicting the convergence performance of the RLS-DCD algorithm is an open issue. Known approaches are found not applicable, as in the RLS-DCD algorithm, the normal equations are not exactly solved at every time instant and the sign function is involved at every update of the filter weights. In this thesis, we introduce an approach for convergence analysis of the RLS-DCD algorithm based on computations with only deterministic correlation quantities. Equalization is a well known method for combatting the ISI in communication channels. Coefficients of an adaptive equalizer can be computed without explicit channel estimation using the channel output and known pilot signal. Channel-estimate (CE) based equalizers which re-compute equalizer coefficients for every update of the channel estimate, can outperform equalizers with the direct adaptation. However, the computational complexity of CE based equalizers for channels with large delay spread, such as the underwater acoustic channel, is an open issue. In this thesis, we propose a low-complexity CE based adaptive linear equalizer, which exploits DCD iterations for computation of equalizer coefficients. The proposed technique has as low complexity as O(Nu(K+M)) operations per sample, where K and M are the equalizer and channel estimator length, respectively, and Nu is the number of iterations such that Nu << K and Nu << M. Moreover, when using the RLS-DCD algorithm for channel estimation, the computation of equalizer coefficients is multiplication-free and division-free, which makes the equalizer attractive for hardware design. Simulation results show that the proposed adaptive equalizer performs close to the minimum mean-square-error (MMSE) equalizer with perfect knowledge of the channel. Decision feedback equalizers (DFEs) can outperform LEs, provided that the effect of decision errors on performance is negligible. However, the complexity of existing CE based DFEs normally grows squarely with the feedforward filter (FFF) length K. In multipath channels with large delay spread and long precursor part, such as in underwater acoustic channels, the FFF length K needs to be large enough to equalize the precursor part, and it is usual that K > M. Reducing the complexity of CE based DFEs in such scenarios is still an open issue. In this thesis, we derive two low complexity approaches for computing CE based DFE coefficients. The proposed DFEs operate together with partial-update channel estimators, such as the RLS-DCD channel estimator, and exploit complex-valued DCD iterations to efficiently compute the DFE coefficients. In the first approach, the proposed DFE has a complexity of O(Nu l log 2l) real multiplications per sample, where l is the equalizer delay and Nu is the number of iterations such that Nu << l. In the second proposed approach, DFE has a complexity as low as O(Nu K)+O(Nu B) + O(Nu M) operations per sample, where B is the feedback filter (FBF) length and Nu << M. Moreover, when the channel estimator also exploits the DCD iterations, e.g. such as in the RLS-DCD adaptive filter, the second approach is multiplication-free and division-free, which makes the equalizer attractive for hardware implementation. Simulation results show that the proposed DFEs perform close to the RLS CE based DFE, where the CE is obtained using the classical RLS adaptive filter and the equalizer coefficients are computed according to the MMSE criterion. Localization is an important problem for many underwater communication systems, such as underwater sensor networks. Due to the characteristics of the underwater acoustic channel, localization of underwater acoustic sources is challenging and needs to be accurate and computationally efficient. The matched-phase coherent broadband matched-field (MF) processor has been previously proposed and shown to outperform other advanced broadband MF processors for underwater acoustic source localization. It has been previously proposed to search the matched phases using the simulated annealing, which is well known for its ability for solving global optimization problems while having high computational complexity. This prevents simultaneous processing of many frequencies, and thus, limits the processor performance. In this thesis, we introduce a novel iterative technique based on coordinate descent optimization, the phase descent search (PDS), for searching the matched phases. We show that the PDS algorithm obtains matched phases similar to that obtained by the simulated annealing, and has significantly lower complexity. Therefore, it enables to search phases for a large number of frequencies and significantly improves the processor performance. The proposed processor is applied to experimental data for locating a moving acoustic source and shown to provide accurate localization of the source well matched to GPS measurements.

Reverberation enhancement is a technology which allows the reverberation time of a room to be increased through the use of an electronic system. These systems have traditionally been applied to improve the acoustics of large concert halls but the technology can also be used in smaller spaces with several possible applications. Previous uses of reverberation enhancement in small rooms have largely consisted of direct transplants of systems designed for large concert halls. This work investigates the complications which arise when using reverberation enhancement in a small room due to the differences in the acoustic properties of the space and also the restriction on the channel count of the system due to physical constraints. The first part of this work deals with increasing the resultant reverberation time of the room without requiring additional system channels. This is achieved through the use of processing within the system. Two methods have been investigated. The first extends the resultant reverberation time without changing the feedback gain. The processing used for this purpose is either electronic reverberation or simple delay, both of which have been shown to allow significant increases in resultant reverberation time. These changes can be predicted accurately using diffuse field theory. The other method uses time-varying processing to increase the maximum stable feedback gain. This has been shown to allow increases in resultant reverberation time but also causes undesirable artefacts which limit the usability of this technique. The second part of this work focuses on the differences in the acoustic properties of small rooms and especially the ways in which these rooms differ from a diffuse field. This includes the consideration of the modal properties of the room at low frequency which are insignificant in a large room. It has been shown that the spatial and frequency variations of the room at low frequency can be reduced through numerical optimisation of the processing within the reverberation enhancement system. Finally, the diffusion of the sound field and the early energy in the impulse response have been considered. It is shown that restrictions on the resultant reverberation time may be required in order to create a subjectively acceptable acoustic response. Overall, this work has shown that by accounting for the properties of the room, excellent performance of the system can be achieved.

Les matériaux absorbants pour l’acoustique sous-marine sont utiles pour lutter contre la détection navale. Ces matériaux doivent évoluer, notamment en termes de performance acoustique et de tenue à la pression hydrostatique, en regard des progrès accomplis dans le domaine de la détection sous-marine. Pour cela, de nouvelles technologies de matériaux sont nécessaires et font l’objet de cette thèse. Les travaux tirent profit des propriétés intéressantes observées dans les structures périodiques et s’orientent autour de deux axes principaux : d’une part les propriétés de filtrage fréquentiel obtenues avec l’utilisation de cristaux phononiques (structures périodiques), d’autre part les phénomènes d’absorption basse fréquence liés à des résonances locales. L’étude des phénomènes précités implique le développement d’outils dédiés à l’analyse des structures périodiques avec la possibilité de prendre en compte le caractère dissipatif de certains matériaux, comme ceux couramment utilisés en acoustique sous-marine. Grâce à ces outils numériques basés sur la méthode des éléments finis, une étude de matériaux à inclusions fluides ou solides est présentée. Ces recherches permettent d’identifier et d’explorer les phénomènes liés à des résonances locales. Des expérimentations ont été effectuées par l’intermédiaire de la mise en place d’un banc d’essai robuste dédié à la mesure de panneaux acoustiques immergés. Les mesures faites en cuve acoustique ont été confrontées avec succès aux prévisions numériques. Enfin, d’autres dispositifs, basés sur différents phénomènes, sont proposés dans le but de repousser la limite basse fréquence de l’absorption des ondes dans l’eau. / Underwater absorbing materials are used on naval ships to avoid being detected. Acoustic performances and hydrostatic resistance still need improvements to face the evolution of naval detection systems. Therefore, new technologies of underwater absorbing materials are studied in this work. This thesis focuses on periodical structures, considering two main topics: stop-band effect from phononic crystal and resonant inclusions designed for low frequency absorption. To study these phenomena, periodic structures analysis is done using specifically developed tools able to consider losses in viscoelastic materials. Thanks to these numerical tools based on the finite element method, a study on materials with fluid or solid inclusions is presented. Some local resonances are detected and exploited. To carry on measurements, the installation of a robust bench test for underwater acoustic panels has been implemented. Measurements and numerical results are in good agreement. Finally, other designs based on different phenomena are proposed in order to stretch the limit of low frequency underwater absorbing materials.

Tese (Doutorado) - Universidade Federal do Rio de Janeiro, Instituto Alberto Luiz Coimbra de Pós Graduação e Pesquisa de Engenharia, Rio de Janeiro, 2015. / Bibliografia: p. 178-201. / Made available in DSpace on 2018-02-14T17:42:11Z (GMT). No. of bitstreams: 0 Previous issue date: 2015. Added 1 bitstream(s) on 2019-02-13T19:12:10Z : No. of bitstreams: 1 00000f04.pdf: 6205125 bytes, checksum: 4f555c737b989e1f3b1782a8ea74e01b (MD5)

Tese (Doutorado) - Naval Postgraduate Scholl, Monterey, CA, 2017. / Devido a falta de auxílios na navegação subaquática, um sistema capaz de ajudar um veículo submersível a manter a precisão da posição, permitindo que ele permaneça subaquático por períodos mais longos, é extremamente desejável. A precisão e confiabilidade / Made available in DSpace on 2018-02-14T17:42:12Z (GMT). No. of bitstreams: 0 Previous issue date: 2017. Added 1 bitstream(s) on 2019-02-13T19:12:12Z : No. of bitstreams: 1 000015dd.pdf: 3906657 bytes, checksum: f7516e210feff674af44c4ed61eae57f (MD5)