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1

Alghassi, Hedayat. "Eye array sound source localization." Thesis, University of British Columbia, 2008. http://hdl.handle.net/2429/5114.

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Sound source localization with microphone arrays has received considerable attention as a means for the automated tracking of individuals in an enclosed space and as a necessary component of any general-purpose speech capture and automated camera pointing system. A novel computationally efficient method compared to traditional source localization techniques is proposed and is both theoretically and experimentally investigated in this research. This thesis first reviews the previous work in this area. The evolution of a new localization algorithm accompanied by an array structure for audio sign
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Prescott, Tom. "Sound design, composition and performance with interactive genetic algorithms." Thesis, Keele University, 2018. http://eprints.keele.ac.uk/5030/.

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A variety of work has been carried out investigating the suitability of interactive genetic algorithms (IGAs) for musical composition. There have been some promising results demonstrating that it is, in principle, an effective approach. Modern sound synthesis and processing techniques (SSPTs) are often very complex and difficult to use. They often consist of tens or hundreds of parameters and a large range of values can be assigned to each parameter. This results in an immense number of parameter combinations; listening to the result of each one is clearly not viable. Furthermore, the effect e
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Markle, Blake L. "A comparative study of time-stretching algorithms for audio signals /." Thesis, McGill University, 2001. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=31119.

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Algorithms exist which will perform independent transformations on frequency or duration of a digital audio signal. These processes have different results different types of audio signals. A comparative study of granular and phase vocoder algorithms, implementation, and their respective effects on audio signals was made to determine which algorithm is best suited to a particular type of audio signal.
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4

Nalavolu, Praveen Reddy. "PERFORMANCE ANALYSIS OF SRCP IMAGE BASED SOUND SOURCE DETECTION ALGORITHMS." UKnowledge, 2010. http://uknowledge.uky.edu/gradschool_theses/50.

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Steered Response Power based algorithms are widely used for finding sound source location using microphone array systems. SRCP-PHAT is one such algorithm that has a robust performance under noisy and reverberant conditions. The algorithm creates a likelihood function over the field of view. This thesis employs image processing methods on SRCP-PHAT images, to exploit the difference in power levels and pixel patterns to discriminate between sound source and background pixels. Hough Transform based ellipse detection is used to identify the sound source locations by finding the centers of elliptic
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5

Khan, Muhammad Salman. "Informed algorithms for sound source separation in enclosed reverberant environments." Thesis, Loughborough University, 2013. https://dspace.lboro.ac.uk/2134/13350.

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While humans can separate a sound of interest amidst a cacophony of contending sounds in an echoic environment, machine-based methods lag behind in solving this task. This thesis thus aims at improving performance of audio separation algorithms when they are informed i.e. have access to source location information. These locations are assumed to be known a priori in this work, for example by video processing. Initially, a multi-microphone array based method combined with binary time-frequency masking is proposed. A robust least squares frequency invariant data independent beamformer designed w
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Virgulti, Marco. "Advancend algorithms for sound reproduction enhancement in portable multimedia devices." Doctoral thesis, Università Politecnica delle Marche, 2015. http://hdl.handle.net/11566/243003.

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In questa tesi e' stato svolto uno studio sulle problematiche relative alla riproduzione audio nel contesto dei sistemi mobile. Le tematiche principali di studio riguardano la spazializzazione, il miglioramento della riproduzione delle basse frequenze e tecniche avanzate di equalizzazione acustica ambientale. Il lavoro puo essere diviso in due parti. Nella prima parte, e' stato illustrato un sistema acustico integrato per sistemi mobili. Le difficolta incontrate in questo contesto sono dovute essenzialmente alle ridotte capacita di calcolo e dalle dimensioni fisiche degli altoparlanti. Lo
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Burt, Warren. "Algorithms, microtonality, performance eleven musical compositions /." Access electronically, 2007. http://www.library.uow.edu.au/adt-NWU/public/adt-NWU20080131.162859/index.html.

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Thesis (Ph.D.)--University of Wollongong, 2007.<br>Typescript. Includes 2 sound discs and 1 DVD-ROM in back pocket. CD 1: The animation of lists; CD 2: And the archytan transpositions. DVD-ROM contains Part Three - Appendix. Includes bibliographical references: leaf 291-301.
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Cousins, David Bruce. "A model-based algorithm for environmentally adaptive bathymetry and sound velocity profile estimation /." View online ; access limited to URI, 2005. http://0-wwlib.umi.com.helin.uri.edu/dissertations/dlnow/3186901.

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Roman, Nicoleta. "Auditory-based algorithms for sound segregation in multisource and reverberant environments." Connect to resource, 2005. http://rave.ohiolink.edu/etdc/view?acc%5Fnum=osu1124370749.

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Thesis (Ph. D.)--Ohio State University, 2005.<br>Title from first page of PDF file. Document formatted into pages; contains i-xxii, xx-xxi, 183 p.; also includes graphics. Includes bibliographical references (p. 171-183). Available online via OhioLINK's ETD Center
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Wang, Xun. "Sound source localization with data and model uncertainties using the EM and Evidential EM algorithms." Thesis, Compiègne, 2014. http://www.theses.fr/2014COMP2164/document.

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Ce travail de thèse se penche sur le problème de la localisation de sources acoustiques à partir de signaux déterministes et aléatoires mesurés par un réseau de microphones. Le problème est résolu dans un cadre statistique, par estimation via la méthode du maximum de vraisemblance. La pression mesurée par un microphone est interprétée comme étant un mélange de signaux latents émis par les sources. Les positions et les amplitudes des sources acoustiques sont estimées en utilisant l’algorithme espérance-maximisation (EM). Dans cette thèse, deux types d’incertitude sont également pris en compte :
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Vidal, Rosemeire Cardozo. "Algoritmo para estimar gravidade de DPOC através de sinais acústicos." Universidade de São Paulo, 2017. http://www.teses.usp.br/teses/disponiveis/3/3152/tde-11072017-151033/.

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O presente estudo tem como objetivo determinar se a gravidade da DPOC poderá ser estimada através da área do gráfico das intensidades sonoras dos sons respiratórios de pacientes com DPOC. O estudo realizado com 51 pacientes com DPOC leve, moderado, grave, muito grave e 7 indivíduos saudáveis não fumantes. Os sons respiratórios de cada participante, foram coletados através de estetoscópio adaptado com um mini microfone. O método compara as áreas das intensidades sonoras em função da frequência de pacientes de DPOC e indivíduos saudáveis. Neste contexto, para atender ao objetivo, um método foi p
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SUMME, LORI ANN. "Noise Reduction in Digital Hearing Aids Using Environmental Sounds." University of Cincinnati / OhioLINK, 2003. http://rave.ohiolink.edu/etdc/view?acc_num=ucin1052327314.

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13

Dhakal, Parashar. "Novel Architectures for Human Voice and Environmental Sound Recognitionusing Machine Learning Algorithms." University of Toledo / OhioLINK, 2018. http://rave.ohiolink.edu/etdc/view?acc_num=toledo1531349806743278.

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14

Magnus, Cristyn. "Evolutionary sound a non-symbolic approach to creating sonic art with genetic algorithms /." Diss., [La Jolla, Calif.] : University of California, San Diego, 2010. http://wwwlib.umi.com/cr/ucsd/fullcit?p3396736.

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Thesis (Ph. D.)--University of California, San Diego, 2010.<br>Title from 1st page of PDF file (viewed Mar. 9, 2010). Available via ProQuest Digital Dissertations. Vita. Includes 3 compact discs containing audio examples for the dissertation. Includes bibliographical references: P. 137-142.
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Berquin, Yann. "Assessing the performances and optimizing the radar sounder design parameters for the EJSM mission (Ganymede and Europa)." Thesis, Grenoble, 2014. http://www.theses.fr/2014GRENU001/document.

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On se propose dans un premier temps d'étudier des jeux de données topographiques sur la lune glacée de Jupiter Ganymède et d'estimer l'impact de la topographie sur les performances du futur radar sondeur. Les principaux résultats sont présentés dans [1]. Une seconde partie est dédiée à l'expression mathématique du problème direct du sondage radar planétaire (physique et instrumentation). On rappelle ainsi comment dériver à partir des formulations de Stratton-Chu les formulations volumiques classiques et surfaciques (i.e. Huygens-Fresnel). On s'attache ensuite à détailler un algorithme performa
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Yong, Stephen. "An algorithm to recover the left-ventricular intracardiac second heart sound from externally recorded phonocardiograms /." Thesis, McGill University, 1990. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=59601.

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Heart sounds recorded on the surface of the thorax can be significantly distorted due to the varying properties of the transmission paths between the heart chambers and body surface. For diagnostic purposes, it is desirable to obtain the sounds at the source. In this thesis, an algorithm has been developed to recover the left-ventricular component of the second heart sound from externally recorded phonocardiograms. The algorithm uses a model of the heart-thorax system and is derived from deconvolution techniques developed for analogous problems in speech processing and seismology. The algorith
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Wang, Xipeng. "CONSTANT FALSE ALARM RATE PERFORMANCE OF SOUND SOURCE DETECTION WITH TIME DELAY OF ARRIVAL ALGORITHM." UKnowledge, 2017. http://uknowledge.uky.edu/ece_etds/105.

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Time Delay of Arrival (TDOA) based algorithms and Steered Response Power (SRP) based algorithms are two most commonly used methods for sound source detection and localization. SRP is more robust under high reverberation and multi-target conditions, while TDOA is less computationally intensive. This thesis introduces a modified TDOA algorithm, TDOA delay table search (TDOA-DTS), that has more stable performance than the original TDOA, and requires only 4% of the SRP computation load for a 3-dimensional space of a typical room. A 2-step adaptive thresholding procedure based on a Weibull noise pe
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Mainwaring, David, and Jonathan Österberg. "Sound Pattern Recognition : Evaluation of Independent Component Analysis Algorithms for Separation of Voices." Thesis, KTH, Skolan för teknikvetenskap (SCI), 2018. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-230746.

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With computers being used for more applications where commands can be spoken it is useful to findalgorithms which can separate voices from each other so that software can turn spoken words intocommands. In this paper our goal is to describe how Independent Component Analysis (ICA) can beused for separation of voices in cases where we have at least the same number of microphones, atdifferent distances from the speakers, as speakers whose voices we wish to separate, the so called``cocktail party problem". This is done by implementing an ICA algorithm on voice recordingscontaining multiple person
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Tsalaile, Thato K. "Digital signal processing algorithms and techniques for the enhancement of lung sound measurements." Thesis, Loughborough University, 2008. https://dspace.lboro.ac.uk/2134/34576.

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Lung sound signal (LSS) measurements are taken to aid in the diagnosis of various diseases. Their interpretation is difficult however due to the presence of interference generated by the heart. Novel digital signal processing techniques are therefore proposed to automate the removal of the heart sound signal (HSS) interference from the LSS measurements. The HSS is first assumed to be a periodic component so that an adaptive line enhancer can be exploited for the mitigation of the HSS interference. The utility of the scheme is verified on synthetic signals, however its performance is found to b
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Cambou, Pierre E. "A Distributed Active Vibration Absorber (DAVA) for Active-Passive Vibration and Sound Radiation Control." Thesis, Virginia Tech, 1998. http://hdl.handle.net/10919/10105.

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This thesis presents a new active-passive treatment developed to reduce structural vibrations and their associated radiated sound. It is a contribution to the research of efficient and low cost devices that implement the advantages of active and passive noise control techniques. A theoretical model has been developed to investigate the potential of this new "active-passive distributed absorber". The model integrates new functions that make it extremely stable numerically. Using this model, a genetic algorithm has been used to optimize the shape of the active-passive distributed absorber. Proto
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21

Souaille, Ava. "Conception Interactive en Design Sonore." Electronic Thesis or Diss., Ecole centrale de Nantes, 2023. http://www.theses.fr/2023ECDN0009.

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La problématique de cette thèse porte sur la définition d’outils et de méthodes interactives pour le design sonore, permettant d'intégrer l’expérience utilisateur dans le processus de conception. Nous proposons un paradigme expérimental d'étude de laperception sonore, basé l'optimisation interactive de sons. En particulier, nous utilisons des Algorithmes Génétiques Interactifs (IGAs) pour résoudre deux problèmes de design sonore: le design de sons d'alerte de véhicules électriques ou hybrides facilement détectables et peu désagréables, ainsi que le design de sons d'intérieur peu intrusifs pour
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22

Feng, Tao. "Design and Analysis of Efficient Adaptive Algorithms for Active Control of Vehicle Interior Sound." University of Cincinnati / OhioLINK, 2017. http://rave.ohiolink.edu/etdc/view?acc_num=ucin1490354549915601.

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Spadaccini, Andrea. "Innovative Traits, Algorithms and Application Scenarios for Mono-multimodal Biometric Recognition." Doctoral thesis, Università di Catania, 2012. http://hdl.handle.net/10761/1096.

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Identity verification is one of the most common and important processes in our daily lives. For centuries, humans have relied on the visual appearance of their peers - or other distinctive traits - to recognize them. Currently, most of the traditional authentication methods infer one s identity by either verifying the knowledge of a shared secret (i.e., password) or the possession of a given object (i.e., token); both methods are sub-optimal, since they do not directly verify the person s identity, but an alternate and less rich representation that could also very easily stolen or inadvertentl
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KUNTE, DEEPTI SHRIRAM. "Sound Quality Prediction Using Neural Networks." Thesis, KTH, Skolan för industriell teknik och management (ITM), 2020. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-283336.

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Sound quality is an important measure depicting the quality of a machine as well as the comfort in its usage. However, it being a subjective measure, not only is it difficult to capture it ahead of time but also necessitates time and cost expensive jury testing. Thus, it is worthwhile to be able to effectively predict the results of the jury study from metrics that can be objectively measured. The aim of the thesis is twofold: first, to establish neural network models to predict subjective sound quality metrics from objective metrics and second, to interpret the model to understand the relativ
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Kern, Alexander Marco. "Quantification of the performance of 3D sound field reconstruction algorithms using high-density loudspeaker arrays and 3rd order sound field microphone measurements." Thesis, Virginia Tech, 2017. http://hdl.handle.net/10919/77516.

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The development and improvement of 3-D immersive audio is gaining momentum through the growing interest in virtual reality. Possible applications reach from recreating real world environments to immersive concerts and performances to exploiting big data acoustically. To improve the immersive experience several measures can be taken. The recording of the sound field, the spatialization and the development of the loudspeaker arrays are some of the greatest challenges. In this thesis, these challenges for improving immersive audio will be explored. First, there will be a short introduction about
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Zarate, Orozco Ismael. "Software and Hardware-In-The-Loop Modeling of an Audio Watermarking Algorithm." Thesis, University of North Texas, 2010. https://digital.library.unt.edu/ark:/67531/metadc33221/.

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Due to the accelerated growth in digital music distribution, it becomes easy to modify, intercept, and distribute material illegally. To overcome the urgent need for copyright protection against piracy, several audio watermarking schemes have been proposed and implemented. These digital audio watermarking schemes have the purpose of embedding inaudible information within the host file to cover copyright and authentication issues. This thesis proposes an audio watermarking model using MATLAB® and Simulink® software for 1K and 2K fast Fourier transform (FFT) lengths. The watermark insertion proc
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Puikkonen, Panu Tapani. "Development of an Adaptive Equalization Algorithm Using Acoustic Energy Density." BYU ScholarsArchive, 2009. https://scholarsarchive.byu.edu/etd/1686.

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Sound pressure equalization of audio signals using digital signal processors has been a subject of ongoing study for many years. The traditional approach is to equalize sound at a point in a listening environment, but because of its specific dependence on the room frequency response between a source and receiver position, this equalization generally causes the spectral response to worsen significantly at other locations in the room. This work presents both a time-invariant and a time-varying implementation of an adaptive acoustic energy density equalization filter for a one-dimensional sound f
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Trinkaus, Trevor R. "Perceptual coding of audio and diverse speech signals." Diss., Georgia Institute of Technology, 1999. http://hdl.handle.net/1853/13883.

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Chen, Howard. "AZIP, audio compression system: Research on audio compression, comparison of psychoacoustic principles and genetic algorithms." CSUSB ScholarWorks, 2005. https://scholarworks.lib.csusb.edu/etd-project/2617.

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The purpose of this project is to investigate the differences between psychoacoustic principles and genetic algorithms (GA0). These will be discussed separately. The review will also compare the compression ratio and the quality of the decompressed files decoded by these two methods.
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Hrabina, Martin. "VÝVOJ ALGORITMŮ PRO ROZPOZNÁVÁNÍ VÝSTŘELŮ." Doctoral thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2019. http://www.nusl.cz/ntk/nusl-409087.

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Táto práca sa zaoberá rozpoznávaním výstrelov a pridruženými problémami. Ako prvé je celá vec predstavená a rozdelená na menšie kroky. Ďalej je poskytnutý prehľad zvukových databáz, významné publikácie, akcie a súčasný stav veci spoločne s prehľadom možných aplikácií detekcie výstrelov. Druhá časť pozostáva z porovnávania príznakov pomocou rôznych metrík spoločne s porovnaním ich výkonu pri rozpoznávaní. Nasleduje porovnanie algoritmov rozpoznávania a sú uvedené nové príznaky použiteľné pri rozpoznávaní. Práca vrcholí návrhom dvojstupňového systému na rozpoznávanie výstrelov, monitorujúceho ok
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Se, Rin Oh. "Figuration for Piano and Electronic Sounds." University of Cincinnati / OhioLINK, 2021. http://rave.ohiolink.edu/etdc/view?acc_num=ucin162324126780976.

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Okuyucu, Cigdem. "Semantic Classification And Retrieval System For Environmental Sounds." Master's thesis, METU, 2012. http://etd.lib.metu.edu.tr/upload/12615114/index.pdf.

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The growth of multimedia content in recent years motivated the research on audio classification and content retrieval area. In this thesis, a general environmental audio classification and retrieval approach is proposed in which higher level semantic classes (outdoor, nature, meeting and violence) are obtained from lower level acoustic classes (emergency alarm, car horn, gun-shot, explosion, automobile, motorcycle, helicopter, wind, water, rain, applause, crowd and laughter). In order to classify an audio sample into acoustic classes, MPEG-7 audio features, Mel Frequency Cepstral Coefficients
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Padden, Dereck J. "Recursive modeling of interpositional transfer functions with a genetic algorithm aided by an adaptive filter for the purpose of altering free-field sound localization /." Online version of thesis, 2007. http://hdl.handle.net/1850/4484.

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Lui, Siu-Hang. "MIDI to SP-MIDI and I-melody transcoding using phrase stealing /." View abstract or full-text, 2005. http://library.ust.hk/cgi/db/thesis.pl?COMP%202005%20LUI.

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Deprez, Romain. "Optimisation perceptive de la restitution sonore multicanale par une analyse spatio-temporelle des premières réflexions." Thesis, Aix-Marseille, 2012. http://www.theses.fr/2012AIXM4746/document.

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L'objectif de cette thèse est l'optimisation de la qualité perçue de la reproduction sonore par un système audio multicanal, dans un contexte de salle d'écoute domestique. Les travaux de recherche présentés s'articulent selon deux axes. Le premier concerne l'effet de salle, et plus particulièrement les aspects physiques et perceptifs liés aux premières réflexions d'une salle. Ces éléments sont décrits spécifiquement, et une expérience psychoacoustique a été menée afin d'étendre les données disponibles quant à leur perceptibilité, c'est à dire leur capacité à modifier la perception du son direc
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Luck, Rodney K. "On the use of two-dimensional orthogonal function expansions to model ocean bathymetric and sound-speed data in the recursive ray acoustics algorithm." Thesis, Monterey, Calif. : Springfield, Va. : Naval Postgraduate School ; Available from National Technical Information Service, 1995. http://handle.dtic.mil/100.2/ADA303056.

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Johansson, Sven. "Active Control of Propeller-Induced Noise in Aircraft : Algorithms & Methods." Doctoral thesis, Karlskrona, Ronneby : Blekinge Institute of Technology, 2000. http://urn.kb.se/resolve?urn=urn:nbn:se:bth-00171.

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In the last decade acoustic noise has become more and more regarded as a problem. In cars, boats, trains and aircraft, low-frequency noise reduces comfort. Lightweight materials and more powerful engines are used in high-speed vehicles, resulting in a general increase in interior noise levels. Low-frequency noise is annoying and during periods of long exposure it causes fatigue and discomfort. The masking effect which low-frequency noise has on speech reduces speech intelligibility. Low-frequency noise is sought to be attenuated in a wide range of applications in order to improve comfort and s
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Koutsomichalis, Marinos G. "A hypermedia and project-based approach to music, sound and media art." Thesis, De Montfort University, 2015. http://hdl.handle.net/2086/11432.

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This thesis describes my artistic practice as essentially project-based, site-responsive and hypermediating. Hypermediacy—i.e. the tendency of certain media or objects to keep their various constituents separate from their structure—is to be understood as opaque, juxtaposed and after a recurring contiguity with different kinds of interfaces. Accordingly, and within the context of the various projects that constitute this thesis, it is demonstrated how, in response to the particular places I work and to the various people I collaborate with, different kinds of materials and methodologies are in
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Sy, Djibril. "Modélisation et optimisation des performances acoustiques d'un tablier d'automobile en alliage de magnésium." Mémoire, Université de Sherbrooke, 2010. http://savoirs.usherbrooke.ca/handle/11143/1589.

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Résumé : Ce projet fait partie du projet MFERD (Magnésium Front End Research and Development) qui vise à développer les technologies permettant de rendre les alliages de Magnésium (Mg) comme un principal matériau structural pour les voitures (aujourd'hui essentiellement constituées d'acier quatre fois plus lourd que le Mg) afin d'en réduire leur masse pour des raisons environnementales et sécuritaires. Dans ce travail de maîtrise nous avons regardé la partie acoustique dans le cas d'un tablier (structure métallique derrière le tableau de bord) en magnésium. En effet, le confort acoustique
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Anderson, Monty J. "Active Control of the Human Voice from a Sphere." BYU ScholarsArchive, 2015. https://scholarsarchive.byu.edu/etd/5295.

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This work investigates the application of active noise control (ANC) to speech. ANC has had success reducing tonal noise. In this work, that success was extended to noise that is not completely tonal but has some tonal elements such as speech. Limitations such as causality were established on the active control of human speech. An optimal configuration for control actuators was developed for a sphere using a genetic algorithm. The optimal error sensor location was found from exploring the nulls associated with the magnitude of the radiated pressure with reference to the primary pressure field.
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Kersten, Stefan. "Statistical modelling and resynthesis of environmental texture sounds." Doctoral thesis, Universitat Pompeu Fabra, 2016. http://hdl.handle.net/10803/400395.

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Environmental texture sounds are an integral, though often overlooked, part of our daily life. They constitute those elements of our sounding environment that we tend to perceive subconsciously but which we miss when they are missing. Those sounds are also increasingly important for adding realism to virtual environments, from immersive artificial worlds through computer games to mobile augmented reality systems. This work spans the spectrum from data-driven stochastic sound synthesis methods to distributed virtual reality environments and their aesthetic and technological implications. We pro
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Krysander, Mattias. "Design and Analysis of Diagnosis Systems Using Structural Methods." Doctoral thesis, Linköping : Linköpings universitet, 2006. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-7417.

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Kumar, Gaurav. "Structual-acoustic properties of automotive panels with shell elements." Thesis, Loughborough University, 2014. https://dspace.lboro.ac.uk/2134/13950.

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The automotive industry has witnessed a trend in the recent years of reducing the bulk weight of the vehicle in order to achieve improved ride dynamics and economical fuel consumption. Unfortunately, reducing the bulk weight often compromises the noise, vibra- tion, and harshness (NVH) characteristics of the vehicle. In general, the automotive body panels are made out of thin sheet metals (steel and aluminium) that have a very low bend- ing stiffness. Hence, it becomes important to find countermeasures that will increase the structural stiffness of these thin body panels without affecting thei
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Garcia, Andrés Xavier. "Development of innovative methodologies to reduce railway rolling noise through Genetic Algorithm-based shape optimization techniques." Doctoral thesis, Universitat Politècnica de València, 2021. http://hdl.handle.net/10251/164194.

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[ES] El ruido de rodadura es un fenómeno producido debido a la interacción entre la rueda y el carril e inducido por las pequeñas irregularidades existentes en la superficie de ambos. Dichas irregularidades, conocidas como "rugosidad", provocan que se genere una vibración tanto en la rueda como el carril cuando el tren circula a una cierta velocidad, lo que consecuentemente lleva a la aparición de radiación acústica. Esto conforma una de las fuentes de molestias más relevantes y principal origen de la contaminación acústica producida por los trenes al circular por regiones urbanas densamente p
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45

Sankarasubramaniam, Yogesh. "New Capacity-Approaching Codes for Run-Length-Limited Channels." Diss., Georgia Institute of Technology, 2006. http://hdl.handle.net/1853/10483.

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Run-Length-Limited (RLL) channels are found in digital recording systems like the Hard Disk Drive (HDD), Compact Disc (CD), and Digital Versatile Disc (DVD). This thesis presents novel encoding algorithms for RLL channels based on a simple technique called bit stuffing. First, two new capacity-achieving variable-rate code constructions are proposed for (d,k) constraints. The variable-rate encoding ideas are then extended to (0,G/I) and other RLL constraints. Since variable-rate codes are of limited practical value, the second half of this thesis focuses on fixed-rate codes. The fixed-rate bit
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46

Заїка, В. В. "Оброблення аудіо сигналів засобами графічних процесорів". Thesis, Сумський державний університет, 2014. http://essuir.sumdu.edu.ua/handle/123456789/39136.

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Останнім часом графічні процесори набули значного поширення як платформа для розв’язання обчислювальних завдань широкого спектру. Зокрема багато алгоритмів оброблення звуку є добре пристосованими для паралельних обчислень. Мета дослідження – визначення цифрових аудіо алгоритмів, які найкраще піддаються паралелізації, та дозволяють зменшити навантаження на центральний процесор за рахунок перенесення їх виконання на графічні ядра.
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47

Dey, Ajoy Kumar, and Susmita Saha. "Acoustic Beamforming : Design and Development of Steered Response Power With Phase Transformation (SRP-PHAT)." Thesis, Blekinge Tekniska Högskola, Sektionen för ingenjörsvetenskap, 2011. http://urn.kb.se/resolve?urn=urn:nbn:se:bth-3226.

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Acoustic Sound Source localization using signal processing is required in order to estimate the direction from where a particular acoustic source signal is coming and it is also important in order to find a soluation for hands free communication. Video conferencing, hand free communications are different applications requiring acoustic sound source localization. This applications need a robust algorithm which can reliably localize and position the acoustic sound sources. The Steered Response Power Phase Transform (SRP-PHAT) is an important and roubst algorithm to localilze acoustic sound sourc
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Chiu, Leung Kin. "Efficient audio signal processing for embedded systems." Diss., Georgia Institute of Technology, 2012. http://hdl.handle.net/1853/44775.

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We investigated two design strategies that would allow us to efficiently process audio signals on embedded systems such as mobile phones and portable electronics. In the first strategy, we exploit properties of the human auditory system to process audio signals. We designed a sound enhancement algorithm to make piezoelectric loudspeakers sound "richer" and "fuller," using a combination of bass extension and dynamic range compression. We also developed an audio energy reduction algorithm for loudspeaker power management by suppressing signal energy below the masking threshold. In the second str
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Dickson, Crispin. "A few aspects of aircraft noise." Licentiate thesis, Stockholm : Teknisk akustik, Kungliga Tekniska högskolan, 2007. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-4510.

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50

Majvald, František. "Metody akustické holografie v blízkém poli v prostředí LabVIEW." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2021. http://www.nusl.cz/ntk/nusl-442539.

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Near-field acoustical holography is a standard method for sound source visualization and localization. In this thesis, commonly used and newly published near-field acoustic holography methods are introduced and analysed. In addition, regularization methods are presented together with finding options of optimal regularization parameter. Based on theory, a LabVIEW library is built containing four implemented near-field acoustical holography algorithms and two regularization methods. To verify the correctness of implementations, a testing application has been made. This application allows testing
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