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Academic literature on the topic 'SounBe'

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Published: 14 March 2023

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Journal articles on the topic "SounBe"

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Castro Solano, M. M. Otto. "La ciudad como fuente de sonidos para la creación sonora." Resonancias: Revista de investigación musical 42 (June 2018): 143–50. http://dx.doi.org/10.7764/res.2018.42.8.

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KIGURE, Tatsuya, Kenji AMAYA, and Yuki ONISHI. "1007 Sound Source Localization using probability distribution of medium property." Proceedings of The Computational Mechanics Conference 2011.24 (2011): 344–46. http://dx.doi.org/10.1299/jsmecmd.2011.24.344.

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Iwasaki, Nobuo, Katsuhiro Inoue, and Hiromu Gotanda. "A Real Time Oriented Sound Source DOA Estimation Based on Sparseness." Transactions of the Institute of Systems, Control and Information Engineers 27, no. 12 (2014): 493–500. http://dx.doi.org/10.5687/iscie.27.493.

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Xia Zhenjie, 夏振杰, 刘强 Liu Qiang, 李昂 Li Ang, 刘悦莹 Liu Yueying, 荆振国 Jing Zhenguo, and 彭伟 Peng Wei. "基于膜片式EFPI光纤麦克风的声源定位系统." Chinese Journal of Lasers 48, no. 9 (2021): 0910002. http://dx.doi.org/10.3788/cjl202148.0910002.

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Ih, Jeong-Guon. "Identification of the Sectional Distribution of Sound Source in a Wide Duct." Journal Of The Acoustical Society Of Korea 33, no. 2 (2014): 87. http://dx.doi.org/10.7776/ask.2014.33.2.087.

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Chaofeng Lan, Chaofeng Lan, Lei Zhang Chaofeng Lan, Shou Lv Lei Zhang, and Rongrong Han Shou Lv. "Study on Noise Control Effect of Secondary Sound Source Distribution in Vehicle Interior." 電腦學刊 32, no. 6 (December 2021): 248–53. http://dx.doi.org/10.53106/199115992021123206022.

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EDEN, Arda. "AÇIK KAYNAK VE ÖZGÜR YAZILIM HAREKETLERİ IŞIĞINDA GNU/LİNUX İLE SES VE MÜZİK." Akademik Müzik Araştırmaları Dergisi 3, no. 6 (June 5, 2017): 1–22. http://dx.doi.org/10.5578/amrj.57425.

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Uemura, Satoshi, Osamu Sugiyama, Ryosuke Kojima, and Kazuhiro Nakadai. "Outdoor Acoustic Event Identification using Sound Source Separation and Deep Learning with a Quadrotor-Embedded Microphone Array." Abstracts of the international conference on advanced mechatronics : toward evolutionary fusion of IT and mechatronics : ICAM 2015.6 (2015): 329–30. http://dx.doi.org/10.1299/jsmeicam.2015.6.329.

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Chibysheva, Olga Anatolyevna, and Natalia Vladimirovna Osadchuk. "Linguocultural Features of Concept SOUND: Objectification of Microconcept “Sounds Caused by Human Activity” by the Material of English Phraseological Units." Filologičeskie nauki. Voprosy teorii i praktiki, no. 12 (December 2021): 3898–903. http://dx.doi.org/10.30853/phil20210643.

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Legler, Gretchen. "Sounds." Women's Review of Books 17, no. 2 (November 1999): 20. http://dx.doi.org/10.2307/4023348.

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Dissertations / Theses on the topic "SounBe"

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Gunawan, David Oon Tao Electrical Engineering &amp Telecommunications Faculty of Engineering UNSW. "Musical instrument sound source separation." Awarded By:University of New South Wales. Electrical Engineering & Telecommunications, 2009. http://handle.unsw.edu.au/1959.4/41751.

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The structured arrangement of sounds in musical pieces, results in the unique creation of complex acoustic mixtures. The analysis of these mixtures, with the objective of estimating the individual sounds which constitute them, is known as musical instrument sound source separation, and has applications in audio coding, audio restoration, music production, music information retrieval and music education. This thesis principally addresses the issues related to the separation of harmonic musical instrument sound sources in single-channel mixtures. The contributions presented in this work include novel separation methods which exploit the characteristic structure and inherent correlations of pitched sound sources; as well as an exploration of the musical timbre space, for the development of an objective distortion metric to evaluate the perceptual quality of separated sources. The separation methods presented in this work address the concordant nature of musical mixtures using a model-based paradigm. Model parameters are estimated for each source, beginning with a novel, computationally efficient algorithm for the refinement of frequency estimates of the detected harmonics. Harmonic tracks are formed, and overlapping components are resolved by exploiting spectro-temporal intra-instrument dependencies, integrating the spectral and temporal approaches which are currently employed in a mutually exclusive manner in existing systems. Subsequent to the harmonic magnitude extraction using this method, a unique, closed-loop approach to source synthesis is presented, separating sources by iteratively minimizing the aggregate error of the sources, constraining the minimization to a set of estimated parameters. The proposed methods are evaluated independently, and then are placed within the context of a source separation system, which is evaluated using objective and subjective measures. The evaluation of music source separation systems is presently limited by the simplicity of objective measures, and the extensive effort required to conduct subjective evaluations. To contribute to the development of perceptually relevant evaluations, three psychoacoustic experiments are also presented, exploring the perceptual sensitivity of timbre for the development of an objective distortion metric for timbre. The experiments investigate spectral envelope sensitivity, spectral envelope morphing and noise sensitivity.
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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 signal localization in three dimensional space is then presented. This method, which has similarities to the structure of the eye, consists of a novel hemispherical microphone array with microphones on the shell and one microphone in the center of the sphere. The hemispherical array provides such benefits as 3D coverage, simple signal processing and low computational complexity. The signal processing scheme utilizes parallel computation of a special and novel closeness function for each microphone direction on the shell. The closeness functions have output values that are linearly proportional to the spatial angular difference between the sound source direction and each of the shell microphone directions. Finally by choosing directions corresponding to the highest closeness function values and implementing linear weighted spatial averaging in those directions we estimate the sound source direction. The experimental tests validate the method with less than 3.10 of error in a small office room. Contrary to traditional algorithmic sound source localization techniques, the proposed method is based on parallel mathematical calculations in the time domain. Consequently, it can be easily implemented on a custom designed integrated circuit.
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Pompei, F. Joseph (Frank Joseph) 1973. "Sound from ultrasound : the parametric array as an audible sound source." Thesis, Massachusetts Institute of Technology, 2002. http://hdl.handle.net/1721.1/7987.

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Thesis (Ph. D.)--Massachusetts Institute of Technology, School of Architecture and Planning, Program in Media Arts and Sciences, 2002.
Vita.
Includes bibliographical references (leaves 91-94).
A parametric array exploits the nonlinearity of the propagation medium to emit or detect acoustic waves in a spatially versatile manner, permitting concise, narrow directivity patterns otherwise possible only with physically very large transducer geometries. This thesis explores the use of the parametric array as an audible sound source, permitting audible sound to be generated with very high directivity compared to traditional loudspeakers of comparable size. The thesis begins with a review of basic underlying mathematics and relevant approximate solutions of nonlinear acoustic systems. Then, these solutions are used to construct suitable methods of ultrasonic synthesis for low-distortion audio reproduction. Geometrical modelling methods for predicting the acoustic distribution are presented and evaluated, and practical applications are explored experimentally. Issues of risk associated with ultrasonic exposure are presented, and the feasibility of a phased-array system for beam control is explored.
F. Joseph Pompei.
Ph.D.
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Olsson, Erik. "Sound source localization from laser vibrometry recordings." Doctoral thesis, Luleå : Division of experimental mechanics, Luleå University of Technology, 2007. http://epubl.ltu.se/1402-1544/2007/23/.

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Benichoux, Victor. "Timing cues for azimuthal sound source localization." Phd thesis, Université René Descartes - Paris V, 2013. http://tel.archives-ouvertes.fr/tel-00931645.

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Azimuth sound localization in many animals relies on the processing of differences in time-of-arrival of the low-frequency sounds at both ears: the interaural time differences (ITD). It was observed in some species that this cue depends on the spectrum of the signal emitted by the source. Yet, this variation is often discarded, as humans and animals are assumed to be insensitive to it. The purpose of this thesis is to assess this dependency using acoustical techniques, and explore the consequences of this additional complexity on the neurophysiology and psychophysics of sound localization. In the vicinity of rigid spheres, a sound field is diffracted, leading to frequency-dependent wave propagation regimes. Therefore, when the head is modeled as a rigid sphere, the ITD for a given position is a frequency-dependent quantity. I show that this is indeed reflected on human ITDs by studying acoustical recordings for a large number of human and animal subjects. Furthermore, I explain the effect of this variation at two scales. Locally in frequency the ITD introduces different envelope and fine structure delays in the signals reaching the ears. Second the ITD for low-frequency sounds is generally bigger than for high frequency sounds coming from the same position. In a second part, I introduce and discuss the current views on the binaural ITD-sensitive system in mammals. I expose that the heterogenous responses of such cells are well predicted when it is assumed that they are tuned to frequency-dependent ITDs. Furthermore, I discuss how those cells can be made to be tuned to a particular position in space irregardless of the frequency content of the stimulus. Overall, I argue that current data in mammals is consistent with the hypothesis that cells are tuned to a single position in space. Finally, I explore the impact of the frequency-dependence of ITD on human behavior, using psychoacoustical techniques. Subjects are asked to match the lateral position of sounds presented with different frequency content. Those results suggest that humans perceive sounds with different frequency contents at the same position provided that they have different ITDs, as predicted from acoustical data. The extent to which this occurs is well predicted by a spherical model of the head. Combining approaches from different fields, I show that the binaural system is remarkably adapted to the cues available in its environment. This processing strategy used by animals can be of great inspiration to the design of robotic systems.
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Share, C. P. "Real-time simulation of sound source occlusion." Thesis, Queen's University Belfast, 2011. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.546425.

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Lam, Alice. "3D sound-source localization using triangulation-based methods." Thesis, University of British Columbia, 2017. http://hdl.handle.net/2429/63551.

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The localization of sound sources in a reverberant environment, such as a classroom or industrial workspace, is an essential first step toward noise control in these spaces. Many sound source localization techniques have been developed for use with microphone arrays. A common characteristic of these techniques is that they are able to provide the direction from which the sound is coming, but not the range (i.e. the distance between the source and receiver).This thesis presents two triangulation-based methods for localizing sound sources in 3D space, including range, using a small hemispherical microphone array. Practical issues with the hemispherical array, such as source resolution and operating frequency limitations, are discussed. The first method - direct triangulation - involves taking multiple sound field measurements at different locations in the room, and then using the combined output of all receivers to triangulate the source. Direct triangulation is conceptually simple and requires no a priori knowledge of the surrounding environment, but proves cumbersome as multiple array measurements are required - this also limits its application to steady-state noise sources. The second method - image source triangulation - requires only one measurement, instead taking into account the early specular reflections from the walls of the room to create "image receivers" from which the source location can be triangulated. Image source triangulation has the advantage of only requiring one measurement and may be more suited to small spaces such as meeting rooms. However, it relies on having accurate pre-knowledge of the room geometry in relation to the microphones. Both triangulation methods are evaluated using simulations and physical in-room measurements, and are shown to be able to localize simple monopole sources in reverberant rooms.
Applied Science, Faculty of
Mechanical Engineering, Department of
Graduate
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Cavalieri, André Valdetaro Gomes. "Wavepackets as sound-source mechanisms in subsonic jets." Thesis, Poitiers, 2012. http://www.theses.fr/2012POIT2253/document.

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On considère les paquets d'ondes hydrodynamiques comme mécanismes de génération de bruit des jets subsoniques. Cette approche résulte tout d'abord de l'analyse de données numériques - DNS d'une couche de mélange (Wei et Freund 2006) et LES d'un jet à Mach 0,9 (Daviller 2010) - permettant de déterminer les propriétés des sources en termes de compacité, d'intermittence et de structure azimutale. L'identification d'un rayonnement intermittent associé aux modifications des structures cohérentes des écoulements permet de proposer un modèle de paquet d'onde pour représenter ce phénomène dans l'analogie de Lighthill, dont l'enveloppe présente des variations temporelles d'amplitude et d'étendue spatiale. Celles-ci sont tirées de données de vitesse de simulations numériques de jets subsoniques, et un accord de l'ordre de 1,5dB entre le champ acoustique simulé et le modèle confirme sa pertinence. L'exploration du concept proposé est ensuite poursuivie expérimentalement, avec des mesures de pression acoustique et de vitesse de jets turbulents subsoniques, permettant la décomposition des champs en modes de Fourier azimutaux. On observe l'accord des directivités des modes 0, 1 et 2 du champ acoustique avec le rayonnementd'un paquet d'onde. Les modes 0 et 1 du champ de vitesse correspondent également à des paquets d'onde, modélisés comme des ondes d'instabilité linéaires à partir des équations de stabilité parabolisées. Finalement, des corrélations de l'ordre de 10% entre les modes axisymétriques de vitesse dans le jet et de pression acoustique rayonnée montrent un lien clair entre les paquets d'onde et l'émission acoustique du jet
Hydrodynamic wavepackets are studied as a sound-source mechanism in subsonic jets. We first analyse numerical simulations to discern properties of acoustic sources such as compactness, intermittency and azimuthal structure. The simulations include a DNS of a two-dimensional mixing layer (Wei and Freund 2006) and an LES of a Mach 0.9 jet (Daviller 2010). In both cases we identify intermittent radiation, which is associated with changes in coherent structures in the flows. A wave-packet model that includes temporal changes in amplitude and axial extension is proposed to represent the identified phenomena using Lighthill's analogy. These parameters are obtained from velocity data of two subsonic jet simulations, and an agreement to within 1.5dB between the model and the acoustic field of the simulations confirms its pertinence. The proposed mechanism is then investigatedexperimentally, with measurements of acoustic pressure and velocity of turbulent subsonic jets, allowing the decomposition of the fields into azimuthal Fourier modes. We find close agreement of the directivities of modes 0, 1 and 2 of the acoustic field with wave-packet radiation. Modes 0 and 1 of the velocity field correspond also to wavepackets, modelled as linear instability waves using parabolised stability equations. Finally, correlations of order of 10% between axisymmetric modes of velocity and far-field pressure show the relationship between wavepackets and sound radiated by the jet
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Kjellson, Angelica. "Sound Source Localization and Beamforming for Teleconferencing Solutions." Thesis, Umeå universitet, Institutionen för matematik och matematisk statistik, 2014. http://urn.kb.se/resolve?urn=urn:nbn:se:umu:diva-89707.

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In teleconferencing the audio quality is key to conducting successful meetings. The conference room setting imposes various challenges on the speech signal processing, such as noise and interfering signals, reverberation, or participants positioned far from the telephone unit. This work aims at improving the received speech signal of a conference telephone by implementing sound source localization and beamforming. The implemented microphone array signal processing techniques are compared to the performance of an existing multi-microphone solution and evaluated under various conditions using a planar uniform circular array. Recordings of test-sequences for the evaluation were performed using a custom-built array mockup. The implemented algorithms did not show good enough performance to motivate the increased computational complexity compared to the existing solution. Moreover, an increase in number of microphones used was concluded to have little or no effect on the performance of the methods. The type of microphone used was, however, concluded to have impact on the performance and a subjective listening evaluation indicated a preference for omnidirectional microphones which is recommended to investigate further.
God ljudkvalitet är en grundsten för lyckade telefonmöten. Miljön i ett konferens-rum medför ett flertal olika utmaningar för behandlingen av mikrofonsignalerna: det kan t.ex. vara brus och störningar, eller att den som talar befinner sig långt från telefonen. Målet med detta arbete är att förbättra den talsignal som tas upp av en konferenstelefon genom att implementera lösningar för lokalisering av talaren och riktad ljudupptagning med hjälp av ett flertal mikrofoner. De implementerade metoderna jämförs med en befintlig lösning och utvärderas under olika brusscenarion för en likformig cirkulär mikrofonkonstellation. För utvärderingen användes testsignaler som spelades in med en specialbyggd enhet. De implementerade algoritmerna kunde inte uppvisa en tillräcklig förbättring i jämförelse med den befintliga lösningen för att motivera den ökade beräkningskomplexitet de skulle medföra. Dessutom konstaterades att en fördubbling av antalet mikrofoner gav liten eller ingen förbättring på metoderna. Vilken typ av mikrofon som användes konstaterades däremot påverka resultatet och en subjektiv utvärdering indikerade en preferens för de rundupptagande mikrofonerna, en skillnad som föreslås undersökas vidare.
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Martin, Keith Dana. "Sound-source recognition : a theory and computational model." Thesis, Massachusetts Institute of Technology, 1999. http://hdl.handle.net/1721.1/9468.

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Thesis (Ph.D.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 1999.
Includes bibliographical references (p. 159-172).
The ability of a normal human listener to recognize objects in the environment from only the sounds they produce is extraordinarily robust with regard to characteristics of the acoustic environment and of other competing sound sources. In contrast, computer systems designed to recognize sound sources function precariously, breaking down whenever the target sound is degraded by reverberation, noise, or competing sounds. Robust listening requires extensive contextual knowledge, but the potential contribution of sound-source recognition to the process of auditory scene analysis has largely been neglected by researchers building computational models of the scene analysis process. This thesis proposes a theory of sound-source recognition, casting recognition as a process of gathering information to enable the listener to make inferences about objects in the environment or to predict their behavior. In order to explore the process, attention is restricted to isolated sounds produced by a small class of sound sources, the non-percussive orchestral musical instruments. Previous research on the perception and production of orchestral instrument sounds is reviewed from a vantage point based on the excitation and resonance structure of the sound-production process, revealing a set of perceptually salient acoustic features. A computer model of the recognition process is developed that is capable of "listening" to a recording of a musical instrument and classifying the instrument as one of 25 possibilities. The model is based on current models of signal processing in the human auditory system. It explicitly extracts salient acoustic features and uses a novel improvisational taxonomic architecture (based on simple statistical pattern-recognition techniques) to classify the sound source. The performance of the model is compared directly to that of skilled human listeners, using both isolated musical tones and excerpts from compact disc recordings as test stimuli. The computer model's performance is robust with regard to the variations of reverberation and ambient noise (although not with regard to competing sound sources) in commercial compact disc recordings, and the system performs better than three out of fourteen skilled human listeners on a forced-choice classification task. This work has implications for research in musical timbre, automatic media annotation, human talker identification, and computational auditory scene analysis.
by Keith Dana Martin.
Ph.D.
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Books on the topic "SounBe"

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Soledad. Le livre des bruits. Paris: L'école des loisirs, 2004.

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Werner, Hans Ulrich. SoundScapeDesign: Klangwelten, Hörzeichen. Basel: Akroama, 1997.

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Werner, Hans U. Soundscapes =: Akustische Landschaften : eine klangökologische Spurensuche. Basel, Switzerland: Geographisches Institut der Universität, 1992.

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Dubov, Christine Salac. Ding dong! and other sounds. New York, NY: Tambourine Books, 1991.

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Mariétan, Pierre. L'écoute du monde: Actes : Congrès mondial d'écologie sonore #2 = World acoustic ecology congress #2 = Congreso mundial de ecologia sonora #2 : Rencontres architecture musique écologie, Arc-et-Senans (F), Saillon (CH), 17-25 août 2012. Nîmes: Lucie éditions, 2015.

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Darling, David J. Sounds interesting: The science of acoustics. New York: Dillon Press, 1991.

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Becerra, Gabriela Sierra. Ruidos. Guadalajara, Méx: Conexión Gráfica, 1997.

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Bennett, David. Sounds. Toronto: Bantam Books, 1989.

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J, Jennings Terry. Making sounds. New York: Gloucester Press, 1990.

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David, Bennett. Sounds. London: Marvel, 1989.

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Book chapters on the topic "SounBe"

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Evangelista, G., S. Marchand, M. D. Plumbley, and E. Vincent. "Sound Source Separation." In DAFX: Digital Audio Effects, 551–88. Chichester, UK: John Wiley & Sons, Ltd, 2011. http://dx.doi.org/10.1002/9781119991298.ch14.

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Mynett, Mark. "Sound at Source." In Metal Music Manual, 35–61. New York ; London : Routledge, 2016.: Routledge, 2017. http://dx.doi.org/10.4324/9781315750071-5.

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de Jong, Christ A. F., Michael A. Ainslie, Floor Heinis, and Jeroen Janmaat. "Offshore Dredger Sounds: Source Levels, Sound Maps, and Risk Assessment." In The Effects of Noise on Aquatic Life II, 189–96. New York, NY: Springer New York, 2016. http://dx.doi.org/10.1007/978-1-4939-2981-8_22.

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Munguía, Rodrigo, and Antoni Grau. "Single Sound Source SLAM." In Lecture Notes in Computer Science, 70–77. Berlin, Heidelberg: Springer Berlin Heidelberg, 2008. http://dx.doi.org/10.1007/978-3-540-85920-8_9.

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Paul, Peter. "Speech Sounds and Sound Systems." In Linguistics for Language Learning, 110–31. London: Macmillan Education UK, 1993. http://dx.doi.org/10.1007/978-1-349-15182-0_9.

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Wu, Kai, and Andy W. H. Khong. "Sound Source Localization and Tracking." In Human–Computer Interaction Series, 55–78. Cham: Springer International Publishing, 2015. http://dx.doi.org/10.1007/978-3-319-19947-4_3.

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Zhou, Junfeng, Feng Wang, Di Guo, Huaping Liu, and Fuchun Sun. "Video-Guided Sound Source Separation." In Intelligent Robotics and Applications, 415–26. Cham: Springer International Publishing, 2019. http://dx.doi.org/10.1007/978-3-030-27526-6_36.

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Oya, Takashi, Shohei Iwase, Ryota Natsume, Takahiro Itazuri, Shugo Yamaguchi, and Shigeo Morishima. "Do We Need Sound for Sound Source Localization?" In Computer Vision – ACCV 2020, 119–36. Cham: Springer International Publishing, 2021. http://dx.doi.org/10.1007/978-3-030-69544-6_8.

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Tohyama, Mikio. "Signal Dynamics and Sound Source Distance." In Signals and Communication Technology, 297–317. Singapore: Springer Singapore, 2017. http://dx.doi.org/10.1007/978-981-10-5889-9_12.

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Agus, Trevor R., Clara Suied, and Daniel Pressnitzer. "Timbre Recognition and Sound Source Identification." In Timbre: Acoustics, Perception, and Cognition, 59–85. Cham: Springer International Publishing, 2019. http://dx.doi.org/10.1007/978-3-030-14832-4_3.

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Conference papers on the topic "SounBe"

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Sugio, Yuuichi, Ryota Kanetake, Akimitsu Tanaka, and Katsutoshi Ooe. "Work of PZT ceramics sounder for sound source artificial larynx." In 2007 International Symposium on Micro-NanoMechatronics and Human Science. IEEE, 2007. http://dx.doi.org/10.1109/mhs.2007.4420859.

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Sugio, Yuuichi, Ryota Kanetake, Akimitsu Tanaka, and Katsutoshi Ooe. "Work of PZT ceramics sounder for sound source artificial larynx." In The 14th International Symposium on: Smart Structures and Materials & Nondestructive Evaluation and Health Monitoring, edited by Yuji Matsuzaki, Mehdi Ahmadian, and Donald J. Leo. SPIE, 2007. http://dx.doi.org/10.1117/12.715742.

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Ooe, Katsutoshi, Ryota Kanetake, and Akimitsu Tanaka. "Acoustic characteristics improvement of PZT ceramic sounder for sound source of artificial larynx." In Microelectronics, MEMS, and Nanotechnology, edited by Dan V. Nicolau. SPIE, 2005. http://dx.doi.org/10.1117/12.638739.

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Tuma, Jiri, Patrik Janecka, Milan Vala, and Lukas Richter. "Sound Source Localization." In 2012 13th International Carpathian Control Conference (ICCC). IEEE, 2012. http://dx.doi.org/10.1109/carpathiancc.2012.6228744.

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Zhu, Na, and Sean Wu. "Track and Trace Multiple Incoherent Sound Sources in 3D Space in Real Time." In ASME 2009 International Mechanical Engineering Congress and Exposition. ASMEDC, 2009. http://dx.doi.org/10.1115/imece2009-13181.

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This paper presents a methodology for tracking and tracing multiple incoherent sound sources in 3D space in real time. A salient feature of this methodology is its capability of handling all types of sound signals, including broadband, narrowband, continuous, impulsive, and tonal (sinusoidal) sounds over the audible frequency range (20 to 20,000 Hz). Locations of sound sources are indicated in terms of the Cartesian coordinates in real time. The target sources are viewed through an automatic tracking camera covering 350 degree solid angle. The hardware includes four microphones, a thermometer, a webcam, a five-channel signal conditioner and a laptop. Thus, the system can be made light, portable, easy to setup and use and inexpensive. The underlying algorithm is a hybrid approach consisting of modeling of sound radiation from a point source in a free field, triangulation and signal processing techniques. To acquire better understanding of the performance of the device, numerical simulations are conducted to study the impacts of signal noise ratio, microphone spacing, source distance and frequency on the spatial resolution and accuracy of the results. Experiments are carried out to validate results over a wide variety of real-world sound signals such as helicopter noise, human conversations, truck pass-by noise, gun shots, impact sounds, clapping, coughing, etc. Satisfactory results are obtained in most cases, even when a source is behind the measurement microphones.
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Balan, Oana, Alin Moldoveanu, Florica Moldoveanu, Ionut Negoi, and Alex Butean. "COMPARATIVE RESEARCH ON SOUND LOCALIZATION ACCURACY IN THE FREE-FIELD AND VIRTUAL AUDITORY DISPLAYS." In eLSE 2015. Carol I National Defence University Publishing House, 2015. http://dx.doi.org/10.12753/2066-026x-15-079.

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The following paper aims to present a comparative study on the audio localization accuracy (directional judgment, absolute spatial perception and the rate of front-back confusions- situation in which the listener perceives the sound coming from the front as coming from the rear and vice-versa) in both free-field and virtual sound source conditions. Sound localization experiments in the free-field rely on the use of loudspeakers for delivering the auditory information to the listener. On the other hand, virtual auditory displays are based on 3D sounds (resulting from the filtering of a particular sound with the Head Related Transfer Function corresponding to the direction of the sound source in space) that are rendered to the listener through a pair of stereophonic headphones. 3D sounds are used in a wide range of applications, as they can simulate the perception of an external sound source in real-world hearing conditions and generally increase situational awareness. Nonetheless, they can introduce several localization errors (caused primarily by the use of non-individualized Head Related Transfer Functions), such as poor performance in the median plane (for vertical localization) and an increase in the rate of front-back confusions, especially for the directions of 0 degrees (to the front region) and 180 degrees (to the rear). As a result, we intend to include in our research a comprehensive psychophysical evaluation, interpretation and analysis of the accuracy of free-field and headphone-presented stimuli in order to bring to light the audio localization particularities that differentiate audio discrimination performance under the two presented conditions. .
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Heath*, Brian. "Sound Source Verification (SSV)." In SEG Technical Program Expanded Abstracts 2015. Society of Exploration Geophysicists, 2015. http://dx.doi.org/10.1190/segam2015-5907593.1.

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Yoon Seob Lim, Jong Suk Choi, and Mun-Sang Kim. "Probabilistic sound source localization." In 2007 International Conference on Control, Automation and Systems. IEEE, 2007. http://dx.doi.org/10.1109/iccas.2007.4406662.

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Matsumoto, Hiroki, Kohshi Nishida, and Ken-ichi Saitoh. "Characteristics of Aerodynamic Sound Sources Generated by Coiled Wires in a Uniform Air Flow." In ASME 2002 International Mechanical Engineering Congress and Exposition. ASMEDC, 2002. http://dx.doi.org/10.1115/imece2002-33408.

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This study deals experimentally with aerodynamic sounds generated by coiled wires in a uniform air flow. The coiled wire is the model of the hair dryer’s heater. In the experiment, the effects of the outer diameter D, wire diameter d and spacing between coils s of the coiled wire on the aerodynamic sound have been clarified. The results of frequency analyses of the aerodynamic sounds show that an aeolian sound is generated by the coiled wire, when s/d is larger than 1. And the peak frequencies of aeolian sounds generated by the coiled wires are higher than the ones generated by a cylinder having the same diameter d. To clarify the characteristics of the aerodynamic sound sources, the directivity of the aerodynamic sound generated by the coiled wire has been examined. And the coherent function between the velocity fluctuation around the coiled wire and the aerodynamic sound has been measured. Moreover the band overall value of coherent output power between the sound and the velocity fluctuation has been calculated. This method has clarified the sound source region of the aeolian sound generated by the coiled wire. These results show that the aeolian sound is generated by the arc part of the coiled wire which is located in the upper side of the air flow.
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Simic, Ines, and Rutger van Aalst. "Underwater Sound Filtering." In SNAME 5th World Maritime Technology Conference. SNAME, 2015. http://dx.doi.org/10.5957/wmtc-2015-247.

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The cocktail party algorithm is one of the most widely used algorithms for source separation of sound. The algorithm aims to find an automated solution for a problem that everyone experiences regularly, namely how to make oneself heard in a noisy environment. The cocktail party algorithm picks up the sound from different microphones, and then applies smart filters once the system has determined which sounds originate from the same source. This problem also becomes topical when developing sensors based on passive sonar, for instance for autonomous aquatic drones who have to develop awareness of ships and other possible obstacles on a busy shipping lane. It is possible to deploy multiple hydrophones to localize sound sources under water, but the system will be hindered considerably by the sound that the drone itself makes, such as the sound produced by the propellers. This paper describes a possible solution to the underwater sound filtering problem, using Blind Source Separation. The problem regards splitting sound from a boat engine and the water waves to prove the possibility to extract one sound fragment from the other on the open sea. The illustrations shown further in the report are tests performed in MATLAB to prove the theory.
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Reports on the topic "SounBe"

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Valdes, James R., and Heather Furey. WHOI 260Hz Sound Source - Tuning and Assembly. Woods Hole Oceanographic Institution, April 2021. http://dx.doi.org/10.1575/1912/27173.

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Sound sources are designed to provide subsea tracking and re‐location of RAFOS floats and other Lagrangian drifters listening at 260Hz. More recently sweeps have been added to support FishChip tracking at 262Hz. These sources must be tuned to the water properties where they are to be deployed as they have a fairly narrow bandwidth. The high‐Q resonator’s bandwidth is about 4Hz. This report documents the tuning, and provides an overview of the sound source assembly.
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TEXAS UNIV AT AUSTIN APPLIED RESEARCH LABS. Plasma Sound Source Basic Research Annual Summary Report. Fort Belvoir, VA: Defense Technical Information Center, September 1994. http://dx.doi.org/10.21236/ada285394.

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Dolotii, Marharyta H., and Pavlo V. Merzlykin. Using the random number generator with a hardware entropy source for symmetric cryptography problems. [б. в.], December 2018. http://dx.doi.org/10.31812/123456789/2883.

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The aim of the research is to test the possibility of using the developed random number generator [1], which utilizes the sound card noise as an entropy source, in the symmetric cryptography algorithms.
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Coulter, R. L. A study of the effects of an additional sound source on RASS performance. Office of Scientific and Technical Information (OSTI), December 1998. http://dx.doi.org/10.2172/564110.

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Job, Jacob. Mesa Verde National Park: Acoustic monitoring report. National Park Service, July 2021. http://dx.doi.org/10.36967/nrr-2286703.

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In 2015, the Natural Sounds and Night Skies Division (NSNSD) received a request to collect baseline acoustical data at Mesa Verde National Park (MEVE). Between July and August 2015, as well as February and March 2016, three acoustical monitoring systems were deployed throughout the park, however one site (MEVE002) stopped recording after a couple days during the summer due to wildlife interference. The goal of the study was to establish a baseline soundscape inventory of backcountry and frontcountry sites within the park. This inventory will be used to establish indicators and thresholds of soundscape quality that will support the park and NSNSD in developing a comprehensive approach to protecting the acoustic environment through soundscape management planning. Additionally, results of this study will help the park identify major sources of noise within the park, as well as provide a baseline understanding of the acoustical environment as a whole for use in potential future comparative studies. In this deployment, sound pressure level (SPL) was measured continuously every second by a calibrated sound level meter. Other equipment included an anemometer to collect wind speed and a digital audio recorder collecting continuous recordings to document sound sources. In this document, “sound pressure level” refers to broadband (12.5 Hz–20 kHz), A-weighted, 1-second time averaged sound level (LAeq, 1s), and hereafter referred to as “sound level.” Sound levels are measured on a logarithmic scale relative to the reference sound pressure for atmospheric sources, 20 μPa. The logarithmic scale is a useful way to express the wide range of sound pressures perceived by the human ear. Sound levels are reported in decibels (dB). A-weighting is applied to sound levels in order to account for the response of the human ear (Harris, 1998). To approximate human hearing sensitivity, A-weighting discounts sounds below 1 kHz and above 6 kHz. Trained technicians calculated time audible metrics after monitoring was complete. See Methods section for protocol details, equipment specifications, and metric calculations. Median existing (LA50) and natural ambient (LAnat) metrics are also reported for daytime (7:00–19:00) and nighttime (19:00–7:00). Prominent noise sources at the two backcountry sites (MEVE001 and MEVE002) included vehicles and aircraft, while building and vehicle predominated at the frontcountry site (MEVE003). Table 1 displays time audible values for each of these noise sources during the monitoring period, as well as ambient sound levels. In determining the current conditions of an acoustical environment, it is informative to examine how often sound levels exceed certain values. Table 2 reports the percent of time that measured levels at the three monitoring locations were above four key values.
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Smith, T., and K. H. Lee. Controlled-source magnetotellurics: source effects. Office of Scientific and Technical Information (OSTI), April 1999. http://dx.doi.org/10.2172/760306.

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Spiesberger, John L. Acquisition of Acoustic Source to Augment Navy Sonars for Mapping Sound Speed and Temperature with Tomography. Fort Belvoir, VA: Defense Technical Information Center, September 1999. http://dx.doi.org/10.21236/ada630196.

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Lockhead, Gregory R. Categorizing Sounds. Fort Belvoir, VA: Defense Technical Information Center, December 1989. http://dx.doi.org/10.21236/ada216417.

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Lucas, A. T. The Advanced Neutron Source liquid deuterium cold source. Office of Scientific and Technical Information (OSTI), August 1995. http://dx.doi.org/10.2172/211649.

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Pack, Adam A., J. Potter, L. M. Herman, M. Hoffmann-Kuhnt, and M. H. Deakos. Determining Source Levels Sound Fields and Body Sizes of Singing Humpback Whales (Megaptera novaeangliae) in the Hawaiian Winter Ground. Fort Belvoir, VA: Defense Technical Information Center, October 2003. http://dx.doi.org/10.21236/ada421803.

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