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Barsanti, Robert J. "Denoising of ocean acoustic signals using wavelet-based techniques." Monterey, Calif. : Springfield, Va. : Naval Postgraduate School ; Available from National Technical Information Service, 1996. http://handle.dtic.mil/100.2/ADA329379.
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Thesis (M.S. in Electrical Engineering and M.S. in Engineering Acoustics) Naval Postgraduate School, December 1996.Thesis advisor(s): Monique P. Fargues and Ralph Hippenstiel. "December 1996." Includes bibliographical references (p. 99-101). Also available online.
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Yagci, Tayfun. "Target Classification And Recognition Using Underwater Acoustic Signals." Master's thesis, METU, 2005. http://etd.lib.metu.edu.tr/upload/3/12606373/index.pdf.
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Nowadays, fulfillment of the tactical operations in secrecy has great importance for especially subsurface and surface warfare platforms as a result of improvements in weapon technologies. Spreading out of the tactical operations to the larger areas has made discrimination of targets unavoidable. Due to enlargement of the weapon ranges and increasing subtle hostile threats as a result of improving technology, &ldquovisual&rdquo
target detection methods left the stage to the computerized acoustic signature detection and evaluation methods. Despite this, the research projects have not sufficiently addressed in the field of acoustic signature evaluation. This thesis work mainly investigates classification and recognition techniques with TRN / LOFAR signals, which are emitted from surface and subsurface platforms and proposes possible adaptations of existing methods that may give better results if they are used with these signals. Also a detailed comparison has been made about the experimental results with underwater acoustic signals.
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Eldred, Randy Michael. "Doppler processing of phase encoded underwater acoustic signals." Thesis, Monterey, California : Naval Postgraduate School, 1990. http://handle.dtic.mil/100.2/ADA241283.
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Thesis (M.S. in Electrical Engineering)--Naval Postgraduate School, September 1990.Thesis Advisor(s): Miller, James H. Second Reader: Tummala, Murali. "September 1990." Description based on title screen as viewed on December 17, 2009. DTIC Identifier(s): Acoustic tomography, inverse problems, Fast Hadamard Transforms, theses. Author(s) subject terms: Acoustic tomography, Fast Hadamard Transform, maximal-length sequences, Doppler processing. Includes bibliographical references (p. 95-96). Also available in print.
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Bissinger, Brett Bose N. K. Culver R. Lee. "Minimum hellinger distance classification of underwater acoustic signals." [University Park, Pa.] : Pennsylvania State University, 2009. http://etda.libraries.psu.edu/theses/approved/WorldWideIndex/ETD-4677/index.html.
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Jack, Susan Heather. "The investigation of underwater acoustic signals using Laser Doppler Anemometry." Thesis, University of Edinburgh, 2000. http://hdl.handle.net/1842/15088.
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Laser Doppler Anemometry (LDA) has been used to study underwater acoustic signals both from emitting hydrophones and underwater explosions. A dual-beam LDA arrangement was used to capture Doppler signals arising from light scattered from particles suspended at the point of interest in the flow. These Doppler signals are analysed using either Hilbert transforms or wavelets, both of which allow instantaneous frequency information to be obtained. When an acoustic signal propagates through a medium it creates refractive index variations within the medium. The apparent motion of the scattering particles, as observed by the detector, which give rise to the Doppler signal, is therefore made up of two components. Firstly, the particles oscillate due to the sound field and secondly, the interference fringes oscillate due to the refractive index variations. This is termed the acousto-optic effect. A theory has been developed to investigate the effect of these refractive index variations on the analysed Doppler signals of an LDA system. Analysis of experimental Doppler signals using the Hilbert transform technique shows close agreement with the theoretical predictions. LDA has also been used to investigate the acoustic signal emitted by an oscillating explosion bubble. This is generated by an underwater spark which creates a similar situation to an underwater explosion in which a shock wave and an oscillating bubble are produced. Analysis of the Doppler signal using wavelets provides information on the bubble period, radius, energy and particle velocity. Explosive materials have traditionally been used for investigation of underwater explosions but they have the disadvantage of obscuring the area with explosion debris thus making optical investigation difficult. It is shown in this work that the use of LDA and analysis of Doppler signals using wavelets is an accurate technique for the investigation of acoustic signals from underwater explosions. This allows investigation of the area close to the explosion centre where measurements have been difficult to achieve with traditional techniques.
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Kendall, Elizabeth Ann Caughey Thomas Kirk. "Range dependent signals and maximum entropy methods for underwater acoustic tomography /." Diss., Pasadena, Calif. : California Institute of Technology, 1985. http://resolver.caltech.edu/CaltechETD:etd-04092008-080843.
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Sanderson, Josh. "Hierarchical Modulation Detection of Underwater Acoustic Communication Signals Through Maximum Likelihood Combining." Wright State University / OhioLINK, 2014. http://rave.ohiolink.edu/etdc/view?acc_num=wright1410872323.
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Heaney, Kevin Donn. "Inverting for source location and internal wave strength using long range ocean acoustic signals /." Diss., Connect to a 24 p. preview or request complete full text in PDF format. Access restricted to UC campuses, 1997. http://wwwlib.umi.com/cr/ucsd/fullcit?p9737384.
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Evans, Benjamin Kerbin. "The effect of coded signals on the precision of autonomous underwater vehicle acoustic navigation." Thesis, Massachusetts Institute of Technology, 1999. http://hdl.handle.net/1721.1/29044.
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Thesis (Ocean E.)--Joint program in Oceanography/Applied Ocean Science and Engineering (Massachusetts Institute of Technology, Dept. of Ocean Engineering and Woods Hole Oceanographic Institution), 1999.Includes bibliographical references (p. 127-128).
Acoustic coded signaling offers potentially significant improvements over traditional "toneburst" methods in many underwater applications where error due to noise and multipath interference is a problem. In this thesis, the use of these spread spectrum techniques is analyzed for navigation of the REMUS autonomous underwater vehicle. The accuracy of the current system using Turyn and Barker sequences, as well as toneburst, is quantified, and the sources of the remaining error are examined.
by Benjamin Kerbin Evans.
Ocean E.
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Blount, Richard J. Jr. "Underwater acoustic model-based signal processing applied to the detection of signals from a planar array of point source elements." Thesis, New York : Kluwer Academic/Plenum Publishers, 1985. http://hdl.handle.net/10945/21597.
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Galindo, Romero Marta. "Spatial Variations in the Acoustic Peak Pressure of Impulsive Low Frequency Anthropogenic Signals in Underwater Marine Environments." Thesis, Curtin University, 2017. http://hdl.handle.net/20.500.11937/59661.
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A method to predict spatial variations in the peak pressure level of impulsive low frequency anthropogenic signal propagating in marine environments is presented. The method is based on the correlation between the peak pressure level and the sound exposure level, and the application of extreme value theory to estimate fluctuations of the peak pressure around its mean value in varying ocean environments. The method was examined using signals from offshore seismic surveys and pile driving.
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Sobreira, Filipa Alexandra Veiga. "Effects of vessel noise on underwater vocalizations of bottlenose dolphins, Tursiops truncatus, in the Sado Estuary." Master's thesis, Universidade de Aveiro, 2017. http://hdl.handle.net/10773/21964.
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Mestrado em Biologia MarinhaMaritime traffic is an important source of disturbance for coastal cetaceans, especially for local and resident populations, like the bottlenose dolphins (Tursiops truncatus) population in the Sado estuary. Vessel noise might mask important signals such as communication calls. To compensate masking effects, animals may change their vocal behavior by shifting vocal rate, call intensity, call type, call frequency and duration. To evaluate the potential impacts on the acoustic behavior of this population, abundance and acoustic characteristics of whistles, echolocation signals and burst-pulsed sounds were analyzed in relation to boat traffic. The samples used were obtained in field recordings of dolphin vocalizations made from March 2014 to April 2017. Boat traffic operating within a 1000 m radius was listed as absent or present. Vocal elements were classified according to visual graphical and aural characteristics in: whistles, slow-click trains, short-burst pulses, creaks, squawks, variable rate click trains, bangs, gulps, squeaks and grunts. Analysis of emission rates was based on the number of recognizable units per minute for all vocal elements. In the presence of vessels, differences in call rates were not significant for all types of vocal elements. For selected vocal elements, different acoustic parameters were examined, using a nonparametric MANOVA, and modifications between vessel presence and absence were found for the following vocal elements: whistles (X2 (7) = 56.42; N = 620; p < 0.001), creaks (X2 (8) = 19.53; N = 94; p = 0.012), grunts (X2 (8) = 80.968; N = 339; p < 0.001), gulps (X2 (7) = 58.76; N = 260; p < 0.001) and squeaks (X2 (10) = 25.894; N = 121; p = 0.004)). These results show modifications in acoustic behavior in the presence of vessels, suggesting that bottlenose dolphins in this population might adjust their vocal frequencies and produce shorter signals to maintain communication. This study shows that although resident bottlenose dolphins in Sado estuary seem to display some tolerance to the noise generated from boats in their habitat, it probably causes significant changes in their communication behaviors.
O tráfego marítimo é uma fonte de perturbação importante para os cetáceos costeiros, especialmente para populações locais e residentes, como a população de golfinhos-roazes (Tursiops truncatus) no estuário do Sado. O ruído provocado por embarcações pode mascarar sinais importantes, como os de comunicação. Para compensar os efeitos de mascaramento, os animais podem modificar o seu comportamento acústico alterando a taxa de emissão, a intensidade do sinal, o tipo de vocalização, a frequência e a duração dos sinais acústicos. Para avaliar os potenciais impactos no comportamento acústico desta população, foram analisados a abundância e as características acústicas dos assobios, dos sinais de ecolocalização e dos sons pulsados em relação ao tráfego de embarcações. As amostras utilizadas foram obtidas através de gravações subaquáticas realizadas na região do estuário do Sado, feitas de março de 2014 a abril de 2017. Os barcos foram considerados como presentes ou ausentes, tendo em conta um raio de 1000 m. Os elementos vocais foram classificados de acordo com as características visuais gráficas e auditivas em: assobios, trens de cliques, buzzes, rangidos, chorincos, trens de taxa variável, bangs, goles, guinchos e grunhidos. A análise das taxas de emissão baseou-se no número de unidades reconhecíveis por minuto para todos os elementos vocais. Na presença de embarcações, não existiram diferenças significativas para as taxas de emissão de todos os tipos de elementos vocais. Para elementos vocais selecionados, foram examinados diferentes parâmetros acústicos, utilizando uma MANOVA não paramétrica, e foram encontradas alterações entre presença e ausência de navios, para as seguintes vocalizações: assobios (X2 (7) = 56,42; N = 620; p <0,001), rangidos (X2 (8) = 19,53; N = 94; p = 0,012), grunhidos (X2 (8) = 80,968; N = 339; p <0,001), goles (X2 (7) = 58,76; N = 260; p < 0,001) e guinchos (X2 (10) = 25,894; N = 121; p = 0,004)). Estes resultados mostram modificações no comportamento acústico na presença de embarcações, revelando que os golfinhos-roazes desta população poderão ajustar as suas frequências vocais e produzir sinais mais curtos para manter a comunicação. Este estudo sugere que, embora os golfinhos-roazes residentes do estuário do Sado possam apresentar alguma tolerância ao ruído gerado por barcos no seu habitat, este provavelmente causa mudanças significativas nos seus comportamentos de comunicação.
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Malfante, Marielle. "Automatic classification of natural signals for environmental monitoring." Thesis, Université Grenoble Alpes (ComUE), 2018. http://www.theses.fr/2018GREAU025/document.
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Ce manuscrit de thèse résume trois ans de travaux sur l’utilisation des méthodes d’apprentissage statistique pour l’analyse automatique de signaux naturels. L’objectif principal est de présenter des outils efficaces et opérationnels pour l’analyse de signaux environnementaux, en vue de mieux connaitre et comprendre l’environnement considéré. On se concentre en particulier sur les tâches de détection et de classification automatique d’événements naturels.Dans cette thèse, deux outils basés sur l’apprentissage supervisé (Support Vector Machine et Random Forest) sont présentés pour (i) la classification automatique d’événements, et (ii) pour la détection et classification automatique d’événements. La robustesse des approches proposées résulte de l’espace des descripteurs dans lequel sont représentés les signaux. Les enregistrements y sont en effet décrits dans plusieurs espaces: temporel, fréquentiel et quéfrentiel. Une comparaison avec des descripteurs issus de réseaux de neurones convolutionnels (Deep Learning) est également proposée, et favorise les descripteurs issus de la physique au détriment des approches basées sur l’apprentissage profond.Les outils proposés au cours de cette thèse sont testés et validés sur des enregistrements in situ de deux environnements différents : (i) milieux marins et (ii) zones volcaniques. La première application s’intéresse aux signaux acoustiques pour la surveillance des zones sous-marines côtières : les enregistrements continus sont automatiquement analysés pour détecter et classifier les différents sons de poissons. Une périodicité quotidienne est mise en évidence. La seconde application vise la surveillance volcanique : l’architecture proposée classifie automatiquement les événements sismiques en plusieurs catégories, associées à diverses activités du volcan. L’étude est menée sur 6 ans de données volcano-sismiques enregistrées sur le volcan Ubinas (Pérou). L’analyse automatique a en particulier permis d’identifier des erreurs de classification faites dans l’analyse manuelle originale. L’architecture pour la classification automatique d’événements volcano-sismiques a également été déployée et testée en observatoire en Indonésie pour la surveillance du volcan Mérapi. Les outils développés au cours de cette thèse sont rassemblés dans le module Architecture d’Analyse Automatique (AAA), disponible en libre accèsThis manuscript summarizes a three years work addressing the use of machine learning for the automatic analysis of natural signals. The main goal of this PhD is to produce efficient and operative frameworks for the analysis of environmental signals, in order to gather knowledge and better understand the considered environment. Particularly, we focus on the automatic tasks of detection and classification of natural events.This thesis proposes two tools based on supervised machine learning (Support Vector Machine, Random Forest) for (i) the automatic classification of events and (ii) the automatic detection and classification of events. The success of the proposed approaches lies in the feature space used to represent the signals. This relies on a detailed description of the raw acquisitions in various domains: temporal, spectral and cepstral. A comparison with features extracted using convolutional neural networks (deep learning) is also made, and favours the physical features to the use of deep learning methods to represent transient signals.The proposed tools are tested and validated on real world acquisitions from different environments: (i) underwater and (ii) volcanic areas. The first application considered in this thesis is devoted to the monitoring of coastal underwater areas using acoustic signals: continuous recordings are analysed to automatically detect and classify fish sounds. A day to day pattern in the fish behaviour is revealed. The second application targets volcanoes monitoring: the proposed system classifies seismic events into categories, which can be associated to different phases of the internal activity of volcanoes. The study is conducted on six years of volcano-seismic data recorded on Ubinas volcano (Peru). In particular, the outcomes of the proposed automatic classification system helped in the discovery of misclassifications in the manual annotation of the recordings. In addition, the proposed automatic classification framework of volcano-seismic signals has been deployed and tested in Indonesia for the monitoring of Mount Merapi. The software implementation of the framework developed in this thesis has been collected in the Automatic Analysis Architecture (AAA) package and is freely available
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Jung, Du San. "Detection of binary phase-shift keying signal in multioath propagation." Thesis, Monterey, Calif. : Springfield, Va. : Naval Postgraduate School ; Available from National Technical Information Service, 2002. http://library.nps.navy.mil/uhtbin/hyperion-image/02Jun%5FJung.pdf.
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Dessalermos, Spyridon. "Undersea acoustic propagation channel estimation." Thesis, Monterey, Calif. : Springfield, Va. : Naval Postgraduate School ; Available from National Technical Information Service, 2005. http://library.nps.navy.mil/uhtbin/hyperion/05Jun%5FDessalermos.pdf.
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Thesis (M.S. in Electrical Engineering and M.S. in Applied Physics)--Naval Postgraduate School, June 2005.Thesis Advisor(s): Joseph Rice, Roberto Cristi. Includes bibliographical references (p. 117-119). Also available online.
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Bienvenu, Kirk Jr. "Underwater Acoustic Signal Analysis Toolkit." ScholarWorks@UNO, 2017. https://scholarworks.uno.edu/td/2398.
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This project started early in the summer of 2016 when it became evident there was a need for an effective and efficient signal analysis toolkit for the Littoral Acoustic Demonstration Center Gulf Ecological Monitoring and Modeling (LADC-GEMM) Research Consortium. LADC-GEMM collected underwater acoustic data in the northern Gulf of Mexico during the summer of 2015 using Environmental Acoustic Recording Systems (EARS) buoys. Much of the visualization of data was handled through short scripts and executed through terminal commands, each time requiring the data to be loaded into memory and parameters to be fed through arguments. The vision was to develop a graphical user interface (GUI) that would increase the productivity of manual signal analysis. It has been expanded to make several calculations autonomously for cataloging and meta data storage of whale clicks. Over the last year and a half, a working prototype has been developed with MathWorks matrix laboratory (MATLAB), an integrated development environment (IDE). The prototype is now very modular and can accept new tools relatively quickly when development is completed. The program has been named Banshee, as the mythical creatures are known to “wail”. This paper outlines the functionality of the GUI, explains the benefits of frequency analysis, the physical models that facilitate these analytics, and the mathematics performed to achieve these models.
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Lindgren, Jakob. "Software defined acoustic underwater modem." Thesis, Mälardalens högskola, Akademin för innovation, design och teknik, 2011. http://urn.kb.se/resolve?urn=urn:nbn:se:mdh:diva-12170.
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Today many types of communication are employed on seagoing vessels, such as radio, satellite and Wi-Fi but only one type of communication is practical for submerged vessels, the acoustic underwater modem. The "off-the-shelf" modems are sometimes difficult to update and replace, especially on a large submarine. But by separating the hardware from the signal processing and making the software modular more versatility can be achieved. The questions that this thesis are asking are: is it possible to implement the signal processing in software? How small or large should the modules be? What kind of architecture should be used? This thesis shows that it is indeed possible to implement simple algorithms that can isolate a signal and read its content regardless of the hardware configuration. Calculations show that up to 13 kbps can be reached at a range of one kilometer. It is most practical to make the entire physical layer into one module and the size of the system could drastically change the type of architecture used.
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Adams, Mark D. "Signal detection optimization for underwater acoustics." Master's thesis, This resource online, 1990. http://scholar.lib.vt.edu/theses/available/etd-02022010-020009/.
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Weintroub, Jonathan. "Fish stock assessment by a statistical analysis of echo sounder signals." Master's thesis, University of Cape Town, 1986. http://hdl.handle.net/11427/19509.
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A means of assessing the quantity of exploitable fish in the sea is a requirement for effective management of the resource. Sonar is widely used in this regard, as it provides a rapid means of assessment. Two acoustic assessment techniques currently used are the echo counting and echo integration· methods. The echo counting method requires that only single fish echoes are present in the backscatter from the shoal, while the echo integration technique requires an a-priori knowledge of the average target strength of the fish in the shoal. A novel method of assessment has been proposed. It relies on the relationship between the statistics of the backscatter from a volume distribution of scatterers and the number of scatterers contributing to the backscatter at any one time. The attraction of the method when applied to the estimation of number density of fish, is that estimates can be produced in the presence of overlapping echoes, and that knowledge of the target strength of the fish is unnecessary. The application of this method to acoustic fish stock assessment is investigated in this work. Current methods of assessment are reviewed and the theory of the statistical method is given. A computer simulation of the scattering problem gives a useful insight into the effects of sample size and density on the accuracy of the method. The method has been applied to the assessment of fish at sea, where it was run in tandem with an echo integrator. The results obtained with the two techniques are compared. Reasons for discrepancies are proposed and problems in the application of the method are identified.
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Walden, Alan Keith. "Signal processing techniques on an underwater acoustic projector." Thesis, Georgia Institute of Technology, 1991. http://hdl.handle.net/1853/17336.
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Chen, Teyan. "Novel adaptive signal processing techniques for underwater acoustic communications." Thesis, University of York, 2011. http://etheses.whiterose.ac.uk/1925/.
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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.
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Liu, Chunshan. "Advanced signal processing techniques for underwater acoustic communication networks." Thesis, University of York, 2011. http://etheses.whiterose.ac.uk/2102/.
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In this thesis, we develop and investigate novel signal processing techniques for underwater acoustic communication networks. Underwater acoustic channels differ from radio communication channels in the lower speed of signal propagation, richer and often sparse multipath arrivals, and more severe Doppler effect. Therefore, many signal processing techniques developed for radio communications may not work equivalently well for underwater acoustic channels. To investigate signal processing techniques in underwater acoustics, efficient simulation of signal transmission is required. Specifically, there is requirement for accurate simulation of doubly-selective underwater channels for different acoustic environments. In this thesis, a low-complexity channel simulator has been developed for scenarios with moving transmitter/receiver. The simulator is based on efficient generation of time-varying channel impulse response obtained using interpolation over a set of waymark impulse responses for a relatively small number of sampling points on the transmitter/receiver trajectory. The waymark impulse responses are generated using an acoustic field computation method, which is the most computationally expensive part of the simulator. To reduce the trajectory sampling rate, and thus, to reduce the complexity of the field computation, an approach for adjusting the time-varying multipath delays has been developed. For setting the trajectory sampling interval, a simple rule has been proposed, based on the waveguide invariant theory. To further reduce the simulator complexity, local spline interpolation is exploited. The developed simulator has been verified by comparing the simulated data with data from real ocean experiments. In particular, applying simulated data to an OFDM modem shows similar performance with that obtained from the data of a deep water experiment. In communication networks, knowledge of positions of communication nodes is important for improving the system performance. A multi-source localization technique has been proposed based on the matched field (MF) processing. The technique locates the nodes by solving a set of basis pursuit de-noising (BPDN) problems corresponding to a set of source frequencies. An efficient technique combining the homotopy approach and coordinate descent search has been developed to solve the BPDN problem. Further reduction in the complexity has been achieved by applying a position grid refinement method. Verified using simulated data generated by the proposed simulator and data from real experiment, the proposed technique outperforms other MF techniques in resolving sources positioned closely to each other, tolerance to noise and capability of locating multiple sources. To provide reliable localization based on MF techniques, accurate knowledge of the underwater acoustic environment is essential. However, such knowledge is not always available. Estimating uncertain environmental parameters can be achieved using MF inversion techniques. This requires solving a global optimization problem. Several global optimization algorithms have been investigated and an algorithm combining the simulated annealing and downhill simplex method has been applied for estimating the sound speed profile in a deep water scenario. Accurate MF localization results have been demonstrated when using the estimated sound speed profile. A very important task of communication receivers is accurate channel estimation. The knowledge of node positions and the environment can be exploited for enhancing the channel estimation accuracy and reducing the estimation complexity. This knowledge can be used to define the structure of the channel impulse response, such as the multipath spread and the sparsity. A channel estimator exploiting the channel sparsity estimated from the node positions has been proposed and investigated. The sparse taps of the channel impulse response are identified by solving a BPDN problem. The estimator employs an iterative tap-by-tap processing and uses local splines to interpolate the time-varying tap coefficients. This allows reduction in the complexity and memory requirement, whereas providing a high estimation accuracy.
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Sarikaya, Tevfik Bahadir. "A Comparative Analysis Of Matched Field Processors For Underwater Acoustic Source Localization." Master's thesis, METU, 2010. http://etd.lib.metu.edu.tr/upload/12612578/index.pdf.
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In this thesis, localization of the underwater sound sources using matched field processing technique is considered. Localization of the underwater sound sources is one of the most important problems encountered in underwater acoustics and signal processing. Many techniques were developed to localize sources in range, depth and bearing angle. However, most of these techniques do not consider or only slightly takes into
account the environmental factors that dramatically effect the propagation of underwater sound. Matched field processing has been developed as a technique that fully considers the environmental factors. Matched field processing has proven to be successful in many applications such as localization of sources in range and depth, the determination of environmental parameters, and the evaluation of model accuracies. In this study, first a comparative analysis of narrowband matched field processors is given. Namely four main processors: Bartlett processor, Minimum Variance Distortionless Response (MVDR) processor, MVDR with neighboring location constraints and MVDR with environmental perturbation constraints are compared in terms of their probability of correct localization under certain environmental conditions. Secondly, a performance assesment for the most common broadband matched field processors
is made. The correct localization performances for incoherent broadband matched field processor, Tolstoy/Michalopoulo's coherent matched field processor and broadband matched field processor with environmental perturbation constraints is given for certain environmental conditions. Finally, a new weighting approach to combine data for broadband matched field processing is introduced. The fact that information from different frequencies may have different reliability depending on the environmental conditions is considered to develop a weighting scheme. It is shown that a performance gain compared to existing processors can be achieved by using the weighting scheme introduced in this study.
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Jai, Wun Hoa Arthur. "Underwater acoustic signal behavior prediction in the region of Kauai Island." Thesis, Massachusetts Institute of Technology, 2004. http://hdl.handle.net/1721.1/33582.
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Thesis (S.M.)--Massachusetts Institute of Technology, Dept. of Ocean Engineering; and, (S.M.)--Massachusetts Institute of Technology, Dept. of Mechanical Engineering, 2004.Includes bibliographical references (p. 175-176).
Behavior of underwater sound propagation over long-ranges has been studied for several decades. The purpose of this is to describe sound propagation phenomena in various ocean environments. The key to understanding and visualizing is mathematical modeling. In the ocean acoustics community, four major mathematical techniques have been commonly used to model behavior of acoustic signal in the ocean environment. And they can be categorized into two different fields, range-independent and range-dependent. The accuracy of each method is depends on the environment characteristics. Since the propagating signal can be characterized through the mathematical modeling, it is then possible to use the propagating signal to perform beamforming and determine the characteristic of beam output.
by Wun Hoa Arthur Jai.
S.M.
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Pistacchio, David J. "Source/receiver motion-induced Doppler influence on the bandwidth of sinusoidal signals." Thesis, Monterey, Calif. : Springfield, Va. : Naval Postgraduate School ; Available from National Technical Information Service, 2003. http://library.nps.navy.mil/uhtbin/hyperion-image/03Dec%5FPistacchio.pdf.
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Thesis (M.S. in Engineering Acoustics)--Naval Postgraduate School, December 2003.Thesis advisor(s): Kevin Smith, Roy Streit. Includes bibliographical references (p. 95-100). Also available online.
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Ooppakaew, Wichian. "Advanced signal processing techniques for underwater acoustic transmission using steerable transducer arrays." Thesis, Northumbria University, 2012. http://nrl.northumbria.ac.uk/11371/.
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The main objective of this research is to design and implement an eight-hydrophone transmitter array for generating bipolar acoustic pulses mimicking those produced by cosmogenic neutrino interaction in sea water. In addition, the research was conducted as part of the ACoRNE collaboration. The work initially investigated a single hydrophone system. Due to the nature of hydrophone, the acoustic output signal does not precisely follow a given driving voltage input. Hence signal processing techniques and hydrophone modelling were applied. A bipolar acoustic generation module was built using 8-bit PIC microcontrollers for processing and control. A NI USB-6211 National Instruments commercial module was used for validation of results. The modelling was compared to experimental data generated in a water tank, showing excellent agreement. This single hydrophone instrument was deployed at the Rona array in 2008. Both 10 kHz and 23 kHz pulses were injected, whilst seven hydrophones at Rona site were chosen as the receiver hydrophone array. Signal processing techniques were applied to identify these pulses. The result showed that the triggered pulses can be detected and identified at Rona over a distance of a few hundred metres. A model for an eight-hydrophone transmission linear array system for the ANTARES site was developed. The simulation showed that the eight hydrophones arranged over an eight-metre spacing structure can mimic the anticipated pancake behaviour predicted from neutrino-induced showers as well as generating the acoustic bipolar pulse shape of sufficient amplitude for detection at ANTARES. An eight-channel arbitrary waveform generator module was designed and built using 16-bit dsPIC microcontrollers. Signal Processing techniques were again applied to calibrate the hydrophone transmitter array. The behaviour of an acoustic transducer array was examined in a laboratory water tank to study the shape and direction of such a signal in water. The results were validated against a PXI-6713 commercial module. Excellent agreement was achieved. Finally, the system was deployed at the ANTARES site in September 2011. A range of test signals including 23 kHz bipolar pulses, sine signals and orthogonal signals were injected into seawater to simulate neutrino interactions and investigate signal coding. Signal processing techniques were applied to the data deployed in order to recognise the signals emitted. However, the vessel was far away from the position planned (c 1km), hence the signal received was too weak and no signal was detected. However, the deployed data is still very useful in order to study the noise background of seawater and much has been learned for future sea campaigns.
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Trulsson, Felix. "A polynomial phase model for estimation of underwater acoustic channels using superimposed pilots." Thesis, Luleå tekniska universitet, Institutionen för system- och rymdteknik, 2019. http://urn.kb.se/resolve?urn=urn:nbn:se:ltu:diva-74460.
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In underwater acoustic communications the time variation in the channel is a huge chal- lenge. The estimation of the impulse response at the receiver is crucial for the decoding of the signal to become accurate. One way is to transmit a superimposed pilot sequence along the unknown message, and by the knowledge of the sequence have the possibility to continuously track the variation in the channel over time. This thesis investigates if it is possible by the aid of superimposed pilot sequences to separate the taps in the channel impulse response and using a parametric method to describe the taps as polynomial phase signals. The method used for separation of the taps was a moving least squares estimator. Thereafter each tap was optimised to a polynomial phase signal (PPS) using a weighted non-linear least squares estimator. The non-linear parameters of the model was then determined with the Levenberg-Marquardt method. The performance of the method was evaluated both for simulated data as well as for data from eld tests. The performance was determined by calculating the mean squared error (MSE) of the model over dierent frame lengths, signal to noise ratio (SNR), weights for the superimposed pilots, rapidness of time variation and impulse response lengths. The method was not sensitive to the properties of the channel. Even though the model had high performance, the complexity of the computations generated long compilation times. Hence, the method needs further work before a real time implementation could be possible.
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Pierce, Robert S. "Signal enhancement of laser generated ultrasound for non-destructive testing." Thesis, Georgia Institute of Technology, 1992. http://hdl.handle.net/1853/18395.
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Novaes, Marcos (Marcos Nogueira). "Multiresolution Signal Cross-correlation." Thesis, University of North Texas, 1994. https://digital.library.unt.edu/ark:/67531/metadc277645/.
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Signal Correlation is a digital signal processing technique which has a wide variety of applications, ranging from geophysical exploration to acoustic signal enhancements, or beamforming. This dissertation will consider this technique in an underwater acoustics perspective, but the algorithms illustrated here can be readily applied to other areas. Although beamforming techniques have been studied for the past fifty years, modern beamforming systems still have difficulty in operating in noisy environments, especially in shallow water.
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Riley, H. Bryan. "Matched-field source detection and localization in high noise environments a novel reduced-rank signal processing approach." Ohio : Ohio University, 1994. http://www.ohiolink.edu/etd/view.cgi?ohiou1173982711.
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Rosario, Alexander Alvarez. "Sistema para monitoramento e análise de paisagens acústicas submarinas." Universidade de São Paulo, 2015. http://www.teses.usp.br/teses/disponiveis/3/3151/tde-14062016-103939/.
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O Monitoramento Acústico Passivo (PAM) submarino refere-se ao uso de sistemas de escuta e gravação subaquática, com o intuito de detectar, monitorar e identificar fontes sonoras através das ondas de pressão que elas produzem. Se diz que é passivo já que tais sistemas unicamente ouvem, sem perturbam o meio ambiente acústico existente, diferentemente de ativos, como os sonares. O PAM submarino tem diversas áreas de aplicação, como em sistemas de vigilância militar, seguridade portuária, monitoramento ambiental, desenvolvimento de índices de densidade populacional de espécies, identificação de espécies, etc. Tecnologia nacional nesta área é praticamente inexistente apesar da sua importância. Neste contexto, o presente trabalho visa contribuir com o desenvolvimento de tecnologia nacional no tema através da concepção, construção e operação de equipamento autônomo de PAM e de métodos de processamento de sinais para detecção automatizada de eventos acústicos submarinos. Foi desenvolvido um equipamento, nomeado OceanPod, que possui características como baixo custo de fabrica¸c~ao, flexibilidade e facilidade de configuração e uso, voltado para a pesquisa científica, industrial e para controle ambiental. Vários protótipos desse equipamento foram construídos e utilizados em missões no mar. Essas jornadas de monitoramento permitiram iniciar a criação de um banco de dados acústico, o qual permitiu fornecer a matéria prima para o teste de detectores de eventos acústicos automatizados e em tempo real. Adicionalmente também é proposto um novo método de detecção-identificação de eventos acústicos, baseado em análise estatística da representação tempo-frequência dos sinais acústicos. Este novo método foi testado na detecção de cetáceos, presentes no banco de dados gerado pelas missões de monitoramento.Passive Acoustic Monitoring (PAM) refers to the use of systems to listen and record underwater soundscape, in order to detect, track and identify sound sources through the pressure waves that they produce. It is said to be passive as these systems only hear, not put noise in the environment, such as sonars. Underwater PAM has various application areas, such as military surveillance systems, port security, environmental monitoring, development of population density rates of species, species identification, etc. National technology in the field is practically nonexistent despite its importance. In this context, this paper aims to contribute to the national technology development in the field by designing, building, and operating a self-contained PAM equipment, also developing signal-processing methods for automated detection of underwater acoustic events. A device, named \"OceanPod\"which has characteristics such as low manufacturing cost, flexibility and ease of setup and use, intended for scientific, industrial research and environmental control was developed. Several prototypes of the equipment were built and used in several missions at seawaters. These missions monitoring, enabled start creating an acoustic database, which provided the raw material for the automated acoustic events detectors and realtime test. Additionally, it is also proposed a new method of detecting, identifying sound events, based on statistical analysis of the time-frequency representation of the acoustic signals. This new method has been tested in the detection of cetaceans present in the database generated by missions monitoring.
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Gieleghem, Ryan Thomas. "Robust acoustic signal detection and synchronization in a time varying ocean environment." Thesis, Massachusetts Institute of Technology, 2012. http://hdl.handle.net/1721.1/78186.
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Thesis (S.M.)--Joint Program in Applied Ocean Science and Engineering (Massachusetts Institute of Technology, Dept. of Mechanical Engineering; and the Woods Hole Oceanographic Institution), 2012.Cataloged from PDF version of thesis.
Includes bibliographical references (p. 99-100).
Signal detection and synchronization in the time varying ocean environment is a difficult endeavor. The current common methods include using a linear frequency modulated chirped pulse or maximal length sequence as a detection pulse, then match filtering to that signal. In higher signal to noise ratio (SNR) environments (- 0 dB and higher) this has been a suitable solution. As the SNR drops lower however, this solution no longer provides an acceptable probability of detection for a given tolerable probability of false alarm. The issue derives from the inherent coherence issues in the ocean environment which limit the useful matched filter length. This thesis proposes an alternative method of detection based on a recursive least squares linearly adaptive equalizer which we term the Adaptive Linear Equalizer Detector (ALED). This detectors performance has demonstrated reliable probability of detection with minimal interfering false alarms with SNR as low as -20 dB. Additionally this thesis puts forth a computationally feasible method for implementing the detector.
by Ryan Thomas Gieleghem.
S.M.
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Conn, Rebecca M. "Underwater source localization with a generalized likelihood ratio processor." Ohio : Ohio University, 1994. http://www.ohiolink.edu/etd/view.cgi?ohiou1176842869.
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Caley, Michael Stephen. "Development of a dynamic underwater acoustic communication channel simulator with configurable sea surface parameters to explore time-varying signal distortion." Thesis, Curtin University, 2016. http://hdl.handle.net/20.500.11937/1105.
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A wide-band phase-coherent multi-path underwater acoustic channel simulation is developed using an approximate quantitative model of the acoustic wave response to a time-varying three-dimensional rough surface. It has been demonstrated over transmission ranges from 100 m to 8 km by experimental channel probing and comparable synthetic replication of the channel probing through the simulated channel, that the simulation is capable of reproducing fine-time-scale Doppler and delay distortions consistent with those generated in real shallow channels.
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Boyle, John K. "Performance Metrics for Depth-based Signal Separation Using Deep Vertical Line Arrays." PDXScholar, 2015. https://pdxscholar.library.pdx.edu/open_access_etds/2198.
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Vertical line arrays (VLAs) deployed below the critical depth in the deep ocean can exploit reliable acoustic path (RAP) propagation, which provides low transmission loss (TL) for targets at moderate ranges, and increased TL for distant interferers. However, sound from nearby surface interferers also undergoes RAP propagation, and without horizontal aperture, a VLA cannot separate these interferers from submerged targets. A recent publication by McCargar and Zurk (2013) addressed this issue, presenting a transform-based method for passive, depth-based separation of signals received on deep VLAs based on the depth-dependent modulation caused by the interference between the direct and surface-reflected acoustic arrivals. This thesis expands on that work by quantifying the transform-based depth estimation method performance in terms of the resolution and ambiguity in the depth estimate. Then, the depth discrimination performance is quantified in terms of the number of VLA elements.
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Muzi, Lanfranco. "Advances in Autonomous-Underwater-Vehicle Based Passive Bottom-Loss Estimation by Processing of Marine Ambient Noise." PDXScholar, 2015. https://pdxscholar.library.pdx.edu/open_access_etds/2612.
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Accurate modeling of acoustic propagation in the ocean waveguide is important to SONAR-performance prediction, and requires, particularly in shallow water environments, characterizing the bottom reflection loss with a precision that databank-based modeling cannot achieve. Recent advances in the technology of autonomous underwater vehicles (AUV) make it possible to envision a survey system for seabed characterization composed of a short array mounted on a small AUV. The bottom power reflection coefficient (and the related reflection loss) can be estimated passively by beamforming the naturally occurring marine ambient-noise acoustic field recorded by a vertical line array of hydrophones. However, the reduced array lengths required by small AUV deployment can hinder the process, due to the inherently poor angular resolution. In this dissertation, original data-processing techniques are presented which, by introducing into the processing chain knowledge derived from physics, can improve the performance of short arrays in this particular task. Particularly, the analysis of a model of the ambient-noise spatial coherence function leads to the development of a new proof of the result at the basis of the bottom reflection-loss estimation technique. The proof highlights some shortcomings inherent in the beamforming operation so far used in this technique. A different algorithm is then proposed, which removes the problem achieving improved performance. Furthermore, another technique is presented that uses data from higher frequencies to estimate the noise spatial coherence function at a lower frequency, for sensor spacing values beyond the physical length of the array. By "synthesizing" a longer array, the angular resolution of the bottom-loss estimate can be improved, often making use of data at frequencies above the array design frequency, otherwise not utilized for beamforming. The proposed algorithms are demonstrated both in simulation and on real data acquired during several experimental campaigns.
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Stylemans, Eric. "Etude d'un système de contre-mesure électroacoustique anti sous-marin destiné à la protection des navires." Valenciennes, 1997. http://www.theses.fr/1997VALE0007.
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Pour perturber les traitements du sous-marin pendant la phase de filoguidage d'une torpille et permettre au bâtiment de surface la mise en place d'une tactique efficace de réaction, un nouveau type de contre-mesure dédiée au leurrage-brouillage spécifique des senseurs du lanceur, utilisée en complément des contre-mesures anti torpilles, est indispensable. La bande d'écoute des sous-marins et des bâtiments de surface étant en partie commune, la conception d'un leurre brouilleur dans cette bande pose un problème opérationnel car toute action visant le lanceur va également perturber le porteur dans sa détection de la menace torpille. Deux nouveaux concepts de contre-mesures ont été étudiés dans ce but: * utiliser un leurre-bouilleur omnidirectionnel déployé par une roquette depuis le bâtiment de surface dans la direction estimée d'approche de la torpille * utiliser une contre-mesure a rejection contrainte à la position du porteur larguée par-dessus bord ou détachée du porteur dans le cas où elle est remorquée.
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YE, Zi. "Traitement statistique de l'information et du signal pour l'internet des objets sous-marins." Thesis, Institut polytechnique de Paris, 2021. https://tel.archives-ouvertes.fr/tel-03179373.
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On assiste au développement des activités humaines liées au monde océanique, mais aucune norme n'a encore émergé pour l'Internet des objets appliqué aux objets autonomes marins. Bien qu'elle possède une bande passante limitée, l'onde acoustique est le seul moyen de communiquer sur des distances importantes et elle est donc utilisée par de nombreux systèmes sous-marins pour communiquer, naviguer ou déduire des informations sur l'environnement. Cela a conduit à une forte demande de réseaux sans fil qui nécessitent à la fois une bonne efficacité spectrale et énergétique avec la faible complexité des algorithmes associés. Par conséquent, au cours de ce doctorat, nous avons proposé plusieurs solutions originales pour relever le défi de développer des techniques numériques, capables de faire face au canal acoustique.En raison d’une diversité inhérente d'espace du signal (SSD), les constellations tournées permettent de meilleures performances théoriques que les constellations conventionnelles et ce, sans détérioration spectrale. Nous passons en revue les propriétés structurelles des constellations tournées M-QAM uniformément projetées, afin de proposer une technique de demapping souple à faible complexité pour les canaux à fading. Puis, nous proposons une technique originale de réduction du PAPR pour les systèmes OFDM utilisant les constellations tournées. Afin de réduire la complexité du décodage aveugle, nous nous appuyons sur les propriétés des constellations tournées M-QAM uniformément projetées, pour concevoir un estimateur de faible complexité. De plus, pour faire face à la sélectivité du canal acoustique, nous avons proposé un turbo-détecteur parcimonieux adaptatif avec seulement quelques coefficients à mettre à jour afin de réduire la complexité. Enfin, nous avons proposé un algorithme original auto-optimisé pour lequel les tailles de pas de l'égaliseur sont mises à jour de manière adaptative et assistées par des informations souples de manière itérative, afin de répondre à l'exigence de convergence rapide et de faible erreur quadratique sur des canaux variant rapidement dans le tempsThere has been recently a large development of human activities associated to the ocean world, where no standard has emerged for the Internet of Things (IoT) linked to marine autonomous objects. Though it has a limited bandwidth, the acoustic wave is the only way to communicate over average to large distances and it is thus used by many underwater systems to communicate, navigate, or infer information about the environment. This led to a high demand for wireless networks that require both spectral efficiency and energy efficiency with the associated low-complexity algorithms. Therefore, in this Ph.D. thesis, we proposed several original solutions to face this challenge.Indeed, due to the inherent Signal Space Diversity (SSD), rotated constellations allow better theoretical performance than conventional constellations with no spectral spoilage. We review the structural properties of uniformly projected rotated M-QAM constellations, so as to propose a low complexity soft demapping technique for fading channels. Then, we present an original blind technique for the reduction of the PAPR for OFDM systems using the rotated constellations with SSD. In order to reduce the complexity of blind decoding for this technique, we again rely on the properties of uniformly projected M-QAM rotated constellations to design a low-complexity estimator. Moreover, to face the selectivity of the acoustic channel, we suggest a sparse adaptive turbo detector with only a few taps to be updated in order to lower down the complexity burden. Finally, we have proposed an original self-optimized algorithm for which the step-sizes of both the equalizer and the phase estimator are updated adaptively and assisted by soft-information in an iterative manner, so as to meet the requirement of fast convergence and low MSE over time-varying channels
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Han, Dong. "Caractérisation des objets enfouis par les méthodes de traitement d'antenne." Thesis, Aix-Marseille 3, 2011. http://www.theses.fr/2011AIX30003/document.
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Cette thèse est consacrée à l'étude de la localisation d'objets enfouis dans acoustiques sous-marins en utilisant les méthodes de traitement d'antenne et les ondes acoustiques. Nous avons proposé un modèle bien adapté en tenant compte le phénomène physique au niveau de l'interface eau/sédiment. La modélisation de la propagation combine donc la contribution de l'onde réfléchie et celle de l'onde réfractée pour déterminer un nouveau vecteur directionnel. Le vecteur directionnel élaboré à partir des modèles de diffusion acoustique est utilisé dans la méthode MUSIC au lieu d'utiliser le modèle d'onde plane habituel. Cette approche permet d'estimer à la fois coordonnées d'objets (angle et distance objet-capteur) de forme connue, quel que soit leur emplacement vis à vis de l'antenne, en champ proche ou en champ lointain. Nous remplaçons l'étape de décomposition en éléments propres par des algorithmes plus rapides. Nous développons un algorithme d'optimisation plus élaboré consiste à combiner l'algorithme DIRECT (DIviding RECTangles) avec une interpolation de type Spline, ceci permet de faire face au cas d'antennes distordues à grand nombre de capteurs, tout en conservant un temps de calcul faible. Les signaux reçus sont des signaux issus de ce même capteur, réfléchis et réfractés par les objets et sont donc forcément corrélés. Pour cela, nous d'abord utilisons un opérateur bilinéaire. Puis nous proposons une méthode pour le cas de groupes indépendants de signaux corrélés en utilisant les cumulants. Ensuit nous présentons une méthode en utilisant la matrice tranche cumulants pour éliminer du bruit Gaussien. Mais dans la pratique, le bruit n'est pas toujours gaussien ou ses caractéristiques ne sont pas toujours connues. Nous développons deux méthodes itératives pour estimer la matrice interspectrale du bruit. Le premier algorithme est basé sur une technique d'optimisation permettant d'extraire itérativement la matrice interspectrale du bruit de la matrice interspectrale des observations. Le deuxième algorithme utilise la technique du maximum de vraisemblance pour estimer conjointement les paramètres du signal et du bruit. Enfin nous testons les algorithmes proposés avec des données expérimentales et les performances des résultats sont très bonnesThis thesis is devoted to the study of the localization of objects buried in underwater acoustic using array processing methods and acoustic waves. We have proposed a appropriate model, taking into account the water/sediment interface. The propagation modeling thus combines the reflected wave and the refracted wave to determine a new directional vector. The directional vector developed by acoustic scattering model is used in the MUSIC method instead of the classical plane wave model. This approach can estimate both of the object coordinates (angle and distance sensor-object) of known form, in near field or far field. We propose some fast algorithms without eigendecompostion. We combine DIRECT algorithm with spline interpolation to cope with the distorted antennas of many sensors, while maintaining a low computation time. To decorrelate the received signals, we firstly use a bilinear operator. We propose a method for the case of independent groups of correlated signals using the cumulants. Then we present a method using the cumulants matrix to eliminate Gaussian noise. But in practice, the noise is not always Gaussian or the characteristics are not always known. We develope two iterative methods to estimate the interspectral matrix of noise. The first algorithm is based on an optimization technique to extract iteratively the interspectral matrix of noise. The second algorithm uses the technique of maximum likelihood to estimate the signal parameters and the noise. Finally we test the proposed algorithms with experimental data. The results quality is very good
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Xu, Kevin. "Maximum likelihood time-domain beamforming using simulated annealing." Thesis, Massachusetts Institute of Technology, 1999. http://hdl.handle.net/1721.1/80046.
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Thesis (S.M.)--Joint Program in Oceanographic Engineering (Massachusetts Institute of Technology, Dept. of Ocean Engineering; and the Woods Hole Oceanographic Institution), 1999.Bibliography: p. 111-112.
by Kevin Xu.
S.M.
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Duboisset-Chareyre, Laure. "Analyse bispectrale de signaux réels : application à la détection de transitoires." Grenoble INPG, 1997. http://www.theses.fr/1997INPG0168.
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En lutte sous-marine, des l'acquisition d'un signal en sortie d'une antenne sonar, la premiere question qui se pose est : <<<>le signal comporte-t-il de l'information ou ne s'agit-il que de bruit?<>>> cette question est determinante pour la securite de l'equipage et du batiment. Depuis peu, on s'interesse a la detection des bruits transitoires. Signaux brefs, emis lors de manoeuvres, ils sont souvent moins discrets que les bruits rayonnes classiques mais leur caractere fugitif les avait rendus jusque-la moins remarquables. Les methodes classiques de detection sont souvent mal adaptees a ce type de signaux. C'est pourquoi nous abordons d'autres methodes pour ameliorer cette detection. Des algorithmes utilisant les transformees temps-frequence et les ordres superieurs sont compares. A l'issue de cette etude, nous montrons l'interet d'une methode utilisant le calcul du bispectre. Ce dernier presente des caracteristiques remarquables que nous decrivons precisement. L'utilisation du detecteur bispectral nous amene a etudier en detail l'environnement reel dans lequel nous travaillons. Nous montrons notamment que le bruit de mer presente des caracteristiques proches de celles d'une gaussienne qui le differencie des transitoires a detecter. Enfin, notre souci etant de proposer un detecteur operationnel, nous avons mis au point la detection bispectrale en temps reel. Cela nous a amenes a faire des choix quant a la programmation du detecteur en terme, par exemple, de longueur de fenetres d'analyse lors de l'estimation du bispectre. Nous expliquons les difficultes rencontrees lors de l'implantation. Nous montrons comment elles ont ete resolues et presentons les performances de la methode et du programme.
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Carmillet, Valerie. "Contribution à la détection en présence de réverbération : applications en acoustique sous-marine." Grenoble INPG, 1998. http://www.theses.fr/1998INPG0004.
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En acoustique sous-marine, la reverberation est souvent consideree comme une source de bruit qui perturbe les systemes de detection par sonar actif. Longtemps, on a lutte contre ce bruit par des moyens classiques comme le filtre adapte qui ne tient pas compte des proprietes statistiques de la reverberation, mais seulement de la forme du signal dont il est issu. Ce memoire a pour objet la mise en place d'un systeme de detection performant en presence de reverberation, exploitant au mieux les connaissances statistiques dont on dispose : forte coloration et non stationnarite, et adapte aux formes de signaux de plus en plus utilises en detection active : les signaux large-bande. Dans un premier temps, on rappelle sous quelles hypotheses le bruit de reverberation peut etre considere comme la sortie d'un filtre aleatoire non stationnaire. Les difficultes rencontrees lors de la mise en oeuvre du detecteur optimal (estimation de la fonction de diffusion), sont alors telles que l'on s'oriente ensuite naturellement vers des methodes sous-optimales. La methode proposee repose sur un blanchiment adaptatif realise par une technique autoregressive (ar) par bloc. La validite de l'hypothese sous-jacente de stationnarite locale est etudiee sur des signaux experimentaux au moyen de la distance d'itakura-saito. Le detecteur sur un bloc est alors calcule par le rapport de vraisemblance generalise. Cette approche est etudiee pour quatre types de signaux utilises dans la pratique (fp, fml, fmh et bpsk) sur des signaux reels de reverberation. Les performances en terme de courbes cor sont evaluees, aussi bien theoriquement que experimentalement. On compare ces resultats avec ceux du detecteur classique considerant le bruit de reverberation comme blanc par bloc mais non stationnaire. La methode proposee est particulierement performante pour detecter des cibles lentes, dont le doppler est proche du doppler moyen de la reverberation.
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Mauuary, Didier. "Détection, estimation et identification pour la tomographie acoustique océanique : étude théorique et expérimentale." Grenoble INPG, 1994. http://www.theses.fr/1994INPG0033.
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La tomographie acoustique oceanique s'est developpee depuis une quinzaine d'annees en apparaissant comme un moyen jusque-la inexistant d'observer la dynamique des oceans sur des espaces de grandes echelles (100, 1000 km) et sur de longues periodes (1 mois, 1 an). Mais le principe de faisabilite de la methode est encore au cur du probleme surtout pour les zones experimentees par les laboratoires francais et europeens. Les resultats experimentaux ont remis en cause les procedes classiques de traitement du signal. Les instruments et la chaine de pretraitement ont fait l'objet d'une etude complete. Nous mettons en evidence les proprietes de la reponse impulsionnelle instrumentale et l'impact du doppler sur le systeme de mesure des temps de propagation des trajets multiples. Les proprietes spatiales du doppler et la sensibilite des signaux utilises en font une grandeur qui peut etre maintenant exploitee sous la forme d'une antenne a ouverture synthetique. Les methodes avancees de traitement du signal necessaires pour estimer les temps de propagation des trajets multiples et pour les identifier aux trajectoires predites par un modele acoustique sont formalisees avec les outils statistiques de decision. L'estimation bayesienne de temps de retard est analysee en detail et les domaines ou elle ameliore la precision des estimateurs sont donnes. Un nouveau concept, resolvant le probleme de l'identification, est propose. Il utilise d'une maniere fondamentale une information d'origine oceanique et un modele acoustique de trace de rayons. Les algorithmes qui en decoulent permettent d'identifier statistiquement les trajets acoustiques instables ou non resolus. Tous les concepts proposes sont valides sur des donnees experimentales
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Real, Gaultier. "An ultrasonic testbench for reproducing the degradation of sonar performance in fluctuating ocean." Thesis, Aix-Marseille, 2015. http://www.theses.fr/2015AIXM4753/document.
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Le milieu océanique est sujet à de nombreuses sources de fluctuations. Les plus importantes sont les ondes internes, très fréquentes et entrainant des fluctuations de la distribution spatiale du champ de célérité du son. En raison de la longue période de ces phénomènes comparée au temps de propagation des ondes acoustiques pour des applications sonar, le processus peut être considéré figé dans le temps pour chaque réalisation stochastique du milieu. Le développement de bancs d’essais permettant de reproduire les effets de la turbulence atmosphérique a permis des avancées considérables dans le domaine de l’optique adaptative. Nous voyons donc un fort intérêt dans la possibilité de reproduire les effets des ondes internes sur la propagation du son en environnement contrôlé. Un protocole expérimental dans une cuve d’eau est proposé: une onde ultrasonore est transmise à travers une lentille acoustique aléatoirement rugueuse, ce qui produit des distorsions du front d’onde reçu. Les fluctuations des signaux reçus sont contrôlées en modifiant les paramètres statistiques de rugosité de la lentille. Ces paramètres sont reliés à l’analyse dimensionnelle permettant de classifier les configurations étudiées selon des régimes de fluctuations et de prédire les moments statistiques du champ acoustique jusqu’à l’ordre quatre. Une excellente correspondance est observée entre notre protocole expérimental et des résultats théoriques et numériques.La dégradation des performances des techniques de détection classiques appliquées à nos données expérimentales souligne le besoin de techniques correctives. Un état de l’art des techniques existantes dans divers domaines est proposéThe ocean medium is subject to many sources of fluctuations. The most critical ones were found to be internal waves, occurring frequently and inducing fluctuations of the spatial distribution of the sound speed field. Because of the fairly long period of this phenomenon as compared to the propagation time of acoustic waves for sonar applications, the process can be considered frozen in time for each stochastic realization of the medium. The development of testbenches allowing to reproduce the effect of atmospheric turbulence on optic waves propagation under laboratory conditions lead to considerable advancements in the field of adaptive optics. We therefore see a vivid interest in being able to reproduce the effects of internal waves on sound propagation in controlled environments. An experimental protocol in a water tank is proposed: an ultrasonic wave is transmitted through a randomly rough acoustic lens, producing distortions of the received wavefront. The induced signal fluctuations are controlled by tuning the statistical parameters of the roughness of the lens. Especially, they are linked to dimensional parameters allowing to classify the configurations into regimes of fluctuations and to predict the statistical moment of the acoustic pressure up to the fourth order. A remarkable relevance of our experimental scheme is found when compared to theoretical and simulation results. The degradation of classical signal processing techniques when applied to our acquired data highlights the need for corrective detection techniques. A review of the existing techniques in other domains is proposed
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Bouhier, Marie Edith. "Amélioration des performances en portée et en précision de localisation angulaire des systèmes de navigation sous-marine." Grenoble INPG, 1986. http://www.theses.fr/1986INPG0122.
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L'amelioration de la precision de localisation angulaire des systemes de navigation base-courte et ultra-courte necessite de considerer des bruits correles ou non sur les deux capteurs de reception selon la taille de la base acoustique par rapport au rayon de correlation spatiale du bruit
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Conan, Ewen. "Traitements adaptés aux antennes linéaires horizontales pour la discrimination en immersion de sources Ultra Basse Fréquence." Thesis, Ecole nationale supérieure Mines-Télécom Atlantique Bretagne Pays de la Loire, 2017. http://www.theses.fr/2017IMTA0016/document.
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Les travaux présentés s'intéressent à la discrimination en immersion d'une source acoustique sous-marine monochromatique ultra basse fréquence (UBF, 0-500 Hz) à l'aide d'une antenne horizontale d'hydrophones. La discrimination en immersion consiste à déterminer si un signal reçu a été émis à proximité de la surface ou par une source immergée. Cette problématique est particulièrement intéressante pour la lutte sous-marine (discrimination entre bâtiments de surface et sous-marins) ou la biologie marine (discrimination entre espèces vocalement actives à la surface et en profondeur). Le champ acoustique généré par une source UBF peut être décomposé en modes, dont les caractéristiques dépendent de l'environnement et de la position de la source. Cette propagation modale est source de dispersion modale : les différents modes se propagent à différentes vitesses. Cela empêche d'utiliser les techniques classiques de traitement d'antenne. Cependant, l'antenne horizontale peut être utilisée comme un filtre spatial pour estimer les propriétés des différents modes : on parle alors de filtrage modal. Si l'antenne est suffisamment longue, les modes sont résolus et les modes filtrés peuvent servir à localiser la source (matched-mode processing). Dans le cas d'une antenne trop courte, les modes sont mal filtrés et la localisation est impossible. Nous cherchons donc une information moins précise mais plus robuste sur la position de la source, d'où le problème de la discrimination en immersion.Dans ces travaux, nous cherchons à exploiter les modes mal filtrés pour prendre une décision sur le caractère immergé ou non de la source. Nous proposons de baser cette décision sur la valeur estimée du taux d'énergie piégée, i.e. la proportion de l'énergie acoustique qui est portée par les modes piégés. Le problème de la discrimination est alors posé comme un test d'hypothèses binaire sur la profondeur de la source. Cette formulation physique du problème permet d'utiliser des méthodes de Monte Carlo pour prédire, à l'aide de simulations, les performances en discrimination dans un contexte donné. Cela permet de comparer diverses méthodes d'estimation du taux d'énergie piégée, et surtout de choisir un seuil auquel comparer ce taux pour décider si la source est en surface ou immergée.La méthode développée pendant la thèse est validée sur des données expérimentales marines. Les résultats alors obtenus sont cohérents avec les conclusions tirées des simulations. La méthode proposée permet notamment d'identifier avec succès une source de surface (le bruit d'un navire en déplacement) ainsi qu'une source immergée (une source UBF tractée à 30 m de profondeur), à l'aide d'une antenne horizontale de 360 mThis work focuses on acoustic source depth discrimination in the ultra-low frequency range (ULF, 0-500 Hz), using a horizontal line array. Depth discrimination is a binary classification problem, aiming to evaluate whether a received signal has been emitted by a source near the surface or by a submerged one. This could serve applications such as anti-submarine warfare or marine biology.The acoustic field generated by a ULF source can be described as a sum of modes, which properties depend on environment and source location. This modal propagation leads to modal dispersion: the different modes propagate at different velocities. This forbid the use of classical beamforming schemes. However, the horizontal array can be used as a spatial filter to estimate the properties of the modes: this is modal filtering. With a sufficient array length, modes are resolved, and the filtered modes can be used to localise the source using matched-mode processing. If the array is too short, the poorly-filtered modes cannot be used for localisation. Therefore, we are looking for a less precise but more robust information on source location, which leads to source depth discrimination.In this work, the poorly-filtered modes are used to decide whether the source is near the surface or submerged. Because some of the modes (the "trapped modes") are weakly excited by a surface source, we propose this decision relies on the estimation of the trapped energy ratio, i.e. the ratio of acoustic energy borne by trapped modes to the total acoustic energy. The problem of depth discrimination is then formulated as a binary hypothesis test on source depth. This physical formulation allows using Monte-Carlo methods and simulations to predict performance in a given context. This enables comparison between several estimators of the trapped energy ratio and the choice of a relevant threshold which this ratio is compared to in order to decide between the two hypotheses. The approach developped in the manuscript is validated by its application to marine experimental data. The results are consistent with the conclusions drawn from simulations. The proposed method enables the succesfull identification of both a surface source (the noise of a travelling ship) and a submerged source (a ULF source towed 30 m below the surface), using a 360-m horizontal array
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Courcoux-Caro, Milan. "Conception optimisée d’antenne pour de la localisation passive de sources acoustiques." Thesis, Brest, École nationale supérieure de techniques avancées Bretagne, 2022. http://www.theses.fr/2022ENTA0001.
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L’objectif de ces travaux de thèse est de proposer une approche de conception d’antenne pour de la localisation de sources acoustiques aériennes ou sous-marines. Dans un premier temps, nous verrons comment décrire la propagation des ondes émises par les sources. Puis nous dresserons les différentes méthodes permettant d’estimer les positions des sources à partir des mesures acquises par les capteurs et du modèle de propagation. Le travail bibliographique sur le domaine de la localisation de sources acoustiques mettra en évidence l’importance des positions des capteurs dans les performances des estimateurs de positions de sources. Cela amènera donc à un second travail bibliographique sur la conception d’antenne, et plus précisément sur la sélection de positions de capteurs. L’état de l’art sur les méthodes de conception d’antenne nous permettra de proposer la méthode suivante : une sélection séquentielle bayésienne orientée par les données. Cette approche prend en compte l’information contenue dans les mesures acquises précédemment dans l’antenne pré-conçue, afin de sélectionner les futurs capteurs. L’application de cette approche est innovante dans un contexte de localisation de sources acoustiques. Dans un deuxième temps, des premiers résultats réalisés sur données synthétiques s’avèrent prometteurs quant à l’application de cette approche dans un cadre de localisation de sources acoustiques sous-marines. Les analyses fournies dans ce manuscrit permettront de juger la pertinence de cette approche dans le contexte testé, ainsi que d’évaluer et de comparer la performance de cette conception d’antenne par rapport à la littérature. Pour finir, nous appliquerons cette approche sur des données réelles provenant d’une expérience mise en place lors de la thèse. Le milieu de propagation sera aérien dans une salle fermée réverbérante avec un grand nombre de capteurs disponibles. Acquérir des données réelles permettront, en plus de l’évaluation des performances de notre approche, de fournir une nouvelle conception d’antenne. Celle-ci est conçue à partir de données synthétiques et est appliquée ensuite sur les données réellesThe objective of this thesis work is to propose an approach of array design for the localization of aerial or underwater acoustic sources. First, we will see how to describe the propagation of waves emitted by the sources. Then we will draw up the different methods allowing to estimate the positions of the sources from the measurements acquired by the sensors and from the propagation model. The bibliographic work on the field of acoustic source localization will highlight the importance of the sensors’ positions in the performance of the source position estimators. This will lead to a second bibliographic work on antenna design, and more precisely on the selection of sensor positions. The state of the art on antenna design methods will allow us to propose the following method: a sequential Bayesian data-driven selection. This approach takes into account the information contained in the measurements previously acquired in the pre-designed antenna, in order to select the future sensors. The application of this approach is innovative in the context of acoustic source localization. In a second step, first results realized on synthetic data prove to be promising for the application of this approach in an underwater acoustic source localization context. The analyses provided in this manuscript will allow to judge the relevance of this approach in the tested context, as well as to evaluate and compare the performance of this antenna design with the literature. Finally, we will apply this approach on real data from an experiment set up during the thesis. The propagation environment will be aerial in a closed reverberant room with a large number of available sensors. Acquiring real data will allow, in addition to evaluating the performance of our approach, to provide a new antenna design. This is designed from synthetic data and then applied to real data
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Simão, Daniel Hayashida. "Análise do consumo energético em redes subaquáticas utilizando códigos fontanais." Universidade Tecnológica Federal do Paraná, 2017. http://repositorio.utfpr.edu.br/jspui/handle/1/2774.
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O presente trabalho aborda a aplicação de códigos fontanais em redes subaquáticas. Tais redes transmitem dados abaixo da água fazendo uso de sinais acústicos e possuem diversas aplicações. No entanto, é sabido que esse tipo de rede é caracterizado por uma baixa velocidade de propagação e largura de banda menor que as redes que operam em meios de transmissão mais conhecidos, tais como a transmissão sem fio via ondas de rádio frequência, resultando num maior atraso na entrega de pacotes. Para tentar minimizar estes atrasos e aumentar a eficiência energética das redes subaquáticas, o trabalho otimizou o sistema de transmissão inserindo um código corretor de erros fontanal no transmissor de mensagens. Dentro desse contexto, foi necessário modelar o consumo energético necessário para a transmissão correta de pacotes de dados em redes subaquáticas utilizando códigos fontanais. Dentre os resultados do trabalho, o mais relevante conclui que o uso dos códigos fontanais é capaz de reduzir em até 30% o consumo de energia quando a distância de transmissão é de 20 km para o caso com a taxa de erro de quadro alvo (FER) de Po = 10^−5, e em ate 25% para a FER alvo de Po = 10^−3.The present work employs fountain codes in an underwater network, in which data is transmitted using acoustic signals and has many applications. However, underwater networks are usually characterized by low propagation speed and smaller bandwidth than networks that use radio frequency signals, resulting in larger transmission delays. Then, aiming at minimizing the delays and increasing the energy efficiency of underwater networks, the present work employs fountain error-correcting codes at the transmitter. To that end, it was first necessary to model the energy consumption of a success data packet transmission in an underwater network using fountain codes. Our results show that the use of fountain codes is able to reduce up to 30% of energy consumption when the transmission distance is of 20 km for the case with a target frame error rate (FER) of Po = 10^−5 , and 25% for the same distance with a target FER of Po = 10^−3.
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Tabella, Gianluca. "Subsea Oil Spill Risk Management based on Sensor Networks." Master's thesis, Alma Mater Studiorum - Università di Bologna, 2019.
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This thesis consists of the evaluation of sensor-based risk management against oil spills using an underwater distributed sensor network. The work starts by highlighting the importance of having a performing leak detection system both from an environmental, safety and economic point of view. The case study is the Goliat FPSO in the Barents Sea which has to meet requirements dictated by Norwegian authorities to prevent oil spills. The modeled network is made of passive acoustic sensors monitoring the subsea manifolds. These sensors send their local 1-bit decision to a Fusion Center which takes a global decision on whether the leakage is occurring. This work evaluates how the choice of adapted Fusion Rules (Counting Rule and Weighted Fusion Rule) can affect the performances of the leak detection system in its current geometry. It will also be discussed how different thresholds, selected for a specific FR or sensor test, can change the system performance. The detection methods are based on statistical signal processing adapted to fit this application within the Oil&Gas field. The work also proposes some new leak localization methods developed so they can be coupled with the proposed leak detection methods, giving a coherent set of operations that the sensors and the FC must perform. Performances of detection techniques are assessed balancing the need for high values of True Positive Rate and Precision and low values of False Positive Rate using indexes based both on the ROC curve (like the Youden's Index) and on the PR curve (the F-scores). Whereas, performances of localization techniques will be assessed on their ability to localize the spill in the shortest time; if this is not possible, parameters like the difference between the estimated and the real leak position will be considered. Finally, some tests are carried out applying the different sets of proposed methods.
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Su, Shun-Chi, and 蘇順吉. "Studies on underwater acoustic stationary and transient signals spectrum features." Thesis, 1998. http://ndltd.ncl.edu.tw/handle/20487262396994551309.
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碩士中正理工學院
電機工程研究所
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Underwater acoustic signals are non-linear, time-varying, and with low signal-to-noise ratio. These properties make the signal analysis difficulty and complex. For resolving targets through the underwater acoustic signals, effective methods are proposed in this thesis to process underwater acoustic signals, Base on these methods, an signal acoustic recognition system is also designed. Traditionally, the Fourier transform (FT) and Morlet wavelet transform (MWT) are the main tool for stationary and transient signals spectrum analysis, respectively. Here in, a modify power spectrum density (PSD) function is used to extract the critical features for stationary underwater acoustic signals, A multi-scaling MWT kernel is also proposed which can depict the underwater transient spectrum successfully. To illustrate the effectiveness of these two novel design methods, some experiments are taken to perform by using simulation and recorded real underwater acoustic signals. Experimented results show that the proposed methods can detect and analyze both stationary and transient underwater acoustic signals successfully. An underwater acoustic signals analysis is also implemented on Matlab base personal computer to detect, analyze, and recognize targets by stationary signal features. It is hoped that an automatic underwater targets recognition system can be realized by methods discussed in this thesis in the future.