Dissertations / Theses on the topic 'Acoustics- Signal processing'
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1
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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Hermand, Jean-Pierre. "Environmentally-Adaptive Signal Processing in Ocean Acoustics." Doctoral thesis, Universite Libre de Bruxelles, 1993. http://hdl.handle.net/2013/ULB-DIPOT:oai:dipot.ulb.ac.be:2013/212734.
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Manning, George Keith. "Signal processing for ultrasonic foetal monitoring." Thesis, University of Edinburgh, 1987. http://hdl.handle.net/1842/12559.
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Keenan, Desmond Barry. "Enhanced signal processing of pulsed doppler ultrasound." Thesis, University of Ulster, 1998. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.342411.
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Rex, James Alexander. "Microphone signal processing for speech recognition in cars." Thesis, University of Southampton, 2000. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.326728.
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Orduna-Bustamante, Felipe. "Digital signal processing for multi-channel sound reproduction." Thesis, University of Southampton, 1995. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.261565.
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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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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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Murphy, Damian Thomas. "Digital waveguide mesh topologies in room acoustics modelling." Thesis, University of York, 2000. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.313846.
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Chen, Zhenxiang. "The applications of ultrasonic imaging and signal processing in two-phase flow measurement." Thesis, Cranfield University, 1996. http://dspace.lib.cranfield.ac.uk/handle/1826/10516.
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The applications of ultrasonic imaging and signal processing in two-phase flow measurement have been investigated in this thesis. The scattering by single and many scatterers was studied experimentally and numerically. The statistical properties of the scattered waves from many scatterers were examined and the corresponding results are presented. Ultrasonic transmission/reflection mode tomography was introduced. The theories of reflection mode computerised tomography were developed, by which reflectivity functions and scattering amplitude functions can be reconstructed. Image restoration and interpretation methods are presented. Computer simulation of ultrasonic measurements were carried out. A ultrasonic tomographic imaging system was developed, in which fan-shaped sound beam insonification was employed. Static physical models were used to simulate two-phase flows. In order to speed up the data acquisition of a tomographic imaging system, the single receiver mode and multiple receiver mode data acquisition arrangements were studied by experiments. Experiments on imaging small and large objects were carried out. Several signal and image processing methods were examined. A modified histogram equalisation algorithm was developed for processing the resultant ultrasonic images. The experiment results show that the proposed image reconstruction methods are satisfactory. Possible future developments are proposed.
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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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Kirchmann, Carl Christian. "Automatic acoustic tests of conference phones." Thesis, Uppsala universitet, Fasta tillståndets elektronik, 2017. http://urn.kb.se/resolve?urn=urn:nbn:se:uu:diva-326253.
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This master thesis was completed at Limes Audio in Umeå. Limes Audio are specialized in speech enhancement software which is mostly used in conference phones. Their software is used to reduce noise, echo cancellation as well as reduction of distortion. When a conference phone is malfunctioning it can be troublesome to find the cause of the problem. Common issues that lead to perceived poor audio quality are rattling plastics, loose parts and an environment with demanding acoustics. There is a need for automatic tests to run on conference phone which can diagnose audio quality and the acoustics of the environment where the conference phone is located in. A previous master thesis by Wilhelmsson called "Estimating Loudspeaker Distortion and Room Reverberation Time Using a Speakerphone" focused on evaluating several tests run in MATLAB for analyzing acoustic characteristics of a room. The goal of this thesis was to continue the work of the master thesis by Wilhelmsson. Previous tests served as a basis for this thesis. The tests where modified and rewritten in C and run on a Linux computer connected to a specific conference phone model. In the extended test made in this thesis using sine sweep signals, the reverberation time as well as the distortion for the conference phone setup was determined. The calculated reverberation time was compared to the results using Room EQ Wizard and gave almost identical results. Total harmonic distortion was not easy to relate to the perceived audio quality. There is a need to further develop methods to take into account other types of distortion than the total harmonic distortion. Rub & buzz is one kind of distortion that could be of interest. To investigate pass or fail thresholds for reverberation time as well as distortion is central for further development of this test software.
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Zelnio, Anne M. "Detection of Small Aircraft using an Acoustic Array." Wright State University / OhioLINK, 2009. http://rave.ohiolink.edu/etdc/view?acc_num=wright1247075795.
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Wu, Tsan-Ming. "Statistical impulse reponse modeling and dereverberation for room acoustics." Diss., Georgia Institute of Technology, 2000. http://hdl.handle.net/1853/14932.
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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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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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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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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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Baker, David L. "Development of a Rotordynamic Signal Processing MATLAB Interface and a Two-Disk Rotor Model." DigitalCommons@CalPoly, 2017. https://digitalcommons.calpoly.edu/theses/1794.
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Using MATLAB and a National Instruments data acquisition card, a signal processing program meant to monitor the behavior of rotordynamic systems in real-time was developed and tested. By using traditional analysis methods in this field of engineering, commonly desired data representations such as bode, polar, orbit, full spectrum plots were able to be produced to a very high accuracy. Additional capabilities offered by this application are slow roll compensation, synchronous and sub-synchronous filtering, and true three dimensional plotting. The verification of this program was done by comparing the results to the ones acquired with Bently Nevada’s “Automated Diagnostics for Rotating Equipment” (ADRE) system. In addition to a data acquisition program, theoretical models of the two-disk rotor were created to estimate the unknown physical parameters of the system. By simulating the rotor with and without gyroscopic effects included, estimates for the stiffness, damping, eccentricity, initial phase, and initial skew values present in the system were determined.
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Efraimsson, Nils. "Onset detection in polyphonic music." Thesis, KTH, Tal, musik och hörsel, TMH, 2017. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-210417.
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In music analysis, the beginning of events in a music signal (i.e. sound onset detection) is important for such tasks as sound segmentation, beat recognition and automatic music transcription. The aim of the present work was to make an algorithm for sound onset detection with better performance than other state-of-the-art1 algorithms. Necessary theoretical background for spectral analysis on a sound signal is given with special focus on the Short-Time Fourier Transform (STFT) and the effects of applying a window to a signal. Previous works based on different approaches to sound onset detection were studied, and a possible improvement was observed for one such approach - namely the one developed by Bello, Duxbury, Davies, & Sandler (2004). The algorithm uses an STFT approach, analyzing a sound signal time frame by time frame. The algorithm’s detection is sequential in nature: It takes a frame from the STFT and makes an extrapolation to the next frame, assuming that the signal is constant. The difference between the extrapolated frame and the actual frame of the STFT constitutes the detection function. The proposed improvement lies in a combination of ideas from other algorithms, analyzing the signal with different frequency bands with frequency dependent settings and a modification of the extrapolation step. The proposed algorithm is compared to the original algorithm and an adaption by Dixon (2006) by analyzing 20 songs using three different window functions. The results were evaluated with the standards set by MIREX (2005-2016). The results of the proposed algorithm are encouraging, showing good recall, but fail to out-perform any of the algorithms it is compared to in both precision and the so-called F-measure. The shortcomings of the proposed algorithm leave room for further improvement, and a number of possible future modifications are exemplified.Ansatsdetektion används inom musikanalys för bland annat automatisk transkription och ljudkomprimering. Ansatsdetektion innebär att lokalisera en händelse i en musiksignal. Med målet att utveckla en algoritm som presterar bättre än aktuella2 algoritmer ges här en genomgång av några nödvändiga teoretiska kunskaper i ämnet, bland annat korttids-Fouriertransformen (STFT) och hur fönsterfunktioner påverkar signalbehandling. Tidigare arbeten inom ansatsdetektion med olika infallsvinklar studeras och en möjlig förbättring av en av dem, den av Bello, Duxbury, Davies, & Sandler (2004), framträder. Algoritmen använder sig av STFT och analyserar ljudsignaler en tidsenhet i taget. Utifrån varje analyserad tidsenhet görs en extrapolation till nästa tidsenhet genom antagandet att signalen är konstant. Skillnaden mellan den extrapolerade tidsenheten och den faktiska tidsenheten i STFTn utgör detektionsfunktionen. Den möjliga förbättringen består i att använda idéer från olika algoritmer för ansatsdetektion – ljudsignalen analyseras i olika frekvensband med bandberoende inställningar för STFTn – och en förändrad extrapoleringsfunktion. Den föreslagna algoritmen jämförs med originalet av Bello, Duxbury, Davies, & Sandler (2004) och även med en variant utvecklad av Dixon (2006) genom att applicera dem på 20 spår med tre olika fönsterfunktioner. Resultaten utvärderas enligt MIREX (2005-2016) standarder och är lovande för algoritmen, då den har en bra träffbild, men både träffsäkerhet och F-värde ligger under de båda andra. Ett flertal möjliga förbättringar av algoritmen iakttas och presenteras.
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Warneke, Andrew Travis. "An evaluation of the efficacy of digital real-time noise control techniques in evoking the musical effect." Thesis, Nelson Mandela Metropolitan University, 2012. http://hdl.handle.net/10948/d1020158.
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This study sought to determine whether or not it may be possible to evoke ‘the musical effect' – the emotional response perceived by music listeners – using white noise as a sound-source and real-time digital signal processing techniques. This information was considered to be valuable as in a world driven by technological progress the potential use of new or different technologies in creating music could lead to the development of new methods of – and tools for – composition and performance. More specifically this research asked the question 'what is music?' and investigated how humans – both trained musicians and untrained people – perceive it. The elements of music were investigated for their affective strengths and new fields of research explored for insights into emotion identification in music. Thereafter the focus shifted into the realm of Digital Signal Processing. Common operations and techniques for signal manipulation were investigated and an understanding of the field as a whole was sought. The culmination of these two separate, yet related, investigations was the design and implementation of a listening experiment conducted on adult subjects. They were asked to listen to various manipulated noise-signals and answer a questionnaire with regard to their perceptions of the audio material. The data from the listening experiment suggest that certain DSP techniques can evoke ‘the musical effect’. Various musical elements were represented via digital techniques and in many cases respondents reported perceptions which suggest that some effect was felt. The techniques implemented and musical elements represented were discussed, and possible applications for these techniques, both musical and non-musical, were explored. Areas for further research were discussed and include the implementation of even more DSP techniques, and also into garnering a more specific idea of the emotion perceived by respondents in response to the experiment material.
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Ghaffari, Ghazaleh. "Estimation of Stapedius-Muscle Activation using Ear Canal Absorbance Measurements : An Application of Signal Processing in Physiological Acoustics." Thesis, Linköpings universitet, Institutionen för medicinsk teknik, 2013. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-98992.
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The stapedius muscle, which is located in the middle ear, goes into contraction when the ear is exposed to high sound intensities. This muscle activation is called ‘the acoustic reflex’. Measurement of the acoustic reflex is clinically of importance since it can reveal diagnostic information about the middle ear’s pathologies. Moreover, this muscle-activation alters the acoustic characteristics of the middle ear (i.e. the acoustic impedance and the power reflectance), which in turn, can significantly manipulate one’s perception of sounds. In the present study, these acoustic characteristics are measured in the ear canal by means of absorbance measures using equivalent Thevenin circuit theory. The quantities are then compared to form the shift responses between the baseline (before the activation) and the post-activator effect. This project investigates the shifts in power reflectance and admittance of the middle ear caused by the stapedius-muscle contraction. The wideband characterization (0.1- 8 kHz) of these acoustic reflex-induced shifts is achieved using chirp signals as a probe and through ipsilateral broadband noise activator. The data acquisition and signal processing of the project are carried out using MATLAB software. The hardware consists of National Instruments USB-6212 data acquisition interface and low noise microphone system Etymotic Research ER-10B+. A group of 10 adults including 5 males and 5 females are recruited as the participants for the project. The measurements of the reflectance shifts indicate that the most robust frequency region affected by the acoustic reflex is up to 4 kHz whereas for the admittance shifts, this region is up to 2 kHz. In addition, it is shown that the stapedius-muscle contraction leads to the attenuation of the lowfrequency transmission into the middle ear (less than 1 kHz) consistent with a stiffnesscontrolled system in this range of frequencies. In contrast, the results imply that the activation of the stapedius muscle leads to a slight enhancement of the frequency transmission in the range of 1-4 kHz (corresponding to the speech frequency band). These findings suggest a beneficial role for the stapedius-muscle contraction in the perception of speech during vocalization. Furthermore, the implemented methods in this project can be useful in better understanding the effect of the stapedius-muscle contraction on the speech perception both in normal hearing and hearing impaired persons.
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Jeon, Woojay. "Pitch detection of polyphonic music using constrained optimization." Thesis, Georgia Institute of Technology, 2002. http://hdl.handle.net/1853/15802.
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Kosek, Paul C. "Improved analysis of musical sounds using time-frequency distributions." Thesis, McGill University, 2005. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=83189.
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The objective of this research is to improve the analysis of musical sounds in comparison to traditional additive analysis, i.e. Fourier Analysis. Namely, the focus of this study is to improve the tracking of time-evolving partials. Traditional analysis methods assume constant amplitudes and frequencies over each successive frame in which a signal is analyzed. Tracking the time-evolution of these partials, however, can require the implementation of complex probabilistic techniques. This thesis presents an alternative method in which the Ambiguity Function, a distribution in both time and frequency, is used to create a clearer, more accurate representation that requires fewer complex methods to track partials. Through the use of a more accurate spectral representation and the inclusion of a chirp rate parameter, partials may be more readily followed based upon spectral parameters alone. This new method that is presented will build upon the traditional methods by first employing Fourier analysis to identify partials, and then utilizing the Analytic Signal and Ambiguity Function to improve individual spectral parameter estimations and partial tracking. The overall intent of this work is that through this method, one may create an improved spectral model that is more useful to musical analysis.
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Pérez, López Andrés. "Parametric analysis of ambisonic audio: a contributions to methods, applications and data generation." Doctoral thesis, Universitat Pompeu Fabra, 2020. http://hdl.handle.net/10803/669962.
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Due to the recent advances in virtual and augmented reality, ambisonics has emerged as
the de facto standard for immersive audio. Ambisonic audio can be captured using
spherical microphone arrays, which are becoming increasingly popular. Yet, many
methods for acoustic and microphone array signal processing are not speci cally tailored
for spherical geometries. Therefore, there is still room for improvement in the eld
of automatic analysis and description of ambisonic recordings. In the present thesis,
we tackle this problem using methods based on the parametric analysis of the sound
eld. Speci cally, we present novel contributions in the scope of blind reverberation
time estimation, diffuseness estimation, and sound event localization and detection.
Furthermore, several software tools developed for ambisonic dataset generation and
management are also presented.
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Chhabra, Manish. "Source Characterization using an Experimental Method and Prediction of Insertion of the Exhaust System." University of Cincinnati / OhioLINK, 2018. http://rave.ohiolink.edu/etdc/view?acc_num=ucin154399673454236.
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Aguda, Britt. "Three dimensional passive localization for single path arrival with unknown starting conditions." ScholarWorks@UNO, 2018. https://scholarworks.uno.edu/td/2513.
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Introduced in this paper is the time difference of arrival (TDoA) conic approximation method (TCAM), a technique for passive localization in three dimensions with unknown starting conditions. The TDoA of a mutually detected signal across pairs of detectors is used to calculate the relative angle between the signal source and the center point of the separation between the detectors in the pair. The relative angle is calculated from the TDoA using a mathematical model called the TDoA approximation of the zenith angle (TAZA). The TAZA angle defines the opening angle of a conic region of probability that contains the signal source, produced by each detector pair. The intersecting region of probability is determined from the conic regions of probability and represents the volumetric region with the highest probability of containing the signal source. TCAM was developed and tested using synthetic data in a simulated environment.
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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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Rouyer, Julien. "Tomographie ultrasonore dédiée à l'imagerie du sein - Validation expérimentale du projet ANAÏS." Phd thesis, Université de Provence - Aix-Marseille I, 2012. http://tel.archives-ouvertes.fr/tel-00733152.
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La tomographie ultrasonore assistée par ordinateur possède un fort potentiel en tant que moyen d'inspection des tissus mammaires pour le dépistage du cancer du sein ; cette technique permet de réduire la dépendance à l'opérateur constatée avec l'échographie conventionnelle. Une antenne de transducteurs (3 MHz) à géométrie semi-circulaire conformée à l'anatomie du sein a été développée pour réaliser une imagerie de réfl ectivité des structures d'intérêt en employant une procédure de reconstruction tomographique. L'antenne comporte 1024 éléments répartis sur un arc de 190 degrés ayant un rayon de courbure de 100 mm. Les acquisitions sont gérées par une électronique à 32 voies parallèles indépendantes en émission/réception et par un multiplexer pour l'adressage des voies vers les éléments de l'antenne. Les circuits d'émission et de réception ont une réquence d'échantillonnage allant jusqu'à 80 MHz avec une précision de 12 bits. Des formes d'ondes arbitraires (pseudo-chirp, codes de Golay) sont transmises a n d'améliorer le rapport signal sur bruit. L'électroacoustique a été caractérisée avec des objets académiques et un hydrophone a n de déterminer les propriétés d'émission du système d'imagerie (réponses impulsionnelles et distribution spatiale du champ) et de développer des outils de correction des données ; ces résultats sont mis en regard avec le formalisme de résolution du problème inverse (algorithme de sommation des rétroprojections elliptiques fi ltrées en champ proche). L'évaluation du système d'imagerie est réalisées sur des objets ponctuels ( fils de 80 m de diamètre), des objets bidimensionnels à faible contraste d'impédance et un fantôme anthropomorphique de sein contenant des inclusions. La technique de compression d'impulsion est utilisée pour traiter les signaux ; l'apport de cette technique à la tomographie ultrasonore est évaluée en regard d'une impulsion large bande. La résolution spatiale est inférieure au tier de la longueur d'onde et les images préliminaires réalisées avec le système sont très satisfaisantes. Des perspectives de développement des méthodes d'inspection et des adaptations du système électroacoustique pour la tomographique anatomique du sein sont proposées au vue des études réalisées durant cette thèse.
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Ren, Qunyan. "Remote sensing of sediment characteristics from the noise field due to a moving ship." Doctoral thesis, Universite Libre de Bruxelles, 2015. http://hdl.handle.net/2013/ULB-DIPOT:oai:dipot.ulb.ac.be:2013/222087.
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Qun-yan Ren received his Diploma in Electronic and Information Engineering, master degree in Underwater Acoustics Engineering both from Harbin Engineering University (HEU) in 2006 and 2009, respectively. Then he became a PhD student at the Environmental Hydroacoustics Laboratory, Ecole polytechnique de Bruxelles, Faculty of Applied Sciences, Universitie libre de Bruxelles (U.L.B.), Belgium, in co-tutelle with the National Key Laboratory of Underwater Acoustic Technology, HEU, China, under the co-supervision of Prof. Jean-Pierre Hermand and Prof. Piao Shengchun from U.L.B. and HEU, respectively. Since 2013, he became a full PhD student at the ULB. In Oct 2011, he obtained a four-year `Aspirantq{} grant from the Belgian National Fund for Scientific Research (F.R.S.-F.N.R.S.).Doctorat en Sciences de l'ingénieur et technologie
info:eu-repo/semantics/nonPublished
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Feng, Zao. "Condition Classification in Underground Pipes Based on Acoustical Characteristics. Acoustical characteristics are used to classify the structural and operational conditions in underground pipes with advanced signal classification methods." Thesis, University of Bradford, 2013. http://hdl.handle.net/10454/9463.
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Duong, Ngoc. "Modélisation gaussienne de rang plein des mélanges audio convolutifs appliquée à la séparation de sources." Phd thesis, Université Rennes 1, 2011. http://tel.archives-ouvertes.fr/tel-00667117.
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Nous considérons le problème de la séparation de mélanges audio réverbérants déterminés et sous-déterminés, c'est-à-dire l'extraction du signal de chaque source dans un mélange multicanal. Nous proposons un cadre général de modélisation gaussienne où la contribution de chaque source aux canaux du mélange dans le domaine temps-fréquence est modélisée par un vecteur aléatoire gaussien de moyenne nulle dont la covariance encode à la fois les caractéristiques spatiales et spectrales de la source. Afin de mieux modéliser la réverbération, nous nous affranchissons de l'hypothèse classique de bande étroite menant à une covariance spatiale de rang 1 et nous calculons la borne théorique de performance atteignable avec une covariance spatiale de rang plein. Les résultats expérimentaux indiquent une ugmentation du rapport Signal-à-Distorsion (SDR) de 6 dB dans un environnement faiblement à très réverbérant, ce qui valide cette généralisation. Nous considérons aussi l'utilisation de représentations temps-fréquence quadratiques et de l'échelle fréquentielle auditive ERB (equivalent rectangular bandwidth) pour accroître la quantité d'information exploitable et décroître le recouvrement entre les sources dans la représentation temps-fréquence. Après cette validation théorique du cadre proposé, nous nous focalisons sur l'estimation des paramètres du modèle à partir d'un signal de mélange donné dans un scénario pratique de séparation aveugle de sources. Nous proposons une famille d'algorithmes Expectation-Maximization (EM) pour estimer les paramètres au sens du maximum de vraisemblance (ML) ou du maximum a posteriori (MAP). Nous proposons une famille d'a priori de position spatiale inspirée par la théorie de l'acoustique des salles ainsi qu'un a priori de continuité spatiale. Nous étudions aussi l'utilisation de deux a priori spectraux précédemment utilisés dans un contexte monocanal ou multicanal de rang 1: un \textit{a priori} de continuité spatiale et un modèle de factorisation matricielle positive (NMF). Les résultats de séparation de sources obtenus par l'approche proposée sont comparés à plusieurs algorithmes de base et de l'état de l'art sur des mélanges simulés et sur des enregistrements réels dans des scénarios variés.
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Geay, Thomas. "Mesure acoustique passive du transport par charriage dans les rivières." Thesis, Grenoble, 2013. http://www.theses.fr/2013GRENU052/document.
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L'analyse des variations spatio-temporelles du charriage est un élément important pour la compréhension de la dynamique fluviale. Ce manuscrit présente les recherches réalisées sur le développement d'une méthode de mesure du transport solide par acoustique passive. Un capteur de pression acoustique est utilisé pour mesurer le bruit généré par le transport par charriage au fond du lit de la rivière. Cette méthode originale a fait l'objet de quelques explorations durant les dernières décennies, qui ont montré que la puissance et le contenu fréquentiel du son généré dépendaient de la granulométrie des matériaux et du flux solide. Mais les applications au milieu naturel sont restées très limitées ; elles sont donc au centre de cette recherche.La première difficulté de la mesure est liée à l'existence de bruit environnant qui se superpose au bruit du charriage. Des mesures acoustiques ont été réalisées dans différentes typologie de rivières, du torrent à la grande rivière navigable. A l'aide de ces différentes expériences et de la bibliographie, les éléments du paysage acoustique d'une rivière sont identifiés. Le paysage acoustique d'une rivière est composé par les processus hydrodynamiques que sont la turbulence, l'agitation de surface et le transport de sédiment par charriage. Le charriage produit des bruits larges bandes, dans la partie haute du spectre et peut être masqué par les bruits de surface dans la région du kilohertz. Des outils de traitement du signal sont proposés afin de repérer les différentes dynamiques contenues dans le signal acoustique mesuré.L'interprétation du signal ne peut être faite sans une bonne compréhension des phénomènes de propagation des ondes acoustiques dans la rivière. On montre que la rivière se comporte comme un guide d'onde et une résolution de l'équation d'onde par une approche modale est proposée. On comprend alors que la propagation des ondes acoustiques est limitée par une fréquence de coupure inversement proportionnelle à la hauteur d'eau. Les observations de terrain faites sur la variation du champ de pression acoustique dans la verticale sont bien reproduites par le modèle d'un guide d'onde de Pekeris. Le modèle est alors utilisé pour montrer l'importance de la profondeur, de la constitution du fond de la rivière ou encore de la géométrie du canal sur la constitution du signal.Finalement, trois chroniques de signaux acoustiques enregistrés dans des rivières différentes sont analysées. Un descripteur est proposé pour chaque chronique de signaux, en fonction des bruits ambiants présents dans l'environnement lors de la mesure. Ce descripteur acoustique est confronté à des mesures comparatives du charriage et de bonnes corrélations sont observée. Elles montrent que la mesure hydrophone permet d'identifier la phase de l'initiation du transport par charriage et qu'elle est intégrative du transport sur une surface importante de la rivière. Ces expériences confirment la simplicité de mise en œuvre de la méthode et précisent les limites d'utilisation de l'acoustique passive, particulièrement pour les rivières à fortes pentes. Elles confirment également la validité des méthodes d'analyse du signal qui ont été utilisées et le besoin de mesures comparatives du milieu pour interpréter le signalAnalysing the spatio-temporal variability of solid transport processes is key to the study of fluvial morphodynamics. Our research focusses on the development of passive acoustics to monitor bedload transport. A hydrophone is used to sense the acoustic pressure in the river in order to record the sound generated by inter-particle collisions. This original method has been mostly developed in laboratories during the past decades. It has been shown that the acoustic power and the frequencies of the monitored signals are linked to bedload fluxes and granulometry. The use of passive acoustics in natural streams has encountered limited success. It is the core of our research.First we address the existence of multiple sound sources in the environment. Acoustic measurements have been realised in several types of rivers: steep channels and large gravel bed rivers. These multiple experiences along with the bibliography have allowed us to describe river soundscapes. Hydrodynamics govern the soundscape, namely turbulence, agitating surfaces, and bedload transport. Inter-particle collisions generate sound in a wide range of frequencies, which depend on their sizes. It can be masked by the occurrence of agitating surface noise in the kilohertz region. Signal processing tools are proposed to study the dynamics of the different processes composing the signal.Signal interpretation could only be achieved by understanding the propagation properties of the acoustic waves in the river. It is shown that the river acts as an acoustic wave guide. A modal approach is suggested to solve the wave equation. The model points to the existence of a cutoff frequency inversely proportional to the water depth. Observations made on the vertical variation of the field pressure are correctly simulated. The signal dependence on water depth, the structure of the bed, and the geometry of the channel are studied using this model.Finally, we analyze three chronicles of acoustic signals recorded in the field. A signal descriptor is constructed for each data set, depending on the ambient noise conditions. This descriptor is compared to other measurements of bedload transport and good correlations are found. Initiation of motion is monitored and the integrative aspect of the acoustic measure is shown. These experiences highlight the simplicity of the method and show some of its limits. It is also shown that measurements of other environmental parameters are needed to interpret the results
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Hill, Adam J. "Analysis, modeling and wide-area spatiotemporal control of low-frequency sound reproduction." Thesis, University of Essex, 2012. http://hdl.handle.net/10545/230034.
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This research aims to develop a low-frequency response control methodology capable of delivering a consistent spectral and temporal response over a wide listening area. Low-frequency room acoustics are naturally plagued by room-modes, a result of standing waves at frequencies with wavelengths that are integer multiples of one or more room dimension. The standing wave pattern is different for each modal frequency, causing a complicated sound field exhibiting a highly position-dependent frequency response. Enhanced systems are investigated with multiple degrees of freedom (independently-controllable sound radiating sources) to provide adequate low-frequency response control. The proposed solution, termed a chameleon subwoofer array or CSA, adopts the most advantageous aspects of existing room-mode correction methodologies while emphasizing efficiency and practicality. Multiple degrees of freedom are ideally achieved by employing what is designated a hybrid subwoofer, which provides four orthogonal degrees of freedom configured within a modest-sized enclosure. The CSA software algorithm integrates both objective and subjective measures to address listener preferences including the possibility of individual real-time control. CSAs and existing techniques are evaluated within a novel acoustical modeling system (FDTD simulation toolbox) developed to meet the requirements of this research. Extensive virtual development of CSAs has led to experimentation using a prototype hybrid subwoofer. The resulting performance is in line with the simulations, whereby variance across a wide listening area is reduced by over 50% with only four degrees of freedom. A supplemental novel correction algorithm addresses correction issues at select narrow frequency bands. These frequencies are filtered from the signal and replaced using virtual bass to maintain all aural information, a psychoacoustical effect giving the impression of low-frequency. Virtual bass is synthesized using an original hybrid approach combining two mainstream synthesis procedures while suppressing each method‟s inherent weaknesses. This algorithm is demonstrated to improve CSA output efficiency while maintaining acceptable subjective performance.
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Silas, Kevin Alexander. "Phase Transform Time Delay Estimation to Counteract Spectral Haystacking Effects in Jet Exhaust Flow Measurements." Thesis, Virginia Tech, 2021. http://hdl.handle.net/10919/104892.
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This study determined a superior data processing technique for correlating an acoustic signal passing through a subsonic jet engine exhaust in order to estimate the traversal time of the signal. Thrust measurement is possible with enough time delay estimates across different portions of the exhaust. This preliminary study did not take the full array of data necessary to measure thrust, but did validate key aspects of the measurement process. The turbulent shear layers of the exhaust spectrally broaden the signal, creating the appearance of spectral "haystacks", making traditional correlation methods unworkable. An experiment was performed to evaluate the ability of a novel sound source to produce a signal from which a reliable and precise time delay estimate could be found. The test apparatus was installed on either side of a Honeywell TFE731-2 turbofan research engine exhaust cone, with the source and receivers placed near the jet exit plane. The signal was then directed across the jet exhaust. This flow environment is considered an extreme challenge for accurate acoustic signal propagation. A key contribution of this paper is the determination that the Phase Transform processor of the Generalized Cross-Correlation (GCC) method produces the most reliable time delay estimates, for the given signal and flow conditions. Several alternative time delay estimators and GCC processors were examined and evaluated on this data. A proposed explanation is provided for why this time delay estimation technique produces the most accurate results, as well as explanations for why the technique became less reliable as the flow environment became more challenging, with an observed 22% anomalous TDE selection rate for the N1Corr = 60% and N1Corr = 70% conditions combined, versus only 6% for the idle and N1Corr = 50% conditions combined. This paper also details the development and first use of a novel acoustic source that produces a two-tone narrowband signal emanating from a single point – the dual Hartmann generator.Master of Science
This study builds on a Computational Tomography (CT) technique that uses an acoustic signal and an array of receivers to measure the velocity and temperature of a gas flow field. In particular, the velocity and temperature field tested involves multiple turbulent and disruptive elements, requiring a loud and specifically designed signal. As such, a novel acoustic signal generator, the dual Hartmann generator, was designed that is both loud and produces a specific two-toned signal. The key contribution of the study was to process the data, comparing the sets of transmitted and received signals, in order to estimate the time delay amongst receiver pairs – a key input in the CT method. Traditional cross-correlation methods were inadequate, and multiple alternatives were evaluated. The Phase Transform (PHAT) technique showed the most promise, and an explanation is given for why this technique is most suitable for this type of signal.
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Arciniegas, Mosquera Andrés Felipe. "Analyse de méthodes statistiques en traitement du signal pour les tomographies acoustique et ultrasonore des arbres sur pied." Thesis, Aix-Marseille, 2014. http://www.theses.fr/2014AIXM4746/document.
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La tomographie acoustique est une méthode d'imagerie permettant de réaliser des cartographies bidimensionnelles du plan radial-transverse des arbres en fonction de la vitesse (ou la lenteur) des ondes élastiques de basse fréquence (<20 kHz). Les images couramment obtenues avec les appareils commerciaux possèdent une résolution spatiale faible (de l'ordre de quelques centimètres) et sont parfois difficiles à interpréter. Cette résolution est limitée par l'utilisation des ondes de basse fréquence, le faible nombre de sondes et la non-prise en compte des propriétés du bois (anisotropie, hétérogénéité). À ce jour, il n'existe pas d'appareils de terrain utilisant les ultrasons, spécialement adaptés à l'imagerie des arbres sur pied. Compte-tenu des limitations évoquées, nous proposons ici deux études cherchant à améliorer la qualité des images des tomographie acoustique et ultrasonore. Dans un premier temps nous comparons des méthodes de traitement du signal pour la mesure du temps de propagation et nous en précisons des limites expérimentales de validité. L'approche développée a permis de sélectionner les méthodes de traitement du signal en caractérisant les erreurs systématiques et aléatoires induites en fonction du niveau de bruit. Dans un deuxième temps, une étude numérique de la robustesse d'algorithmes de reconstruction est proposée. Deux nouveaux algorithmes sont présentés et comparés à deux algorithmes classiques utilisés dans les appareils commerciaux. Cette comparaison est axée sur les critères liés aux contraintes expérimentales (faible nombre de sondes et mesures bruitées) et aux exigences techniques (faible temps de calcul) pour une utilisation sur le terrainAcoustic tomography is an imaging technique used to perform two-dimensional mappings of the radial-transverse plane of trees, based on the velocity (or the slowness) of low frequency elastic waves (<20 kHz). The images currently obtained with the commercial devices have a low spatial resolution (of the order of a few centimeters) and are difficult to interpret. These resolution is limited by the use of low frequency waves, the low number of sensors and the fact of not taking into account the properties of wood (anisotropy, heterogeneity). To date, there are no field devices using ultrasound, specially adapted for standing trees imaging. Taking into account the limitations mentioned previously, we present hereby two studies that aim to improve the quality of acoustic and ultrasonic tomography images. In the first part of this work we compare signal-processing methods for the measurement of the propagation time and we specify experimental limits of validity. The approach developed permitted to choose the signal-processing methods by characterizing their systematic and random errors associated with the noise level. In the second part of this work, a numerical study of the robustness of reconstruction algorithms is proposed. Two new reconstruction algorithms are presented and compared to two conventional algorithms used in the commercial devices. These comparison is based on the criteria related to experimental constraints (low number of sensors and noisy measurements) and technical requirements (low computation time) for use in the field
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Fuller, Ryan Michael. "Adaptive Noise Reduction Techniques for Airborne Acoustic Sensors." Wright State University / OhioLINK, 2012. http://rave.ohiolink.edu/etdc/view?acc_num=wright1355361066.
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Freedman, Joseph Saul. "Using helicopter noise to prevent brownout crashes: an acoustic altimeter." Thesis, Georgia Institute of Technology, 2010. http://hdl.handle.net/1853/34833.
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This thesis explores one possible method of preventing helicopter crashes caused by brownout using the noise generated by the helicopter rotor as an altimeter. The hypothesis under consideration is that the helicopter's height, velocity, and obstacle locations with respect to the helicopter, can be determined by comparing incident and reflected rotor noise signals, provided adequate bandwidth and signal to noise ratio. Heights can be determined by measuring the cepstrum of the reflected helicopter noise. The velocity can be determined by measuring small amounts of Doppler distortion using the Mellin-Scale Transform. Height and velocity detection algorithms are developed, optimized for this application, and tested using a microphone array. The algorithms and array are tested using a hemianechoic chamber and outside in Georgia Tech's Burger Bowl. Height and obstacle detection are determined to be feasible with the existing array. Velocity detection and surface mapping are not successfully accomplished.
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Hult, Peter. "Bioacoustic principles used in monitoring and diagnostic applications /." Linköping : Univ, 2002. http://www.bibl.liu.se/liupubl/disp/disp2002/tek778s.pdf.
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Ishizawa, William Habaro. "Análise das concentrações energéticas no limiar entre fonemas vozeados e não-vozeados e suas implicações para fins de reconhecimento de locutores dependente do discurso." Universidade de São Paulo, 2015. http://www.teses.usp.br/teses/disponiveis/76/76132/tde-16042015-104351/.
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Atualmente, diversos trabalhos e aplicações são desenvolvidos com foco na área de reconhecimento computacional de locutores. À medida que o interesse por diversas aplicações reais dentro dessa área emerge, principalmente em biometria, na qual a segurança e a eficácia são de extrema importância, torna-se cada vez mais necessário que estudos sejam feitos, na mesma proporção, visando avaliá-las. Desse modo, a proposta do presente trabalho é a de mensurar a acurácia de um sistema de reconhecimento de locutores baseado em características elementares, isto é, energias de sub-bandas de frequências, em associação com um classificador probabilístico, estudando a viabilidade de extraí-las das transições entre trechos vozeados e não-vozeados (TTVNV) dos sinais. Testes são realizados com diferentes quantidades de locutores e discurso fixado. A acurácia obtida nos testes variam de 20.18% a 92.53%. Os resultados obtidos são comparados e relatados, complementando as afirmações existentes na literatura sobre o uso das TTVNV com dados quantitativos.Nowadays, many works and applications are developed focusing on computational speaker recognition. As the interest for several real applications within this area emerges, especially in biometrics, where the safety and the efficacy of the applications are extremely important, studies need to be developed in the same proportion, to evaluate the effectiveness of such approaches. Based on that, this work intends to measure the accuracy of a speaker recognition system that uses elementar features, i.e., sub-band frequency energies, associated with a probabilistic classifier, studying the viability of extracting them from the transition between voiced and unvoiced speech tags (TTVNV). Tests are carried out with different numbers of speakers and a text-dependent approach. The accuracy of the tests varies from 20.18% to 92.53%. The results are compared and reported, complementing the existent information on the use of TTVNV with quantitative data.
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Frenander, Hannes. "High-order finite difference approximations for hyperbolic problems : multiple penalties and non-reflecting boundary conditions." Doctoral thesis, Linköpings universitet, Beräkningsmatematik, 2017. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-134127.
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In this thesis, we use finite difference operators with the Summation-By-Partsproperty (SBP) and a weak boundary treatment, known as SimultaneousApproximation Terms (SAT), to construct high-order accurate numerical schemes.The SBP property and the SAT’s makes the schemes provably stable. The numerical procedure is general, and can be applied to most problems, but we focus on hyperbolic problems such as the shallow water, Euler and wave equations. For a well-posed problem and a stable numerical scheme, data must be available at the boundaries of the domain. However, there are many scenarios where additional information is available inside the computational domain. In termsof well-posedness and stability, the additional information is redundant, but it can still be used to improve the performance of the numerical scheme. As a first contribution, we introduce a procedure for implementing additional data using SAT’s; we call the procedure the Multiple Penalty Technique (MPT). A stable and accurate scheme augmented with the MPT remains stable and accurate. Moreover, the MPT introduces free parameters that can be used to increase the accuracy, construct absorbing boundary layers, increase the rate of convergence and control the error growth in time. To model infinite physical domains, one need transparent artificial boundary conditions, often referred to as Non-Reflecting Boundary Conditions (NRBC). In general, constructing and implementing such boundary conditions is a difficult task that often requires various approximations of the frequency and range of incident angles of the incoming waves. In the second contribution of this thesis,we show how to construct NRBC’s by using SBP operators in time. In the final contribution of this thesis, we investigate long time error bounds for the wave equation on second order form. Upper bounds for the spatial and temporal derivatives of the error can be obtained, but not for the actual error. The theoretical results indicate that the error grows linearly in time. However, the numerical experiments show that the error is in fact bounded, and consequently that the derived error bounds are probably suboptimal.
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Abel, John Trevor. "Development of a CubeSat Instrument for Microgravity Particle Damper Performance Analysis." DigitalCommons@CalPoly, 2011. https://digitalcommons.calpoly.edu/theses/537.
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Spacecraft pointing accuracy and structural longevity requirements often necessitate auxiliary vibration dissipation mechanisms. However, temperature sensitivity and material degradation limit the effectiveness of traditional damping techniques in space. Robust particle damping technology offers a potential solution, driving the need for microgravity characterization. A 1U cubesat satellite presents a low cost, low risk platform for the acquisition of data needed for this evaluation, but severely restricts available mass, volume, power and bandwidth resources. This paper details the development of an instrument subject to these constraints that is capable of capturing high resolution frequency response measurements of highly nonlinear particle damper dynamics.
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Bouchikhi, Abdelkhalek. "Analyse des signaux AM-FM par Transformation d'Huang Teager: application à l'acoustique sous marine." Phd thesis, Université Rennes 1, 2010. http://tel.archives-ouvertes.fr/tel-00818032.
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La Décomposition Modale Empirique (EMD) est un outil de traitement de signal piloté par les données et dédié aux signaux non-stationnaires issus ou non de systèmes linéaires. L'idée de base de l'EMD est l'interpolation des extrema par des splines pour extraire de composantes oscillantes appelées modes empiriques intrinsèques (IMFs) et un résidu. Dans cette thèse, un nouvel algorithme de l'EMD est introduit où au lieu d'une interpolation rigide, un lissage est utilisé pour la construction des enveloppes supérieures et inférieures du signal à décomposer. Ce nouvel algorithme est plus robuste au bruit que l'EMD conventionnelle et réduit le nombre d'IMFs "artificielles" (sur-décomposition). En combinant le nouvel algorithme et la méthode de séparation d'énergie (ESA) basée sur l'Opérateur d'Energie de Teager-Kaiser (OETK), un nouveau schéma de démodulation des signaux AM-FM multi-composante appelé EMD-ESA est introduit. Différentes versions de l'EMD-ESA sont analysées en terme de performance. Pour l'analyse Temps-Fréquence (TF), une nouvelle formulation de la carte TF de l'EMD-ESA appelée Transformation de Teager-Huang (THT) est présentée. Cette nouvelle Représentation TF (RTF) ne présentant pas de termes d'interférences est comparée aux RTF classiques telles que le spectrogramme, le scalogramme, la distribution de Wigner-Ville Distribution (WVD), la Pseudo-WVD et la réallocation de la Pseudo-WVD. En combinant la nouvelle formulation de la THT et la transformée de Hough, une nouvelle méthode de détection des signaux multi-composante à modulation linéaire de fréquence dans le plan TF est présentée. Cette méthode de détection est appelée transformation de Teager-Huang-Hough (THHT). Les résultats de la THHT sont comparés à ceux de la transformée WVD-Hough. Finalement, l'analyse TF par THT et par des RTF classiques (WVD, spectrogramme, etc.) de signaux réels de rétrodiffusion par des coques cylindriques de dimensions et de caractéristiques physiques différentes est présentée. Les résultats obtenus montrent l'apport de la THT comme un outil TF.
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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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Cexus, Jean-Christophe. "Analyse des signaux non-stationnaires par transformation de Huang, opérateur de Teager-Kaiser et transformation de Huang-Teager (THT)." Rennes 1, 2005. http://www.theses.fr/2005REN1S124.
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L'objectif de cette thèse est le traitement et l'analyse des signaux non-stationnaires, multi-composantes. Nous proposons de nouveaux outils de filtrage et de débruitage basés sur la Transformation de Huang (ou Décomposition modale empirique : EMD). Un nouvel opérateur d'énergie pour l'analyse temporelle (démodulation, détection, interaction, similarité,. . . ) des signaux est introduit. Cet opérateur généralise celui de Teager-Kaiser (TK). Nous établissons les liens théoriques entre cet opérateur et les représentations temps-fréquence de la classe de Cohen. Pour l'analyse temps-fréquence, nous introduisons une nouvelle méthode basée sur l'utilisation conjointe de l'EMD et de l'opérateur de TK : la Transformation de Huang-Teager (THT). Pour illustrer ces concepts, des résultats de filtrage, de débruitage, de détection, d'analyse temps-fréquence de signaux sont présentés. Nous terminons par l'analyse et classification des échos de cibles sonars par THT.
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Socheleau, François-Xavier. "Communications acoustiques sous-marines sur canal fortement dispersif en temps et en fréquence : point de vue de la théorie de l'information." Phd thesis, Université de Bretagne occidentale - Brest, 2011. http://tel.archives-ouvertes.fr/tel-00638836.
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Le canal acoustique sous-marin (ASM) est réputé comme difficile. Il présente à la fois des effets de réflexion/réfraction générant de la dispersion temporelle importante, une atténuation fortement croissante avec la fréquence qui restreint la bande-passante disponible, ainsi qu'une forte variabilité du milieu, qui, combinée avec une faible célérité des ondes acoustiques, induit une dispersion fréquentielle conséquente. Contrairement à la recherche dans le domaine des radio-communications qui repose principalement sur des modèles de propagations bien établis, la recherche en communication ASM ne dispose pas de tels modèles et s'appuie majoritairement sur des expérimentations en mer. L'objectif principal de ce travail de thèse est de montrer que malgré notre connaissance limitée sur la réalité physique du canal, il est possible, en utilisant des outils théoriques adaptés, de déterminer en laboratoire des directives pertinentes aussi bien pour la sélection des modulations et la validation des modems correspondants que pour les prédictions de débit. Dans ce contexte, la théorie de l'information s'avère être un outil très précieux pour inférer un maximum de conclusions sur le canal et induire des directives pratiques sur les formats de modulation. Nous développons dans un premier temps un point de vue de théorie de l'information sur la modélisation de canaux acoustiques sous-marins doublement dispersifs en s'appuyant sur le principe d'entropie maximale. Dans un cadre d'inférence plus empirique, nous proposons ensuite un modèle de canal alimenté par des données réelles qui repose sur l'hypothèse selon laquelle une réponse impulsionnelle de canal mesurée en mer n'est qu'une observation d'un processus aléatoire sous-jacent. Nous dérivons dans un troisième temps de nouvelles bornes de capacité sur le canal ASM en considérant des hypothèses plus réalistes que l'état de l'art sur le sujet : le canal est ici supposé doublement dispersif, la puissance crête limitée et la réalisation courante du canal inconnue de l'émetteur et du récepteur (contexte non-cohérent). Enfin, nous nous intéressons à la structure "propre" du canal et cherchons à optimiser la forme d'onde de transmission pour trouver un bon compromis entre les notions très intriquées de robustesse et de débit qui caractérisent les performances des systèmes de communication.
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Koseli, Volkan. "Experimental And Theoretical Investigation Of Complex Flows By Ultrasound Doppler Velocimetry." Phd thesis, METU, 2009. http://etd.lib.metu.edu.tr/upload/12610727/index.pdf.
Full textAbstract:
Non-invasive and fast flow measurement techniques have had increasing importance
for the last decades. Scientists are looking for such quick techniques to be able to
monitor real velocities without disturbing flow itself. Ultrasound Doppler
velocimetry (UDV) being one of such techniques promising with advantages of
getting simultaneous velocity measurements from several points and of applicability
for opaque liquids as well. UDV is a technique which is still being developed for
new applications and analysis of complex flows.
In this study effect of sinusoidal oscillating, turbulent (random) and viscoelastic fluid
motions on UDV signals were investigated theoretically and experimentally.
Obtained mathematical relations for random and viscoelastic motions were utilized
to get statistics of flow and distribution of relaxation spectrum, respectively.
Analytical analysis and numerical simulation of sinusoidal oscillating flow depicted
that there is a critical value for the ratio of oscillation amplitude to oscillation
frequency for a specified set of measurement parameters of UDV. Above this critical
value UDV is not successful to determine mean flow velocity. Mathematical
relations between velocity probability density function (PDF) &ndashvelocity auto correlation function (ACF) and UDV signal spectrum were obtained in the analysis v of flow with random velocity. Comparison of velocity ACFs from direct velocity measurements and from raw in-phase (I) and quadrature (Q) signals through derived relation, revealed that time resolution of UDV technique is not enough for getting a good velocity ACF and thus turbulence spectrum. Using I and Q signals rather than measured velocities to get velocity ACF, increased the time resolution in the order of number of pulses used for getting one velocity value (Nprn). Velocity PDF obtained from UDV spectrum was compared with the one obtained from measured velocities with the assumption of Gaussian PDF. Both velocity PDFs were consistent. Also some parameters of pipe turbulence from literature were compared with the presented findings from velocity ACF obtained from I and Q signals through derived relation. Results showed good compatibility. In the last part of the study, complex viscosity of a linear viscoelastic fluid mathematically related to spectrum of UDV for a pipe flow with small-amplitude oscillating pressure field. Generalized Maxwell model was employed to express complex viscosity terms. Zero frequency (mean flow) component of UDV spectrum was used to obtain an equation for relaxation viscosities of generalized Maxwell model. Results have revealed that UDV technique can also be used to probe some of viscoelastic material functions. In conclusion, UDV is relatively new but a promising technique for the measurement and analysis of complex flows in a non-invasive manner.
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Laude, Vincent. "Contributions au traitement optique du signal et aux ondes élastiques guidées." Habilitation à diriger des recherches, Université de Franche-Comté, 2002. http://tel.archives-ouvertes.fr/tel-00005804.
Full textAbstract:
Ce document présente une synthèse de mes travaux de recherche postérieurement à la thèse de doctorat. Depuis le début de mes travaux de thèse en 1992 et jusqu'au début de l'année 2000, mes recherches ont été effectuées dans le domaine du traitement optique du signal. Depuis, mes principales activités de recherche concernent le domaine de la microsonique, et plus précisément des ondes élastiques guidées ou de surface. Je présente successivement mes contributions à ces deux domaines, puis les perspectives de cette thématique de recherche, et les directions que j'entends prendre dans les années à venir. A partir de 1995, à la suite de ma thèse, j'ai poursuivi mes travaux en corrélation optique. Je me suis intéressé aux aspects théoriques de la corrélation optique, ainsi qu'à des moyens pratiques d'utiliser les solutions optimales obtenues dans les corrélateurs existants. J'ai également étudié de nouvelles applications des modulateurs spatiaux de lumière pour le traitement optique du signal, et plus particulièrement des systèmes originaux d'imagerie programmable et un capteur de front d'onde à balayage. Par la suite, je me suis intéressé au traitement des impulsions laser ultrabrèves, domaine dans lequel le contrôle de la forme temporelle des impulsions présente un intérêt considérable. J'ai étudié plus particulièrement des moyens de contrôler et de mesurer la dispersion des impulsions ultrabrèves. J'ai également mené des études sur la possibilité d'obtenir des temps de groupe superluminaux pour les ondes optiques. Dans le domaine de la microsonique, mes travaux sont orientés vers la compréhension de la propagation des ondes électro-acoustiques (liées aux milieux piézoélectriques) dans les microstructures.
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Monks, T. "Acoustic convolvers for analogue signal processing." Thesis, University of Oxford, 1985. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.355785.
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Pomponi, Eraldo. "Advanced signal processing in acoustic emission." Doctoral thesis, Università Politecnica delle Marche, 2009. http://hdl.handle.net/11566/242269.
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