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Showing papers on "Acoustic source localization published in 2009"


Proceedings ArticleDOI
10 Oct 2009
TL;DR: A new localization system “Selective Attention System”, implemented into a humanoid robot, and the experimental validation is successfully verified even when the robot microphones move dynamically, addressing sound source localization working in dynamic environments for robots.
Abstract: As robotic technology plays an increasing role in human lives, “robot audition”, human-robot communication, is of great interest, and robot audition needs to be robust and adaptable for dynamic environments. This paper addresses sound source localization working in dynamic environments for robots. Previously, noise robustness and dynamic localized sound selection have been enormous issues for practical use. To correct the issues, a new localization system “Selective Attention System” is proposed. The system has four new functions: localization with Generalized EigenValue Decomposition of correlation matrices for noise robustness(“Localization with GEVD”), sound source cancellation and focus (“Target Source Selection”), human-like dynamic Focus of Attention (“Dynamic FoA”), and correlation matrix estimation for robotic head rotation (“Correlation Matrix Estimation”). All are achieved by the dynamic design of correlation matrices. The system is implemented into a humanoid robot, and the experimental validation is successfully verified even when the robot microphones move dynamically.

99 citations


Journal ArticleDOI
TL;DR: In this article, a characterization of how these unknown nonidealities degrade direction-finding accuracy, via Cramer-Rao bound analysis, is presented. But the authors do not consider the effect of the unknown non-ideality in the acoustic vector sensor's gain response, phase response, collocation or orthogonal orientation among its constituent velocity sensors.
Abstract: An acoustic vector-sensor (also known as vector-hydrophone in underwater applications) is composed of two or three spatially collocated but orthogonally oriented acoustic velocity sensors, plus possibly a collocated acoustic pressure sensor. Such an acoustic vector sensor is versatile for direction-finding, due to its azimuth-elevation spatial response's independence from the incident source's frequency, and bandwidth. However, previously unavailable in the open literature is how the acoustic vector sensor's far-field direction-of-arrival estimates may be adversely affected by any unknown nonideality in the acoustic vector sensor's gain response, phase response, collocation, or orthogonal orientation among its constituent velocity sensors. This paper pioneers a characterization of how these various unknown nonidealities degrade direction-finding accuracy, via Cramer-Rao bound analysis.

83 citations


Proceedings ArticleDOI
10 Oct 2009
TL;DR: The localization of multiple sound sources in the 3D-space based on the MUSIC algorithm was implemented and evaluated in a humanoid robot embedded in real noisy environments, achieving localization accuracies and insertion rates comparable with the case where the ideal number of sources is given.
Abstract: With the goal of improving human-robot speech communication, the localization of multiple sound sources in the 3D-space based on the MUSIC algorithm was implemented and evaluated in a humanoid robot embedded in real noisy environments. The effects of several parameters related to the MUSIC algorithm on sound source localization and real-time performances were evaluated, for recordings in different environments. Real-time processing could be achieved by reducing the frame size to 4 ms, without degrading the sound localization performance. A method was also proposed for determination of the number of sources, which is an important parameter that influences the performance of the MUSIC algorithm. The proposed method achieved localization accuracies and insertion rates comparable with the case where the ideal number of sources is given.

82 citations


Journal ArticleDOI
TL;DR: In this article, the authors describe a method to derive information about the acoustic emission of a flow using particle image velocimetry (PIV) data, which allows studying sound sources, the related flow phenomena and their acoustic radiation into the far field, simultaneously.
Abstract: The present paper describes a method to derive information about the acoustic emission of a flow using particle image velocimetry (PIV) data The advantage of the method is that it allows studying sound sources, the related flow phenomena and their acoustic radiation into the far field, simultaneously In a first step the time history of two-dimensional instantaneous pressure fields is derived from planar PIV data In a successive step the Curle’s acoustic analogy is applied to the pressure data to obtain the acoustic radiation of the flow The test cases studied here are two rectangular cavity flows at very low Mach number with different aspect ratios L/H The main sound source is located at the cavity trailing edge and it is due to the impingement of vortices shed in the shear layer It is shown that the flow emits sound with a main directivity in the upstream direction for the smaller aspect ratio and the directivity is more uniform for the larger aspect ratio In the latter case the acoustic pressure spectra has a broader character due to the impact of the downstream recirculation zone onto the shear layer instabilities, destroying their regular pattern and alternating the main sound source

61 citations


Journal ArticleDOI
01 Dec 2009
TL;DR: It is shown that the AML source localization algorithm can be used to localize actual animals in their natural habitat using a platform that is practical to deploy and ambiguities from spatial aliasing of high frequency signals are readily eliminated.
Abstract: Field biologists use animal sounds to discover the presence of individuals and to study their behavior. Collecting bio-acoustic data has traditionally been a difficult and time-consuming process in which researchers use portable microphones to record sounds while taking notes of their own detailed observations. The recent development of new deployable acoustic sensor platforms presents opportunities to develop automated tools for bio-acoustic field research. In this work, we implement both two-dimensional (2D) and three-dimensional (3D) AML-based source localization algorithms. The 2D algorithm is used to localize marmot alarm-calls of marmots on the meadow ground. The 3D algorithm is used to localize the song of Acorn Woodpecker and Mexican Antthrush birds situated above the ground. We assess the performance of these techniques based on the results from four field experiments: two controlled test of direction-of-arrival (DOA) accuracy using a pre-recorded source signal for 2D and 3D analysis, an experiment to detect and localize actual animals in their habitat, with a comparison to ground truth gathered from human observations, and a controlled test of localization experiment using pre-recorded source to enable careful ground truth measurements. Although small arrays yield ambiguities from spatial aliasing of high frequency signals, we show that these ambiguities are readily eliminated by proper bearing crossings of the DOAs from several arrays. These results show that the AML source localization algorithm can be used to localize actual animals in their natural habitat using a platform that is practical to deploy.

60 citations


Journal ArticleDOI
TL;DR: In this paper, three common methods, namely the equivalent radiated power, the lumped parameter model and an approximation based on the volume velocity, are investigated for estimating the radiated sound power functions of a car and the radiation of a diesel engine under realistic load cases.
Abstract: The radiated sound power value is often used to evaluate the sound radiation of a machine or a product. Since its estimation requires the sound pressure on a surrounding surface of the radiating object, the sound power value is mostly computed under high numerical costs due to the acoustic field that has to be modeled. Therefore, approximations of the sound pressure are widely popular. In this article three common methods namely the equivalent radiated power, the lumped parameter model and an approximation based on the volume velocity are investigated. It is the goal of this paper to test these methods on realistic examples. The radiated sound power functions of the floor panel of a car and the radiation of a diesel engine under realistic load cases are estimated.

57 citations


Journal ArticleDOI
TL;DR: In this article, a sound propagation through a rarefied gas is investigated on the basis of the linearized kinetic equation taking into account the influence of receptor, and the kinetic equation is solved via a discrete velocity method with a numerical error of 0.1%.
Abstract: A sound propagation through a rarefied gas is investigated on the basis of the linearized kinetic equation taking into account the influence of receptor. A plate oscillating in the normal direction to its own plane is considered as a sound source, while a stationary parallel plate is considered as being the receptor of sound. The main parameters determining the solution of the problem are the oscillation speed parameter, which is defined as the ratio of intermolecular collision frequency to the sound frequency, and the rarefaction parameter defined as the ratio of the distance between source and receptor to the molecular mean free path. The kinetic equation is solved via a discrete velocity method with a numerical error of 0.1%. The numerical calculations are carried out for wide ranges of the oscillation and rarefaction parameters. The concept of integral phase parameter is introduced to obtain the sound speed correctly in all regimes of the gas rarefaction and sound frequency. Analytical solutions are obtained in the limits of small and large parameters of frequency and rarefaction.

54 citations


Journal ArticleDOI
TL;DR: A theoretical analysis of the primary and secondary sound fields around a spherical sound source reveals that the natural quiet zones for the spherical source have a shell-shape, showing potential for quiet zones with extents that are significantly larger than the well-known limit of a tenth of a wavelength for monopole sources.
Abstract: Active control of sound has been employed to reduce noise levels around listeners’ head using destructive interference from noise-canceling sound sources. Recently, spherical loudspeaker arrays have been studied as multiple-channel sound sources, capable of generating sound fields with high complexity. In this paper, the potential use of a spherical loudspeaker array for local active control of sound is investigated. A theoretical analysis of the primary and secondary sound fields around a spherical sound source reveals that the natural quiet zones for the spherical source have a shell-shape. Using numerical optimization, quiet zones with other shapes are designed, showing potential for quiet zones with extents that are significantly larger than the well-known limit of a tenth of a wavelength for monopole sources. The paper presents several simulation examples showing quiet zones in various configurations.

51 citations


Journal ArticleDOI
TL;DR: In this article, the authors considered the problem of reconstructing 2D temperature and wind fields by using acoustic tomography setups and showed that the classical time-of-flight measurements are not sufficient to reconstruct wind fields.
Abstract: Acoustic tomography is a type of inverse problem. The idea of estimating physical quantities that influence sound propagation by measuring the parameters of propagation has proven to be successful in many practical domains, including temperature and wind estimation in the atmosphere. However, in most of the previous work in this area, the algorithms used have not been proven mathematically to provide the correct solution to the inverse problem. This paper considers the problem of reconstructing 2D temperature and wind fields by using acoustic tomography setups. Primarily, it shows that the classical time-of-flight measurements are not sufficient to reconstruct wind fields. As a solution, an additional set of measurements related solely to the parameters of sound propagation—more precisely, to the angles of departure/arrival of sound waves—is suggested. To take the full benefit of this additional information, the bent-ray model of sound propagation is introduced. In this work, it is also shown that, when a temperature and a source-free 2D wind field are observed on bounded domains, the complete reconstruction is possible using only measurements of the time of flight. Conversely, the angles of departures/arrivals are sufficient to reconstruct a temperature and a curl-free 2D wind fields on bounded domains. Further, an iterative reconstruction algorithm is proposed and possible variations to the main scheme are discussed. Finally, the performed numerical simulations confirm the theoretical results, demonstrate fast convergence, and show the advantages of the adopted bent-ray model for sound propagation over the straight-ray model.

50 citations


Journal ArticleDOI
TL;DR: Error sensitivity considerations indicate that ESM-based NAH is less sensitive to measurement errors when it is based on particle velocity input data than when it was based on measurements of sound pressure data, and this is confirmed by a simulation study and by experimental results.
Abstract: The advantage of using the normal component of the particle velocity rather than the sound pressure in the hologram plane as the input of conventional spatial Fourier transform based near field acoustic holography (NAH) and also as the input of the statistically optimized variant of NAH has recently been demonstrated. This paper examines whether there might be a similar advantage in using the particle velocity as the input of NAH based on the equivalent source method (ESM). Error sensitivity considerations indicate that ESM-based NAH is less sensitive to measurement errors when it is based on particle velocity input data than when it is based on measurements of sound pressure data, and this is confirmed by a simulation study and by experimental results. A method that combines pressure- and particle velocity-based reconstructions in order to distinguish between contributions to the sound field generated by sources on the two sides of the hologram plane is also examined.

40 citations


Proceedings ArticleDOI
19 Apr 2009
TL;DR: A way to produce an acoustical map of the scene by computing the averaged directivity pattern of BSS demixing systems, which allows application for multiple dimensions, in the near field as well as in the far field.
Abstract: In this paper, we propose a versatile acoustic source localization framework exploiting the self-steering capability of Blind Source Separation (BSS) algorithms. We provide a way to produce an acoustical map of the scene by computing the averaged directivity pattern of BSS demixing systems. Since BSS explicitly accounts for multiple sources in its signal propagation model, several simultaneously active sound sources can be located using this method. Moreover, the framework is suitable to any microphone array geometry, which allows application for multiple dimensions, in the near field as well as in the far field. Experiments demonstrate the efficiency of the proposed scheme in a reverberant environment for the localization of speech sources.

Patent
01 Apr 2009
TL;DR: In this article, an acoustic signal is transmitted into a formation with a source oriented in a first source orientation, and an acoustic waveform is received in response with a receiver oriented in the first direction.
Abstract: In an acoustic logging system utilizing one or more acoustic sources, each with a specified radiation pattern around a source orientation, an acoustic signal is transmitted into a formation with a source oriented in a first source orientation. An acoustic waveform is received in response with a receiver oriented in a first direction. The slowness of the formation in the first direction is calculated using the received acoustic waveform.

Patent
08 Apr 2009
TL;DR: In this article, a system for locating and identifying an acoustic event is described, where an acoustic sensor (20) having a pair of concentric opposing microphones (24, 26) at a fixed distance on a microphone axis (MA) is used to measure an acoustic intensity, from which a vector incorporating the acoustic event was identified.
Abstract: A system (10) is provided for locating and identifying an acoustic event. An acoustic sensor (20) having a pair of concentric opposing microphones (24, 26) at a fixed distance on a microphone axis (MA) is used to measure an acoustic intensity, from which a vector incorporating the acoustic event is identified. A second acoustic sensor or movement of the first acoustic sensor (20) is used to provide a second vector incorporating the acoustic event. Combination of the first and the second vector locates the acoustic event in space. A command unit (120) in communication with the acoustic sensors (20) can be used for combining the vectors as well as comparing a signal spectra of the acoustic event to stored identified spectra to provide an identification of acoustic event.

Journal ArticleDOI
TL;DR: A new theory is proposed for the reconstruction of curl-free vector field, whose divergence serves as acoustic source, and the numerical results suggest that reconstruction of a vector field using the proposed theory is not sensitive to variation in the detecting distance.
Abstract: A new theory is proposed for the reconstruction of curl-free vector field, whose divergence serves as acoustic source. The theory is applied to reconstruct vector acoustic sources from the scalar acoustic signals measured on a surface enclosing the source area. It is shown that, under certain conditions, the scalar acoustic measurements can be vectorized according to the known measurement geometry and subsequently be used to reconstruct the original vector field. Theoretically, this method extends the application domain of the existing acoustic reciprocity principle from a scalar field to a vector field, indicating that the stimulating vectorial source and the transmitted acoustic pressure vector (acoustic pressure vectorized according to certain measurement geometry) are interchangeable. Computer simulation studies were conducted to evaluate the proposed theory, and the numerical results suggest that reconstruction of a vector field using the proposed theory is not sensitive to variation in the detecting distance. The present theory may be applied to magnetoacoustic tomography with magnetic induction (MAT-MI) for reconstructing current distribution from acoustic measurements. A simulation on MAT-MI shows that, compared to existing methods, the present method can give an accurate estimation on the source current distribution and a better conductivity reconstruction.

Journal ArticleDOI
TL;DR: A process layer is used to recover the pressure field that the studied source would have radiated in free space, which requires the knowledge of both acoustic pressure and velocity fields on a closed surface surrounding the source.
Abstract: Due to excessive reverberation or to the presence of secondary noise sources, characterization of sound sources in enclosed space is rather difficult to perform. In this paper a process layer is used to recover the pressure field that the studied source would have radiated in free space. This technique requires the knowledge of both acoustic pressure and velocity fields on a closed surface surrounding the source. The calculation makes use of boundary element method and is performed in two steps. First, the outgoing pressure field is extracted from the measured data using a separation technique. Second, the incoming field then scattered by the tested source body is subtracted from the outgoing field to recover free field conditions. The studied source is a rectangular parallelepiped with seven mid-range loudspeakers mounted on it. It stands at 40 cm from the rigid ground of a semi-anechoic chamber which strongly modifies the radiated pressure field, especially on the underside. After the measured data have been processed, the loudspeaker positions are recovered with a fairly good accuracy. The acoustic inverse problem is also solved to calculate the velocity field on the source surface.

Journal ArticleDOI
TL;DR: In this article, a sound field generated under a flat or concave reflector was studied by both experimental measurement and numerical calculation, and the calculated result agreed well with the experimental data.
Abstract: Using a standing-wave field generated between a sound source and a reflector, it is possible to trap small objects at nodes of the sound pressure distribution in air. In this study, a sound field generated under a flat or concave reflector was studied by both experimental measurement and numerical calculation. The calculated result agrees well with the experimental data. The maximum force generated between a sound source of 25.0 mm diameter and a concave reflector is 0.8 mN in the experiment. A steel ball of 2.0 mm in diameter was levitated in the sound field in air.

Journal ArticleDOI
TL;DR: In this paper, a new technique based on the time reversal sink concept is used to detect active sound sources with a limited number of measurement points, which allows high-resolution imaging and provides a new method of characterization and detection of sound sources.
Abstract: Numerous practical applications ― such as non destructive evaluation of industrial structures, acoustic characterization of musical instruments, and acoustic mapping of sound sources in a known propagation medium ― involve source detection and characterization. In the past, this problem has been investigated using different beamforming and backpropagation methods. In this work, a new technique, based on the time reversal sink concept, is used to detect active sound sources with a limited number of measurement points. The theory and application of super-resolution focusing of sound and vibration using a time-reversal sink (TRS) have been studied, both in ultrasonic regime and in audible range. A high-resolution imaging technique based on a numerical time reversal sink has recently been developed by the authors for vibrational imaging of active sources in a dispersive medium. In this paper, the numerical time reversal sink imaging technique is adapted to the case of high-resolution acoustic imaging of active sound sources in a three-dimensional free field. This technique allows high-resolution imaging and provides a new method of characterization and detection of sound sources. All results show the high resolution imaging capabilities of this new technique when compared with classical time-reversal (TR) backpropagation. More than simply detecting the position of the acoustic source, this technique allows to detect the size of the active sources. This technique provides an alternative to other imaging and source detection techniques, such as three-dimensional acoustic holography and beamforming.

Journal ArticleDOI
TL;DR: In this article, an analysis of the acoustic wave propagation in a model power transformer is presented, where partial discharges are simulated at predefined coordinates in the core and winding of the transformer.
Abstract: Analyses of acoustic wave propagation in a model power transformer are presented in the paper. The acoustic wave is induced by partial discharges that are simulated at predefined coordinates in the core and winding. Propagation of the numerical calculated acoustic wave is analyzed within the transient state. Achieved results indicate that the space and time distributions of the acoustic pressure depend on the induction position. Furthermore, a greater pressure gradient is observed in domains with higher speed of sound while the largest amplitude occurs at the vicinity of the induction position.

Journal Article
TL;DR: In this paper, a statistical evaluation of the binaural sound source localization performance during listening tests by human subjects was conducted, and it was shown that the convolution of a measured head-related transfer function (HRTF) with the room impulse response generated by a hybrid image source model with a stochastic scattering process using secondary sources provides an adequate model for the prediction of Binaural room impulse responses (BRIR) for directional sound localization in the frontal horizontal plane.
Abstract: On the basis of a statistical evaluation of the binaural sound source localization performance during listening tests by human subjects, it is shown that the convolution of a measured head-related transfer function (HRTF) with the room impulse response generated by a hybrid image source model with a stochastic scattering process using secondary sources provides an adequate model for the prediction of binaural room impulse responses (BRIR) for directional sound localization in the frontal horizontal plane. The source localization performance for sound stimuli presented to test subjects in a natural way was compared to that presented via headphones. Listening via headphones tends to decrease the localization performance only, and only slightly when localizing narrow-band high-frequency stimuli. Binaural sound presented to test subjects via headphones was generated on the basis of measurements and simulations. Two different headphone conditions were evaluated. The overall localization performance for simulated headphone sound obtained using a simulated BRIR was found to be equivalent to the one using a measured BRIR. The study also confirms expectations that the deterioration of sound source localization performance in reverberant rooms is closely linked to the direct-to-reverberant ratio for given loudspeaker and listener positions, rather than to the reverberation time of the room as such.

Journal ArticleDOI
TL;DR: A search space clustering method designed to speed up the SRP-PHAT based sound source localization algorithm for intelligent home robots equipped with small scale microphone arrays is proposed.
Abstract: Sound source localization (SSL) is a major function of robot auditory systems for intelligent home robots. The steered response power-phase transform (SRP-PHAT) is a widely used method for robust SSL. However, it is too slow to run in real time, since SRP-PHAT searches a large number of candidate sound source locations. This paper proposes a search space clustering method designed to speed up the SRP-PHAT based sound source localization algorithm for intelligent home robots equipped with small scale microphone arrays. The proposed method reduces the number of candidate sound source locations by 30.6% and achieves 46.7% error reduction compared to conventional methods.

Journal ArticleDOI
TL;DR: A sound speed estimation method in which an optimal sound speed, the speed that makes the echo signal delays at the transducer elements be best matched to the theoretical delays, is estimated by maximizing the beamformed echo signal amplitude with respect to the sound speed and the reflector displacement from the central axis of the ultrasound beam is proposed.

Journal ArticleDOI
TL;DR: An autonomous mobile robotic system for finding, investigating, and modeling ambient noise sources in the environment, improving upon the initial localization accuracy, identifying volume and directivity, and building a classification vector useful for detecting the sound source in the future is described.
Abstract: In this work, we describe an autonomous mobile robotic system for finding, investigating, and modeling ambient noise sources in the environment. The system has been fully implemented in two different environments, using two different robotic platforms and a variety of sound source types. Making use of a two-step approach to autonomous exploration of the auditory scene, the robot first quickly moves through the environment to find and roughly localize unknown sound sources using the auditory evidence grid algorithm. Then, using the knowledge gained from the initial exploration, the robot investigates each source in more depth, improving upon the initial localization accuracy, identifying volume and directivity, and, finally, building a classification vector useful for detecting the sound source in the future.

Proceedings ArticleDOI
11 May 2009
TL;DR: In this paper, the performance of two types of four-microphone tetrahedral probes is investigated in the context of more fully characterizing rocket noise source regions, which is required to determine the vibroacoustic impact on flight hardware and structures in the vicinity of the launch pad.
Abstract: Energy-based acoustical measurements are investigated in the context of more fully characterizing rocket noise source regions. Near-field measurements made on statically fired GEM-60 motors are described and the performance of two types of four-microphone tetrahedral probes is discussed. Vector intensity plots reveal the magnitude and directionality of the near-field sound radiation as a function of frequency, position, and time in the plume. HE development of the next-generation space flight vehicles has prompted renewed interest regarding source characterization and near-field propagation models of rocket noise. This source characterization is required to determine the vibroacoustic impact on flight hardware and structures in the vicinity of the launch pad. Brigham Young University has been involved in an effort to develop and validate an energy-based acoustic probe suitable for use in rocket fields, in particular the RS-68B engine to be used on the Ares V vehicle. Energy-based acoustical measurements require estimation of both the collocated acoustic pressure and the threedimensional particle velocity. From the pressure, a scalar, and the particle velocity vector, a number of energybased quantities can be calculated, including vector acoustic intensity, specific acoustic impedance, potential, kinetic, and total energy densities, and the Lagrangian density. Knowledge of one or more of these quantities may

Journal ArticleDOI
TL;DR: The aim was to develop a simple, reliable tool for predicting enclosure insertion loss using statistical energy analysis and 12 out of the 15 test configurations were found to be reliable and were compared with theoretical models, which in fact correlate closely with the experimental work.
Abstract: Among noise control techniques, enclosures are widely used. It is known that enclosure acoustic efficiency is strongly influenced by the presence of openings or leaks. Modeling of diffuse field sound transmission loss (TL) of apertures and slits is therefore critical when the enclosure acoustic performance characteristics need to be predicted with confidence either for design or for modifying existing enclosures. Recently, a general model for diffuse field sound TL of rectangular and circular apertures has been developed and validated with respect to existing analytical or numerical models. This paper presents an experimental validation of this new model. The aim was to develop a simple, reliable tool for predicting enclosure insertion loss using statistical energy analysis. Twelve out of the 15 test configurations were found to be reliable and were compared with theoretical models, which in fact correlate closely (without adjustment) with the experimental work.

Patent
06 Jul 2009
TL;DR: In this article, the authors present a method for detecting the position of a sound source using at least two microphones, which includes the steps of: emitting a reproduced sound from the sound source, observing the reproduced sound and an observed sound at the microphones, converting the reproduced sounds and the observed sound into electrical signals, transforming the signals of the reproduced and of the observed sounds into frequency spectra by a frequency spectrum transformer apparatus, calculating Crosspower Spectrum Phase (CSP) coefficients of the signals by a CSP coefficient calculator apparatus, and calculating distances between the position and the positions of
Abstract: A position detection method, system, and computer readable article of manufacture tangibly embodying computer readable instructions for executing the method for detecting the position of a sound source using at least two microphones. The method includes the steps of: emitting a reproduced sound from the sound source; observing the reproduced sound and an observed sound at the microphones; converting the reproduced sound and the observed sound into electrical signals; transforming the signals of the reproduced sound and of the observed sound into frequency spectra by a frequency spectrum transformer apparatus; calculating Crosspower Spectrum Phase (CSP) coefficients of the frequency spectra of the signals by a CSP coefficient calculator apparatus; and calculating distances between the position of the sound source and the positions of the microphones based on the calculated CSP coefficients by a distance calculating apparatus, thereby detecting the position of the sound source.

Journal ArticleDOI
TL;DR: An approach is proposed in principle that could allow cavitation to be included within the proposed definitions of acoustic dose and acoustic dose-rate.
Abstract: Acoustic dose is defined as the energy deposited by absorption of an acoustic wave per unit mass of the medium supporting the wave. Expressions for acoustic dose and acoustic dose-rate are given for plane-wave conditions, including temporal and frequency dependencies of energy deposition. The relationship between the acoustic dose-rate and the resulting temperature increase is explored, as is the relationship between acoustic dose-rate and radiation force. Energy transfer from the wave to the medium by means of acoustic cavitation is considered, and an approach is proposed in principle that could allow cavitation to be included within the proposed definitions of acoustic dose and acoustic dose-rate.

Proceedings ArticleDOI
12 May 2009
TL;DR: This work proposes a framework that simultaneously localizes the mobile robot and multiple sound sources using a microphone array on the robot and an eigenstructure-based generalized cross correlation method for estimating time delays between microphones under multi-source environments.
Abstract: Sound source localization is an important function in robot audition. The existing works perform sound source localization using static microphone arrays. This work proposes a framework that simultaneously localizes the mobile robot and multiple sound sources using a microphone array on the robot. First, an eigenstructure-based generalized cross correlation method for estimating time delays between microphones under multi-source environments is described. A method to compute the far field source directions as well as the speed of sound using the estimated time delays is proposed. In addition, the correctness of the sound speed estimate is utilized to eliminate spurious sources, which greatly enhances the robustness of sound source detection. The arrival angles of the detected sound sources are used as observations in a bearings-only SLAM procedure. As the source signals are not persistent and there is no identification of the signal content, data association is unknown which is solved using FastSLAM. The experimental results demonstrate the effectiveness of the proposed approaches.

Journal ArticleDOI
TL;DR: In this article, a double-leaf metallic panel partition clamp mounted on a rigid acoustic baffle has been investigated analytically, and it is found that at a specific frequency, sound transmission through the structure is significantly enhanced for incident sound with a certain incidence angle.

Patent
18 Nov 2009
TL;DR: In this article, the angular position of a sound source of a first sound wave as seen from the apparatus is determined on the basis of moments of arrival of the second sound wave at the respective sound sensors which are represented by the electric signals.
Abstract: An apparatus includes sound sensors for converting sounds into electric signals. An angular position of a sound source of a first sound wave as seen from the apparatus is determined on the basis of moments of arrival of the first sound wave at the respective sound sensors which are represented by the electric signals. An angular position of a sound source of a second sound wave as seen from the apparatus is determined on the basis of moments of arrival of the second sound wave at the respective sound sensors which are represented by the electric signals. Calculation is given of a relative angle between the determined angular position of the sound source of the first sound wave and the determined angular position of the sound source of the second sound wave. A condition of the apparatus is controlled in response to the calculated relative angle.

Journal ArticleDOI
TL;DR: In this article, the authors measured the sound absorption coefficient of flat panels subject to small angle sound incidence, in an industrial hall using an experimental device equipped with an acoustic array, where the directivity of this array has been optimized so that the major part of the received acoustic energy would come from one portion only of the investigated facing, attenuating the reflected beams due to the reverberation.