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Gunawan, David Oon Tao Electrical Engineering & 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 jetHydrodynamic 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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Beauvois, Michael W. "A computer model of auditory stream segregation." Thesis, Loughborough University, 1991. https://dspace.lboro.ac.uk/2134/33091.
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A simple computer model is described that takes a novel approach to the problem of accounting for perceptual coherence among successive pure tones of changing frequency by using simple physiological principles that operate at a peripheral, rather than a central level. The model is able to reproduce a number of streaming phenomena found in the literature using the same parameter values. These are: (1) the build-up of streaming over time; (2) the temporal coherence and fission boundaries of human listeners; (3) the ambiguous region; and (4) the trill threshold. In addition, the principle of excitation integration used in the model can be used to account for auditory grouping on the basis of the Gestalt perceptual principles of closure, proximity, continuity, and good continuation, as well as the pulsation threshold. The examples of Gestalt auditory grouping accounted for by the excitation integration principle indicate that the predictive power of the model would be considerably enhanced by the addition of a cross-channel grouping mechanism that worked on the basis of common on sets and offsets, as more complex stimuli could then be processed by the model.
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Mak, Lin Chi Mechanical & Manufacturing Engineering Faculty of Engineering UNSW. "Non-Line-of-Sight localisation of a sound source." Awarded by:University of New South Wales. Mechanical & Manufacturing Engineering, 2009. http://handle.unsw.edu.au/1959.4/44702.
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This thesis proposes two acoustic localisation techniques that are accurate in Non-Line-of-Sight (NLoS) conditions and system implementation of the proposed techniques. Such conditions can cause positive bias errors, namely NLoS errors, in the measured Time-of-Arrivals (ToAs) of first-arrival signals received by microphones and thus reduce the positioning accuracy. The primary issue of the thesis is to precisely estimate and correct the NLoS errors by modelling the received first-arrival signals. The first proposed technique uses multiple on-ground microphones to locate a sound source. The proposed technique approximately estimates and corrects the NLoS errors based on an initial guess of the sound source position and a map. The localisation is then achieved by iteratively correcting the ToAs and updating the sound source location. The strength of the proposed technique is that its accuracy is not notably affected by small or known obstacles. The proposed technique is implemented into two localisation systems of controlled and uncontrolled sound sources. The performance of the proposed technique is investigated by its comparison with three other time-based localisation techniques in series of experiments and simulations, showing at least 10% improvement by the proposed technique under various background noise levels. The second proposed technique localises a sound source using a single on-ground microphone subject to an assumption of a single diffraction in the first-arrival signal. To predict the angular and radial coordinates of the sound source relative to the diffraction point, a new magnitude delay frequency profile is proposed. The profile can be estimated by applying the uniform geometrical theory of diffraction and be extracted from measured signals using a derived formulation. Similar to the first technique, the second proposed technique estimates the measured delay of the first-arrival signal for computing the radial coordinate. The angular coordinate is then obtained by matching the estimated and measured profiles at the measured delay. A key achievement of the second proposed technique is enabling NLoS localisation using only one microphone without any time-consuming pre-measurement. This technique is implemented into a localisation system of a controlled sound source and validated experimentally with three different sound sources and under two background noise levels.
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Johansson, Anders M. "Acoustic sound source localisation and tracking : in indoor environments /." Karlskrona : Department of Signal Processing, Blekinge Institute of Technology, 2008. http://www.bth.se/fou/Forskinfo.nsf/allfirst2/beeb3e73884ff408c125744f00370a08?OpenDocument.
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Johansson, Anders. "Acoustic Sound Source Localisation and Tracking : in Indoor Environments." Doctoral thesis, Blekinge Tekniska Högskola [bth.se], School of Engineering - Dept. of Signal Processing, 2008. http://urn.kb.se/resolve?urn=urn:nbn:se:bth-00401.
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With advances in micro-electronic complexity and fabrication, sophisticated algorithms for source localisation and tracking can now be deployed in cost sensitive appliances for both consumer and commercial markets. As a result, such algorithms are becoming ubiquitous elements of contemporary communication, robotics and surveillance systems. Two of the main requirements of acoustic localisation and tracking algorithms are robustness to acoustic disturbances (to maximise localisation accuracy), and low computational complexity (to minimise power-dissipation and cost of hardware components). The research presented in this thesis covers both advances in robustness and in computational complexity for acoustic source localisation and tracking algorithms. This thesis also presents advances in modelling of sound propagation in indoor environments; a key to the development and evaluation of acoustic localisation and tracking algorithms. As an advance in the field of tracking, this thesis also presents a new method for tracking human speakers in which the problem of the discontinuous nature of human speech is addressed using a new state-space filter based algorithm which incorporates a voice activity detector. The algorithm is shown to achieve superior tracking performance compared to traditional approaches. Furthermore, the algorithm is implemented in a real-time system using a method which yields a low computational complexity. Additionally, a new method is presented for optimising the parameters for the dynamics model used in a state-space filter. The method features an evolution strategy optimisation algorithm to identify the optimum dynamics’ model parameters. Results show that the algorithm is capable of real-time online identification of optimum parameters for different types of dynamics models without access to ground-truth data. Finally, two new localisation algorithms are developed and compared to older well established methods. In this context an analytic analysis of noise and room reverberation is conducted, considering its influence on the performance of localisation algorithms. The algorithms are implemented in a real-time system and are evaluated with respect to robustness and computational complexity. Results show that the new algorithms outperform their older counterparts, both with regards to computational complexity, and robustness to reverberation and background noise. The field of acoustic modelling is advanced in a new method for predicting the energy decay in impulse responses simulated using the image source method. The new method is applied to the problem of designing synthetic rooms with a defined reverberation time, and is compared to several well established methods for reverberation time prediction. This comparison reveals that the new method is the most accurate.
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Liao, Wei-Hsiang. "Modelling and transformation of sound textures and environmental sounds." Thesis, Paris 6, 2015. http://www.theses.fr/2015PA066725/document.
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Le traitement et la synthèse des sons environnementaux sont devenue un sujet important. Une classe des sons, qui est très important pour la constitution d'environnements sonore, est la classe des textures sonores. Les textures sonores sont décrit par des relations stochastiques et qui contient des composantes non-sinusoïdales à caractère fortement bruité. Il a été montré récemment que la reconnaissance de textures sonores est basée sur des mesures statistiques caractérisant les enveloppes dans les bandes critiques. Il y actuellement très peu d'algorithmes qui permettent à imposer des propriétés statistiques de façon explicite lors de la synthèse de sons. L'algorithme qui impose l'ensemble de statistique qui est perceptivement relevant pour les textures sonore est très couteuse en temps de calcul. Nous proposons une nouvelle approche d'analyse-synthèse qui permet une analyse des statistiques relevant et un mécanisme efficace d'imposer ces statistiques dans le domaine temps-fréquence. La représentation temps-fréquence étudié dans cette thèse est la transformée de Fourier à court terme. Les méthodes proposées par contre sont plus générale et peuvent être généralisé à d'autres représentations temps-fréquence reposant sur des banques de filtres si certaines contraintes sont respectées. L'algorithme proposé dans cette thèse ouvre plusieurs perspectives. Il pourrait être utilisé pour générer des textures sonores à partir d'une description statistique créée artificiellement. Il pourrait servir de base pour des transformations avancées comme le morphing, et on pourrait aussi imaginer à utiliser le modèle pour développer un contrôle sémantique de textures sonoresThe processing of environmental sounds has become an important topic in various areas. Environmental sounds are mostly constituted of a kind of sounds called sound textures. Sound textures are usually non-sinusoidal, noisy and stochastic. Several researches have stated that human recognizes sound textures with statistics that characterizing the envelopes of auditory critical bands. Existing synthesis algorithms can impose some statistical properties to a certain extent, but most of them are computational intensive. We propose a new analysis-synthesis framework that contains a statistical description that consists of perceptually important statistics and an efficient mechanism to adapt statistics in the time-frequency domain. The quality of resynthesised sound is at least as good as state-of-the-art but more efficient in terms of computation time. The statistic description is based on the STFT. If certain conditions are met, it can also adapt to other filter bank based time-frequency representations (TFR). The adaptation of statistics is achieved by using the connection between the statistics on TFR and the spectra of time-frequency domain coefficients. It is possible to adapt only a part of cross-correlation functions. This allows the synthesis process to focus on important statistics and ignore the irrelevant parts, which provides extra flexibility. The proposed algorithm has several perspectives. It could possibly be used to generate unseen sound textures from artificially created statistical descriptions. It could also serve as a basis for transformations like stretching or morphing. One could also expect to use the model to explore semantic control of sound textures
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Lee, Chung. "Sound texture synthesis using an enhanced overlap-add approach /." View abstract or full-text, 2008. http://library.ust.hk/cgi/db/thesis.pl?CSED%202008%20LEE.
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Noshino, Eri. "Sound-Intensity-Dependent Compensation for the Small Interaural Time Difference Cue for Sound Source Localization." Kyoto University, 2008. http://hdl.handle.net/2433/124080.
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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 elliptical edge pixel regions typical of source patterns. Monte Carlo simulations of an eight microphone perimeter array with single and multiple sound sources are used to simulate the test environment and area under receiver operating characteristic (ROCA) curve is used to analyze the algorithm performance. Performance was compared to a simpler algorithm involving Canny edge detection and image averaging and an algorithms based simply on the magnitude of local maxima in the SRCP image. Analysis shows that Canny edge detection based method performed better in the presence of coherent noise sources.
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Ramamurthy, Anand. "EXPERIMENTAL EVALUATION OF MODIFIED PHASE TRANSFORM FOR SOUND SOURCE DETECTION." UKnowledge, 2007. http://uknowledge.uky.edu/gradschool_theses/478.
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The detection of sound sources with microphone arrays can be enhanced through processing individual microphone signals prior to the delay and sum operation. One method in particular, the Phase Transform (PHAT) has demonstrated improvement in sound source location images, especially in reverberant and noisy environments. Recent work proposed a modification to the PHAT transform that allows varying degrees of spectral whitening through a single parameter, andamp;acirc;, which has shown positive improvement in target detection in simulation results. This work focuses on experimental evaluation of the modified SRP-PHAT algorithm. Performance results are computed from actual experimental setup of an 8-element perimeter array with a receiver operating characteristic (ROC) analysis for detecting sound sources. The results verified simulation results of PHAT- andamp;acirc; in improving target detection probabilities. The ROC analysis demonstrated the relationships between various target types (narrowband and broadband), room reverberation levels (high and low) and noise levels (different SNR) with respect to optimal andamp;acirc;. Results from experiment strongly agree with those of simulations on the effect of PHAT in significantly improving detection performance for narrowband and broadband signals especially at low SNR and in the presence of high levels of reverberation.
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Shub, Daniel E. (Daniel Eric) 1974. "The role of the precedence effect in sound source lateralization." Thesis, Massachusetts Institute of Technology, 2001. http://hdl.handle.net/1721.1/86768.
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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 with the location information is utilized to estimate the sources. To further enhance the estimated sources, binary time-frequency masking based post-processing is used but cepstral domain smoothing is required to mitigate musical noise. To tackle the under-determined case and further improve separation performance at higher reverberation times, a two-microphone based method which is inspired by human auditory processing and generates soft time-frequency masks is described. In this approach interaural level difference, interaural phase difference and mixing vectors are probabilistically modeled in the time-frequency domain and the model parameters are learned through the expectation-maximization (EM) algorithm. A direction vector is estimated for each source, using the location information, which is used as the mean parameter of the mixing vector model. Soft time-frequency masks are used to reconstruct the sources. A spatial covariance model is then integrated into the probabilistic model framework that encodes the spatial characteristics of the enclosure and further improves the separation performance in challenging scenarios i.e. when sources are in close proximity and when the level of reverberation is high. Finally, new dereverberation based pre-processing is proposed based on the cascade of three dereverberation stages where each enhances the twomicrophone reverberant mixture. The dereverberation stages are based on amplitude spectral subtraction, where the late reverberation is estimated and suppressed. The combination of such dereverberation based pre-processing and use of soft mask separation yields the best separation performance. All methods are evaluated with real and synthetic mixtures formed for example from speech signals from the TIMIT database and measured room impulse responses.
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Zantalis, Dimitrios. "Guided matching pursuit and its application to sound source separation." Thesis, University of York, 2016. http://etheses.whiterose.ac.uk/13204/.
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In the last couple of decades there has been an increasing interest in the application of source separation technologies to musical signal processing. Given a signal that consists of a mixture of musical sources, source separation aims at extracting and/or isolating the signals that correspond to the original sources. A system capable of high quality source separation could be an invaluable tool for the sound engineer as well as the end user. Applications of source separation include, but are not limited to, remixing, up-mixing, spatial re-configuration, individual source modification such as filtering, pitch detection/correction and time stretching, music transcription, voice recognition and source-specific audio coding to name a few. Of particular interest is the problem of separating sources from a mixture comprising two channels (2.0 format) since this is still the most commonly used format in the music industry and most domestic listening environments. When the number of sources is greater than the number of mixtures (which is usually the case with stereophonic recordings) then the problem of source separation becomes under-determined and traditional source separation techniques, such as “Independent Component Analysis” (ICA) cannot be successfully applied. In such cases a family of techniques known as “Sparse Component Analysis” (SCA) are better suited. In short a mixture signal is decomposed into a new domain were the individual sources are sparsely represented which implies that their corresponding coefficients will have disjoint (or almost) disjoint supports. Taking advantage of this property along with the spatial information within the mixture and other prior information that could be available, it is possible to identify the sources in the new domain and separate them by going back to the time domain. It is a fact that sparse representations lead to higher quality separation. Regardless, the most commonly used front-end for a SCA system is the ubiquitous short-time Fourier transform (STFT) which although is a sparsifying transform it is not the best choice for this job. A better alternative is the matching pursuit (MP) decomposition. MP is an iterative algorithm that decomposes a signal into a set of elementary waveforms called atoms chosen from an over-complete dictionary in such a way so that they represent the inherent signal structures. A crucial part of MP is the creation of the dictionary which directly affects the results of the decomposition and subsequently the quality of source separation. Selecting an appropriate dictionary could prove a difficult task and an adaptive approach would be appropriate. This work proposes a new MP variant termed guided matching pursuit (GMP) which adds a new pre-processing step into the main sequence of the MP algorithm. The purpose of this step is to perform an analysis of the signal and extract important features, termed guide maps, that are used to create dynamic mini-dictionaries comprising atoms which are expected to correlate well with the underlying signal structures thus leading to focused and more efficient searches around particular supports of the signal. This algorithm is accompanied by a modular and highly flexible MATLAB implementation which is suited to the processing of long duration audio signals. Finally the new algorithm is applied to the source separation of two-channel linear instantaneous mixtures and preliminary testing demonstrates that the performance of GMP is on par with the performance of state of the art systems.
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Chapman, David P. "Playing with sounds : a spatial solution for computer sound synthesis." Thesis, University of Bath, 1996. https://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.307047.
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Tinney, Charles E. "Low-dimensional techniques for sound source identification in high speed jets." Related electronic resource: Current Research at SU : database of SU dissertations, recent titles available full text, 2005. http://wwwlib.umi.com/cr/syr/main.
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Rådsten-Ekman, Maria. "MAY NOISY SOUND ENVIRONMENTS BE IMPROVED BY ADDING PLEASANT WATER SOUNDS?" Thesis, Stockholms universitet, Psykologiska institutionen, 2010. http://urn.kb.se/resolve?urn=urn:nbn:se:su:diva-43834.
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Sheaffer, J. "From source to brain : modelling sound propagation and localisation in rooms." Thesis, University of Salford, 2013. http://usir.salford.ac.uk/29210/.
Full textAbstract:
Human localisation of sound in enclosed spaces is a cross-disciplinary research topic, with important applications in auditory science, room acoustics, spatial audio and telecommunications. By combining an accelerated model of $3$D sound propagation in rooms with a perceptual model of spatial processing, this thesis provides an integrated framework for studying sound localisation in enclosed spaces on the horizontal plane, with particular emphasis on room acoustics applications. The room model is based on the finite difference time domain (FDTD) method, which has been extended to include physically-constrained sources and binaural receivers based on laser-scanned listener geometries. The underlying algorithms have been optimised to run on parallel graphics hardware, thus allowing for a high spatial resolution, and accordingly, a significant decrease of numerical dispersion evident in the FDTD method. The perceptual stage of the model features a signal processing chain emulating the physiology of the auditory periphery, binaural cue selection based on interaural coherence, and a final decision maker based on supervised learning. The entire model is shown to be capable of imitating human sound localisation in different listening situations, including free field conditions and at the presence of sound occlusion, diffraction and reflection. Results are validated against subjective data found in the literature, and the model's applications to the fields of room acoustics and spatial audio are demonstrated and discussed.
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Braun, Michael E. "Sound source contributions for the prediction of vehicle pass-by noise." Thesis, Loughborough University, 2014. https://dspace.lboro.ac.uk/2134/16322.
Full textAbstract:
Current European legislation aims to limit vehicle noise emissions since many people are exposed to road traffic noise in urban areas. Vehicle pass-by noise is measured according to the international standard ISO 362 in Europe. More recent investigations of urban traffic have led to the proposal of a revised ISO 362 which includes a constant-speed test in addition to the traditional accelerated test in order to determine the pass-by noise value. In order to meet the legal pass-by noise requirements, vehicle manufacturers and suppliers must analyse and quantify vehicle noise source characteristics during the development phase of the vehicle. In addition, predictive tools need to be available for the estimation of the final pass-by noise value. This thesis aims to contribute to the understanding of vehicle pass-by noise and of the characteristics of the vehicle noise sources contributing to pass-by noise. This is supported through an extensive literature review in which current pass-by noise prediction methods are reviewed as well. Furthermore, three vehicle noise sources are replicated experimentally under laboratory conditions. This involves an orifice noise source, represented by a specially designed loudspeaker on a moving trolley, shell noise, represented by a metal cylinder structure, and tyre cavity and sidewall noise, represented by an annular membrane mounted on a tyre-like structure. The experimentally determined directivity characteristics of the acoustically excited noise sources are utilised in the pass-by noise prediction method. The predictive results are validated against experimental measurements of the three vehicle-like noise sources made within an anechoic chamber.
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Preston, Giles Andrew. "Modelling sound source regions for the prediction of coaxial jet noise." Thesis, University of Southampton, 1995. https://eprints.soton.ac.uk/173779/.
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Cobos, Serrano Máximo. "Application of sound source separation methods to advanced spatial audio systems." Doctoral thesis, Universitat Politècnica de València, 2010. http://hdl.handle.net/10251/8969.
Full textAbstract:
This thesis is related to the field of Sound Source Separation (SSS). It addresses the development
and evaluation of these techniques for their application in the resynthesis of high-realism sound scenes by
means of Wave Field Synthesis (WFS). Because the vast majority of audio recordings are preserved in twochannel
stereo format, special up-converters are required to use advanced spatial audio reproduction formats,
such as WFS. This is due to the fact that WFS needs the original source signals to be available, in order to
accurately synthesize the acoustic field inside an extended listening area. Thus, an object-based mixing is
required.
Source separation problems in digital signal processing are those in which several signals have been mixed
together and the objective is to find out what the original signals were. Therefore, SSS algorithms can be applied
to existing two-channel mixtures to extract the different objects that compose the stereo scene. Unfortunately,
most stereo mixtures are underdetermined, i.e., there are more sound sources than audio channels. This
condition makes the SSS problem especially difficult and stronger assumptions have to be taken, often related to
the sparsity of the sources under some signal transformation.
This thesis is focused on the application of SSS techniques to the spatial sound reproduction field. As a result,
its contributions can be categorized within these two areas. First, two underdetermined SSS methods are
proposed to deal efficiently with the separation of stereo sound mixtures. These techniques are based on a
multi-level thresholding segmentation approach, which enables to perform a fast and unsupervised separation of
sound sources in the time-frequency domain. Although both techniques rely on the same clustering type, the
features considered by each of them are related to different localization cues that enable to perform separation
of either instantaneous or real mixtures.Additionally, two post-processing techniques aimed at
improving the isolation of the separated sources are proposed. The performance achieved by
several SSS methods in the resynthesis of WFS sound scenes is afterwards evaluated by means of
listening tests, paying special attention to the change observed in the perceived spatial attributes.
Although the estimated sources are distorted versions of the original ones, the masking effects
involved in their spatial remixing make artifacts less perceptible, which improves the overall
assessed quality. Finally, some novel developments related to the application of time-frequency
processing to source localization and enhanced sound reproduction are presented.Cobos Serrano, M. (2009). Application of sound source separation methods to advanced spatial audio systems [Tesis doctoral no publicada]. Universitat Politècnica de València. https://doi.org/10.4995/Thesis/10251/8969
Palancia
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Zheng, Haosheng, and Kaichun Zhang. "Noise Analysis of Computer Chassis and Secondary Sound Source Noise Reduction." Thesis, Blekinge Tekniska Högskola, Institutionen för maskinteknik, 2019. http://urn.kb.se/resolve?urn=urn:nbn:se:bth-18547.
Full textAbstract:
This article focuses on computer noise analysis and noise reduction processing. With the popularity of computers, people are increasingly demanding the comfort of using computers. Solving the noise problem of the computer case can make the working environment more comfortable. People working in a noisy environment for a long time can cause anxiety and the quality of work is not high. The main purpose of this paper is to analyse the characteristics of computer noise and to reduce the noise of the chassis through the secondary sound source. Through the comparison of the experimental and simulation results, the noise reduction effect of the secondary sound source on the computer case is obtained. This paper can provide a scientific reference for the manufacture of computer chassis and improvement of noise.
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Ziemann, Astrid, and Kati Balogh. "Gekoppelter Atmosphäre-Boden-Einfluss auf die Schallausbreitung einer höher gelegenen Schallquelle." Universitätsbibliothek Leipzig, 2017. http://nbn-resolving.de/urn:nbn:de:bsz:15-qucosa-221985.
Full textAbstract:
Im Genehmigungsverfahren für den Bau hochliegender Schallquellen (z.B. Windenergieanlagen) muss der Nachweis geführt werden, dass von den Anlagen keine schädlichen Umwelteinwirkungen ausgehen. Es ist es daher notwendig, die Schallausbreitung derartiger Quellen grundsätzlich zu untersuchen. Eine Schwierigkeit stellt dabei die gekoppelte Wirkung von Temperatur-, Windgeschwindigkeits- und Windrichtungsprofil in Zusammenhang mit dem Bodeneinfluss auf die Schallausbreitung dar. Dieser zeitlich und räumlich variable Atmosphäreneinfluss wird insbesondere bei Langzeituntersuchungen der Schallimmission bisher nur unzureichend in den operationellen Modellen beschrieben. Das Ziel der Studie besteht deshalb darin, die gekoppelte Wirkung von Atmosphäre- und Boden-Einfluss auf die Schallausbreitung in einem Bereich bis zu 2 km Entfernung von der Schallquelle mit dem Modell SMART (Sound propagation model of the atmosphere using ray-tracing ) zu untersuchenThe licensing procedure for the construction of high-placed sound sources (e.g. wind power stations) demands to proof that no (significant and) harmful impact on environment is outgoing from these systems. Therefore, it is necessary to analyse the sound propagation of such a kind of sources. In this context one central problem has to be managed: the coupled effect of temperature, wind speed and wind direction profiles combined with the influence of surface on sound propagation. The temporally and spatially variable influence of the atmosphere is only insufficiently described by the operational models, especially in relation to long-time investigations of sound immission. Consequently, the aim of this study was to investigate the coupled effect of atmospheric and surface influence on sound propagation up to distances of 2 km away from the sound source using the model SMART (Sound propagation model of the atmosphere using ray-tracing)
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Verviers, Claire Juliette. "The Influence Of Sound Properties On The Semantic Associations Of Product Sounds." Master's thesis, METU, 2010. http://etd.lib.metu.edu.tr/upload/12612412/index.pdf.
Full textAbstract:
To be able to design product sounds that elicit a predetermined expression a study was
performed to find how sound properties influence the experience of their expression. Two
explorative studies using figurative against abstract visual stimuli were performed to
create insight in how people experience sounds and to create a list of usable semantic
associations. This list was ordered in 25 expression categories each under one descriptive
semantic association. A third study using mind mapping was conducted to examine what
sound properties were considered as influences on a few of these categories and to
optimize the categorization. The sound properties that were considered as most influential
were sharpness and noisiness. The final descriptive semantic associations were placed
on a scale with the axes unpleasant-pleasant and calm-active. From these the following
were considered to be most usable: activated, angry, boring, calm, chaotic, cheerful, eerie,
energetic, pleasant, relaxed, trustworthy and unpleasant. In a fourth study the sounds of
six domestic appliances were chosen and adjusted for sharpness, noisiness and their
combination. They were evaluated for their valued expression on the 12 semantic
associations by 30 participants. The results showed that increased sharpness elicited a
more unpleasant and activated expression and decreased sharpness elicited a more
pleasant and calm expression. This indicates that a general influence of sound properties
can be established to design sounds for expression.
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Bekkos, Audun. "Source Direction Determination with Headphones : An Adaptable Model for Binaural Surround Sound." Thesis, Norges teknisk-naturvitenskapelige universitet, Institutt for elektronikk og telekommunikasjon, 2012. http://urn.kb.se/resolve?urn=urn:nbn:no:ntnu:diva-18578.
Full textAbstract:
An adaptable binaural model for surround sound has been developed in this master’s thesis. The adaptability is based on measurements of the listener’s head. This model is based on what was found to be the best suited material combination of successful models in earlier studies. This includes an ellipsoidal model for interaural time difference, an one-pole, one-zero head shadow filter and the use of Blauert’s directional bands for spectral manipulation. The model can play back six channel surround content using the standardized 5.1 surround sound loudspeaker setup. This standardized loudspeaker placement is used when creating virtual sound sources. Arbitrary sound directions are made in the horizontal plane by creating virtual sound sources using vector base amplitude panning between the standardized loudspeaker positions.To test the performance of this model, a listening test was conducted. The hypothesis tested was that the adaptable model would produce equal or lower localization error, compared to the commercial model. 20 test subjects participated. The test featured three different test types; standardized 5.1 loudspeaker setup, a commercial model for surround sound in headphones, and the adaptable model. Localization accuracy for ten selected directions in the right half plane was tested. The results from the adaptable model were compared to the result of the commercial model. The loudspeaker setup acted as a reference.Mean localization error was found to be thrice as high for the adaptable model, compared to the commercial model. Both models had the same standard deviation. 95% of the confidence intervals for these models did not overlap, i.e. there is a significant difference between the two methods. With this one can safely conclude that the commercial model provided a smaller localization error than the adaptable model. Hence the hypothesis has to be disproved.Both the commercial model and the thesis model performed significantly worse than the loudspeaker setup. One difference between commercial model, and the thesis model, was that that the commercial model had added room reflections and reverberation. This can create the sensation that the sound is coming from outside the head, and make it easier to localize. This contradicts with the knowledge that reverberation diffuses the sound field, making the direct sound that provides the directional information become less prominent.
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Nakamura, Keisuke. "Robust Sound Source Localization Based on Microphone Array Processing for Robot Audition." 京都大学 (Kyoto University), 2013. http://hdl.handle.net/2433/174833.
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Hummersone, Christopher. "A psychoacoustic engineering approach to machine sound source separation in reverberant environments." Thesis, University of Surrey, 2011. http://epubs.surrey.ac.uk/2923/.
Full textAbstract:
Reverberation continues to present a major problem for sound source separation algorithms, due to its corruption of many of the acoustical cues on which these algorithms rely. However, humans demonstrate a remarkable robustness to reverberation and many psychophysical and perceptual mechanisms are well documented. This thesis therefore considers the research question; can the reverberation-performance of existing psychoacoustic engineering approaches to machine source separation be improved. The precedence effect is a perceptual mechanism that aids our ability to localise sounds in reverberant environments. Despite this, relatively little work has been done on incorporating the precedence effect into automated sound source separation. Consequently, a study was conducted that compared several computational precedence models and their impact on the performance of a baseline separation algorithm. The algorithm included a precedence model, which was replaced with the other precedence models during the investigation. The models were tested using a novel metric in a range of reverberant rooms and with a range of other mixture parameters. The metric, termed Ideal Binary Mask Ratio, is shown to be robust to the effects of reverberation and facilitates meaningful and direct comparison between algorithms across different acoustic conditions. Large differences between the performances of the models were observed. The results showed that a separation algorithm incorporating a model based on interaural coherence produces the greatest performance gain over the baseline algorithm. The results from the study also indicated that it may be necessary to adapt the precedence model to the acoustic conditions in which the model is utilised. This effect is analogous to the perceptual Clifton effect, which is a dynamic component of the precedence effect that appears to adapt precedence to a given acoustic environment in order to maximise its effectiveness. However, no work has been carried out on adapting a precedence model to the acoustic conditions under test. Specifically, although the necessity for such a component has been suggested in the literature, neither its necessity nor benefit has been formally validated. Consequently, a further study was conducted in which parameters of each of the previously compared precedence models were varied in each room in order to identify if, and to what extent, the separation performance varied with these parameters. The results showed that the reverberation-performance of existing psychoacoustic engineering approaches to machine source separation can be improved and can yield significant gains in separation performance.
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Fulford, Ross Anthony. "Structure-borne sound power and source characterisation in multi-point-connected systems." Thesis, University of Liverpool, 1995. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.284262.
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Wilkie, Sonia. "The effect of audio cues and sound source stimuli on looming perception." Thesis, Queen Mary, University of London, 2015. http://qmro.qmul.ac.uk/xmlui/handle/123456789/8974.
Full textAbstract:
Objects that move in depth (looming) are ubiquitous in the real and virtual worlds. How humans interact and respond to these approaching objects may affect their continued survival in both the real and virtual words, and is dependent on the individual's capacity to accurately interpret depth and movement cues. In computer-generated environments, including hyper and virtual reality,film, and gaming, these cues are often complex sounds with multiple audio cues that are creatively designed for maximum effect. To accurately generate a dynamic and rich perception of looming objects, the design of such complex stimuli should be based on a firm scientific foundation that encompasses what we know about how people visually and aurally perceive events and interactions. Conversely, many psychological studies investigating auditory looming depict the object's movement using simple audio cues, such as an increase in the amplitude, which are applied to tones that are not regularly encountered in the natural world, such as sine, triangle, or square waves. Whilst the results from these studies have provided important information on human perception and responses, technological advances now allow us to present complex audiovisual stimuli and to collect measurements on human perception and responses to real and hyper-real stimuli. The research in this thesis begins to address the gap that exists between the research corpus and industry usage. This is initially accomplished by conducting a feature analysis of the audio cues and complex sounds constructed by sound designers for film scenes presenting objects moving in depth. This is followed by a perceptual study measuring human responses, both physical and emotional, to the complex audio cues designed for the film scenes. Using physical models, we then select a number of audio cues for closer inspection and introduce the parameter of `room reflections' as an audio cue. We investigate whether or not human responses to various audio cues differ when they are presented individually or in combination, or when they are applied to an artificial (square wave) sound source or a real world sound source. Finally, we test the capacity of these audio cues to bias multimodal auditory-visual perception of an approaching object.
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Itoyama, Katsutoshi. "Source Separation of Musical Instrument Sounds in Polyphonic Musical Audio Signal and Its Application." 京都大学 (Kyoto University), 2011. http://hdl.handle.net/2433/142116.
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Wang, Cheng-Kang, and 汪正剛. "Multiple Sound Source Direction Estimation and Sound Source Number Estimation." Thesis, 2008. http://ndltd.ncl.edu.tw/handle/23799296724314561439.
Full textAbstract:
碩士國立交通大學
電機與控制工程系所
96
This work proposes a multiple sound source direction estimation and sound source number estimation method by using distributed microphone array without the information of sound velocity. This work also proposes an eigenstructure-based generalized cross correlation method (ES-GCC) for estimating time delay between microphones. Upon obtaining the time delay information, the sound source direction and velocity can be estimated by least square method. In multiple sound source case, the time delay combination among microphones is arranged such that the estimated sound speed value falls within an acceptable range. By accumulating the estimation result of sound source direction and using adaptive K-means++ algorithm, we can estimate the sound source number and direction for each source. Experimental results are carried out in the real environment to evaluate the performance of the proposed algorithm.
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Azarfar, Alireza. "Biological sound source localization." Master's thesis, 2012. http://hdl.handle.net/10400.1/10835.
Full textAbstract:
Este projeto propõe um sistema de localização de uma fonte sonora, tanto no plano
horizontal e vertical, biologicamente inspirado. São implementados vários modelos,
desde o núcleo cochlear (NC) at´e ao col´ıculo inferior (CI), para a obtenção
de uma localização precisa e de confiança.
Para a localização sonora azimutal, a diferença de tempo interaural (DTI) ´e
extraída no núcleo medial da oliva superior (MOS), enquanto no núcleo lateral da
oliva superior (LOS) é extraída a diferença do nível interaural (DNI)e o DTI dos
envelopes dos sinoid. Estas características são combinadas no IC.
Além de células de deteção de coincidências no MOS e células sensíveis do
DNI no LOS, neurónios do tipo-V , do tipo-I e células de mapeamento do azimute
no CI são modeladas. É proposta uma distribuicão avançada de células no CI para
manter as funções da DTI a qualquer frequência dentro do intervalo fisiológico
da cabeça. De forma que as DTIs convirjam para um único resultado nas bandas
de frequências diferentes, as projeções adicionais do núcleo dorsal do lemnisco
lateral (NDLL) e o MOS são modelados.
Para localizar um som no plano vertical, o modelo beneficia de neurónios no
núcleo cochlear dorsal (NCD) que são sensíveis a “nothces” espectrais e das suas
projeções para neurónios do tipo-O no colículu inferior.
Testamos o nosso modelo numa cabeça robótica num ambiente de laboratório
e numa sala livre de eco, e também num conjunto de dados de uma função de
transferência relacionada com a cabeça (TFRC). Para comparar o nosso modelo
com um modelo matemáticos, implementamos um método baseado na correlação
cruzada generalizada (CCG) com ponderação PHAT. Os resultados experimentais
demonstram um bom desempenho, em caso de vários sons normais.
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Lin, Chi-Hao, and 林祺豪. "Probabilistic Structure from Sound and Sound Source Localization." Thesis, 2008. http://ndltd.ncl.edu.tw/handle/48001457364720395043.
Full textAbstract:
碩士國立臺灣大學
資訊工程學研究所
96
Auditory perception is one of the most important functions for robotics applications. Microphone arrays are widely used for auditory perception in which the spatial structure of microphones is usually known. The thesis first describes the affine Structure from Sound (SFS) algorithm. The structure from sound is a problem to simultaneously localize microphones and sound sources. However, the existing method does not take measurement uncertainty into account and does not provide uncertainty estimates of the SFS results. In this thesis, we propose a probabilistic structure from sound (PSFS) approach using the unscented transform. The PSFS algorithm not only localizes microphones and sound sources but also estimates the uncertainties of the SFS results. In addition, a probabilistic sound source localization (PSSL) approach using the PSFS results is provided to improve sound source localization accuracy. The ample results of simulation and experiments using low cost, off-the-shell microphones demonstrate the feasibility and performance of the proposed PSFS and PSSL approaches.
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Lin, Chi-Hao. "Probabilistic Structure from Sound and Sound Source Localization." 2008. http://www.cetd.com.tw/ec/thesisdetail.aspx?etdun=U0001-2807200820013400.
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Yang, Shan-hsiang, and 楊善翔. "Three-dimensional Sound Source Localization." Thesis, 2009. http://ndltd.ncl.edu.tw/handle/00117977278704040317.
Full textAbstract:
碩士國立臺灣科技大學
資訊工程系
97
In this thesis, we study and implement a system to detect the direction of a sound source in three-dimensional space. For the hardware part, an equilateral triangle microphone array composed of only three microphones is used to input the voice signals. For the software part, VAD (Voice Activity Detection), TDOA (Time Delay of Arrival) estimation and direction detection are executed in order. In the processing of VAD, we propose a spectral-entropy plus SNR-verification based method to distinguish speech/non-speech frames. To estimate TDOA, an approximation algorithm is used to compute a generalized cross correlation function. We propose a synchronous phase replication method to solve the problem of unstable phase. In addition, we propose a parabolic interpolation based method to increase the accuracy of estimated TDOA values. Then, the distances between the estimated vector of TDOA values and the vectors of theoretical value are computed in order to find the direction of a sound source. Also, the accuracy is improved by using interpolation. Furthermore, we propose a weighted voting mechanism to determine the final direction angle from the angles obtained in several speech frames. According to the results of on-line experiments, our system can do real-time processing by using small amount of computations. The averaged error of azimuth angle is 3.43 degrees and the averaged error of elevation is 2.08 degrees. Therefore, the overall performance of our sound source localization system is good.
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McNeese, Andrew Reed. "An investigation of the combustive sound source." Thesis, 2010. http://hdl.handle.net/2152/ETD-UT-2010-08-1699.
Full textAbstract:
This thesis describes the development and testing of the Combustive Sound
Source (CSS), which is a broadband underwater sound source. The CSS is being
developed as a clean, safe, and cost effective replacement to underwater explosive
charges, which exhibit an inherent danger to marine life and researchers using the
charges. The basic operation of the CSS is as follows. A combustible mixture
of gas is held below the surface of the water in a combustion chamber and ignited
with an electric spark. A combustion wave propagates through the mixture
and converts the fuel and oxidizer into a bubble of combustion products, which
expands due to an increase in temperature, and then ultimately collapses to a
smaller volume than before ignition, producing a high intensity, low frequency
acoustic signal. The thesis begins by discussing the background, history, and purpose
of developing the CSS. It continues by describing the current apparatus and
the essential components and convenient features added to the latest mechanical
design. The general operation is discussed along with a description of an experiment
conducted to determine the acoustic output and robustness of the current
CSS. The results of this experiment are presented in terms of the effect of volume,
ignition depth, oxidizing gas, combustion chamber size, and repeatability of
acoustic signatures. Discussion of apparatus robustness is presented to suggest
improvements for future CSS designs.text
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Lee, Po-Lin, and 李柏霖. "The Investigations on the Sound Field Generated by Moving Sound Source." Thesis, 2009. http://ndltd.ncl.edu.tw/handle/83380965990684888265.
Full textAbstract:
博士國立清華大學
動力機械工程學系
98
Abstract The Phenomenon of the sound field generated by a moving sound source has been investigated in the present work with two-part subject. The first part is to establish the mathematical model and the corresponding numerical scheme; the second part is to employ the established numerical scheme in practical acoustic problems. In order to establish the mathematical model, a novel governing equation is derived and it can be viewed as a modified Ffowcs Williams-Hawkings equation (FW-H-equation). The major characteristic of the novel governing equation is to include the interior and the exterior domain. In addition, the acoustic position vector, the quality describes the relations about the distance and the direction between the observer and the sound source, is represented for applying to the condition of the sound source moving with variable speed. As to the solution of the governing equation, it is expressed in the form of the Surface Integral Formula (SIF) by convoluting the free-space Green’s function in time domain. Then, this SIF is used to numerical implementation in the concept of the Boundary Element Method in time domain (BEMTD). After establishing the numerical scheme and verifying the correctness of the calculating results, two acoustic problems were investigated for revealing the ability of the numerical scheme. The first case considers that a moving line source with variable speed. This case reveals that the restriction of the constant speed is released in the established numerical scheme. For the simulation results, it shows that the effect of the variable speed not only influenced the variation rate of the frequency modulation, i.e., Doppler effect, but also the time about the maximum acoustic pressure being observed. In addition, the rate of the amplitude variation is shaper than that in the constant speed case when the line source is approaching to the observer point. The second case investigates the binaural hearing perceived by a moving sound source. For understanding the weighting of the eventful cues about perceiving the direction and the speed of the moving sound source, the sound pressure at the entrance of the external ear canal was calculated by the established numerical scheme. Furthermore, the Hilbert Huang transformation (HHT) is used to find the instantaneous frequencies of acoustic signal. Results show that the Interaural Level Difference (ILD) and the frequencies shifting are eventful than Interaural Time Difference (ITD). The perceived loudness level will be larger in the motional case than that found in the stationary case. These engineering problems are shown that the acoustic properties are different for comparing with the sound field generated by a moving sound source and that by a stationary sound source. As to the analytical methodology developed in the present work, it turns out that it indeed can be used to simulate and analyze these acoustic problems whenever the sound source moves or not. Furthermore, some meaningful phenomenon relating to these problems then can be observed through discussing the calculating results.
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Ling, Hung, and 林宏. "Study Sound Source Localization Based on TDOA." Thesis, 2009. http://ndltd.ncl.edu.tw/handle/33588336422359344240.
Full textAbstract:
碩士亞洲大學
資訊工程學系碩士班
97
Science and technology change with each passing day in recent years. The interface needs more accurate and convenient between machine and people, one of them used the microphone that respond to location of the sound source. This research used Microphone array to find the TDOA of sound source localization on Cross Correlation function. There are two methods of sound source localization. First kind of method is using TDOA on hyperbola. The other is using TDOA on neural network. The research perform simulation with MATLAB. Based on the simulation results ,the prove the feasibility of the research. And offer a reference of technical research in the future.
47
Wu, Xiang. "Binaural Sound Source Localisation in Complex Conditions." Phd thesis, 2020. http://hdl.handle.net/1885/205556.
Full textAbstract:
There has been a growing interest in the reproduction of human spatial hearing behaviours, arising from the development of spatial audio signal processing techniques. To accurately localise single or multiple sound sources using humanoid apparatus, it is essential to be able to exploit the spatial-related features of the human subject filtering effect, which requires an understanding of both the feature characteristics and the mapping relationship to the source locations. In this thesis, we analyse and evaluate the localisation feature characteristics of binaural signal, and explore a method for constructing a localisation mapping model.
As a result of the reflecting and diffracting of human-like apparatus, sound waves are filtered before being captured by the eardrum, and the filtering effects result in various behaviours in the frequency domain. This thesis first summarises the characteristics of those behaviours and evaluates their importance to localisation. We analyse and evaluate the correlation between source location and three main interaural cues, which are interaural level differences, interaural time difference and interaural phase difference. Then, we explore the process to exploit those features using, and develop a novel feature vector by combining the most valuable spectra. Following this, by employing mutual information as the evaluation metric for frequencies selection, we propose a new feature location mapping model that embeds the feature evaluation process. The new mapping uses a multiple-tree structured model based on the random forest that shows high tolerance to noise. Through computational simulations and practical experiments, the model presents an improvement in both accuracy and robustness according to the comparison of the angular error and localisation correct rate. Finally, by combining our localisation method with the recent proposed direct path transfer function estimation method based on a convolutive transfer function model, we design a binaural localisation system for an unknown environment.
The remainder of this thesis demonstrates the possibility of using the active localisation cues in a binaural system. Based on observations of human active head rotation behaviour, we investigate the effect of dynamic features in binaural localisation. The analysis shows that head rotation enriches the variation of localisation features, which resolve the problem of cone-of-confusion and simplifies vertical-wise localisation in a 3-D space. In addition, we develop a multiple-source localisation method based on the head rotation process, which indicates that dynamic features would be the solution to many localisation problems caused by the limitation on the number of signal channels.
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Κεττένης, Χρίστος. "Σύστημα εντοπισμού ηχητικής πηγής." Thesis, 2011. http://nemertes.lis.upatras.gr/jspui/handle/10889/4390.
Full textAbstract:
Στη παρούσα Διπλωματική εργασία μελετήθηκε, σχεδιάστηκε και υλοποιήθηκε ένα σύστημα εντοπισμού θέσης ηχητικής πηγής. Συγκεκριμένα, αυτή η πηγή
μπορεί να είναι ο κτύπος των δακτύλων ενός χρήστη πάνω στην επιφάνεια ενός τραπεζιού, καθώς “πληκτρολογεί”, ή ο ήχος ενός μολυβιού ή μίας
κιμωλίας που παράγεται από το γράψιμο σε χαρτί ή σε έναν πίνακα, αντιστοίχως. Στόχος της εφαρμογής αυτής είναι η μετατροπή της επιφάνεια ενός
γραφείου ή ενός πίνακα σε ένα “φτηνό” αλλά αποδοτικό ηλεκτρονικό μέσο εισόδου γραφικών ή εγγραφής, δηλαδή σε μία ηλεκτρονική ταμπλέτα. Τελικώς,
αυτή η εργασία επικεντρώνεται στα προβλήματα τα οποία μειώνουν την ακρίβεια αυτού του εντοπισμού και επιπλέον σχεδιάστηκε και κατασκευάστηκε
το υλικό και το λογισμικό που απαιτείται για την υποστήριξη του προαναφερθέντος συστήματος.In this diploma Thesis a sound source positioning system, was studied, designed and implemented. The sound, in particular, could be produced by "tapping a keyboard" onto a table surface or it is the noise produced while writing with a pencil or a piece of chalk, onto paper or on the surface of a blackboard, respectively. The aim of this application is to transform the surface of a desk or a blackboard into a "cheap" but effective electronic input device, in other, words an electronic tablet. Lastly, this Thesis is focused on the problems causing the reduction of accuracy in estimating the posistion of the acoustic source and also on the design and construction of the hardware and software that support the produced system.
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Goel, Priyank. "Harmonic Sound Source Separation in Monaural Music Signals." Thesis, 2013. http://hdl.handle.net/2005/2803.
Full textAbstract:
Sound Source Separation refers to separating sound signals according to their sources from a given observed sound. It is efficient to code and very easy to analyze and manipulate sounds from individual sources separately than in a mixture. This thesis deals with the problem of source separation in monaural recordings of harmonic musical instruments. A good amount of literature is surveyed and presented since sound source separation has been tried by many researchers over many decades through various approaches.
A prediction driven approach is first presented which is inspired by old-plus-new heuristic used by humans for Auditory Scene Analysis. In this approach, the signals from different sources are predicted using a general model and then these predictions are reconciled with observed sound to get the separated signal. This approach failed for real world sound recordings in which the spectrum of the source signals change very dynamically.
Considering the dynamic nature of the spectrums, an approach which uses covariance matrix of amplitudes of harmonics is proposed. The overlapping and non-overlapping harmonics of the notes are first identified with the knowledge of pitch of the notes. The notes are matched on the basis of their covariance profiles. The second order properties of overlapping harmonics of a note are estimated with the use of co-variance matrix of a matching note. The full harmonic is then reconstructed using these second order characteristics. The technique has performed well over sound samples taken from RWC musical Instrument database.
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Chung, Yi-Hao, and 鍾毅豪. "A bi-sensor system for sound source estimation." Thesis, 1999. http://ndltd.ncl.edu.tw/handle/90880198860324681031.
Full textAbstract:
碩士淡江大學
資訊工程學系
87
Due to the great advance in the technique of microprocessor and the sharp drop of cost, digital signal processing applications are widely spread. In addition, the array signal processing technique has been vibrantly developed, no matter in noise deduction or in signal enhancement we have achieved certain degree of satisfactory result. Another interesting application of array signal processing is to use multi-sensors to estimate the signal source. In this thesis we investigate methods in array signal processing and implement a system for estimating the source location of a sound. Our system is a real-time embedded system implemented with Texas Instrument TMS320C542 StarterKit. Like all animals with two ears, we use two sensors (microphones) to receive input sound and estimate the location of the sound source. We shall apply all benefits from array signal processing in noise deduction, signal enhancement and sound source estimation in our system. We hope our system can be further developed a hearing aid system for human being.