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Showing papers on "Acoustic source localization published in 2008"


Journal ArticleDOI
TL;DR: In this article, the authors proposed a topology optimization method based on continuous material interpolation functions in the density and bulk modulus to reduce the sound pressure amplitude in the shadow zone behind a sound barrier.

282 citations


Journal ArticleDOI
TL;DR: A unified maximum likelihood framework of these two techniques is presented, and it is demonstrated how such a framework can be adapted to create efficient SSL and beamforming algorithms for reverberant rooms and unknown directional patterns of microphones.
Abstract: In distributed meeting applications, microphone arrays have been widely used to capture superior speech sound and perform speaker localization through sound source localization (SSL) and beamforming. This paper presents a unified maximum likelihood framework of these two techniques, and demonstrates how such a framework can be adapted to create efficient SSL and beamforming algorithms for reverberant rooms and unknown directional patterns of microphones. The proposed method is closely related to steered response power-based algorithms, which are known to work extremely well in real-world environments. We demonstrate the effectiveness of the proposed method on challenging synthetic and real-world datasets, including over six hours of recorded meetings.

199 citations


Journal ArticleDOI
TL;DR: A sound-intensity-dependent mechanism for compensating for the small ITD cues in the coincidence detector neurons in the nucleus laminaris (NL) of the chicken aged from 3 to 29 d after hatching is described.
Abstract: Interaural time difference (ITD) is a major cue for sound source localization. However, animals with small heads experience small ITDs, making ITD detection difficult, particularly for low-frequency sound. Here, we describe a sound-intensity-dependent mechanism for compensating for the small ITD cues in the coincidence detector neurons in the nucleus laminaris (NL) of the chicken aged from 3 to 29 d after hatching. The hypothesized compensation mechanisms were confirmed by simulation. In vivo single-unit recordings revealed an improved contrast of ITD tuning in low-best-frequency (

77 citations


Journal ArticleDOI
TL;DR: It was shown that noise reduction algorithms can have a large influence on localization and that the ADM only preserves localization in the forward direction over azimuths where limited or no noise reduction is obtained, and the statistical Wiener filter approach introduces a better combination of sound source localization and noise reduction performance than theADM approach.
Abstract: This paper evaluates the influence of three multimicrophone noise reduction algorithms on the ability to localize sound sources. Two recently developed noise reduction techniques for binaural hearing aids were evaluated, namely, the binaural multichannel Wiener filter (MWF) and the binaural multichannel Wiener filter with partial noise estimate (MWF-N), together with a dual-monaural adaptive directional microphone (ADM), which is a widely used noise reduction approach in commercial hearing aids. The influence of the different algorithms on perceived sound source localization and their noise reduction performance was evaluated. It is shown that noise reduction algorithms can have a large influence on localization and that (a) the ADM only preserves localization in the forward direction over azimuths where limited or no noise reduction is obtained; (b) the MWF preserves localization of the target speech component but may distort localization of the noise component. The latter is dependent on signal-to-noise ratio and masking effects; (c) the MWF-N enables correct localization of both the speech and the noise components; (d) the statistical Wiener filter approach introduces a better combination of sound source localization and noise reduction performance than the ADM approach.

77 citations


Patent
18 Jan 2008
TL;DR: In this article, a sound zoom method, medium, and apparatus generating a signal in which a target sound is removed from sound signals input to a microphone-array by adjusting a null width that restricts a directivity sensitivity of the microphone array, and extracting a signal corresponding to the target sound from the sound signals by using the generated signal.
Abstract: A sound zoom method, medium, and apparatus generating a signal in which a target sound is removed from sound signals input to a microphone-array by adjusting a null width that restricts a directivity sensitivity of the microphone array, and extracting a signal corresponding to the target sound from the sound signals by using the generated signal. Thus, a sound located at a predetermined position away from the microphone array can be selectively obtained so that a target sound is efficiently obtained.

75 citations


Journal ArticleDOI
TL;DR: A comparison of fish localization behavior with directional cues available in the form of local particle motion vectors is reported on.
Abstract: Sound source localization of the midshipman fish (Porichthys notatus) was studied using the phonotactic response of gravid females to synthetic advertisement calls. Playback experiments were conducted in a 12‐ft‐diameter outdoor concrete tank at the Bodega Marine Laboratory using a J‐9 transducer placed at the center of the tank. The sound field in the tank was measured at 5‐cm intervals using an eight‐hydrophone array to measure the pressure gradients from which particle motion vectors were calculated. The acoustic measurements confirmed that the J‐9 projector was operating as a monopole source. Animals were released 90 cm away from the sound source, and 60 positive phonotactic responses from naive gravid females were video taped and analyzed. The phonotactic responses consisted primarily of straight to somewhat curved tracks to the sound source. Abrupt changes in trajectory to the sound source were rarely observed. The results confirm that fish can locate sound sources in the near field.

68 citations


Proceedings ArticleDOI
06 May 2008
TL;DR: In this paper, the authors compare the performance of different sound source localization techniques in real-time implementation and compare them to a sub-optimal LS search method using adaptive eigenvalue decomposition.
Abstract: Comparing the different sound source localization techniques, proposed in the literature during the last decade, represents a relevant topic in order to establish advantages and disadvantages of a given approach in a real-time implementation. Traditionally, algorithms for sound source localization rely on an estimation of time difference of arrival (TDOA) at microphone pairs through GCC-PHAT When several microphone pairs are available the source position can be estimated as the point in space that best fits the set of TDOA measurements by applying global coherence field (GCF), also known as SRP-PHAT, or oriented global coherence field (OGCF). A first interesting analysis compares the performance of GCF and OGCF to a suboptimal LS search method. In a second step, Adaptive Eigenvalue Decomposition is implemented as an alternative to GCC-PHAT in TDOA estimation. Comparative experiments are conducted on signals acquired by a linear array during WOZ experiments in an interactive-TV scenario. Changes in performance according to different SNR levels are reported.

63 citations


Journal ArticleDOI
TL;DR: In this article, the sound pressure distribution of a standing-wave field was measured using a small microphone and calculated numerically using Rayleigh's formula by adding multiply reflected waves, which is a noncontact manipulation technique in micromachine technology.
Abstract: A noncontact manipulation technique is necessary in micromachine technology. Using a standing-wave field generated between a transducer and a reflector, it is possible to trap particles at nodes of a sound pressure field. In the present paper, a sound field has been studied by both experimental measurement and numerical calculation. The sound pressure distribution of a standing-wave field was measured using a small microphone and calculated numerically using Rayleigh's formula. Although Rayleigh's formula is usually used to calculate direct sound pressure from a sound source, it has been shown that the sound pressure of the standing-wave field can be calculated by Rayleigh's formula by adding multiply reflected waves.

61 citations


Proceedings ArticleDOI
12 May 2008
TL;DR: Novel methods for detecting and localizing multiple wideband acoustic sources using spherical apertures using frequency-independent multiple-source localization and detection methods are discussed.
Abstract: This paper discusses novel methods for detecting and localizing multiple wideband acoustic sources using spherical apertures. In contrast to traditional methods the techniques presented here are not based on processing the output of individual microphones directly. Instead, the microphone signals are used to decompose the wave- field into its spherical harmonics which are subsequently used as a basis for novel frequency-independent multiple-source localization and detection methods.

54 citations


Journal ArticleDOI
TL;DR: In this article, a one-dimensional acoustic source location theory is developed to incorporate modal acoustic emission, and the arrival times and wave velocities needed for source location are influenced by mode and frequency.
Abstract: Traditional acoustic source location techniques are greatly affected by the threshold value of the acoustic signals. Based on the theory of modal acoustic emission, acoustic signals have characteristics indicative of multi-mode, broad band and dispersion. One-dimensional acoustic source location theory is developed to incorporate these characteristics. The arrival times and wave velocities needed for source location are influenced by mode and frequency. Based on the number of sensors used, two universal source location techniques have been investigated. To obtain the arrival time of one mode at certain frequency, the Gabor wavelet transform is applied to the analysis of acoustic signals. One-dimensional acoustic source location experiments in a thin plate have been performed using a lead break as acoustic source. The acoustic source location techniques have been greatly improved and the accuracy of these acoustic source location methods has been verified. Copyright © 2007 John Wiley & Sons, Ltd.

53 citations


Patent
03 Jul 2008
TL;DR: In this article, a beam formation calculating section is used to calculate the phase of a wavefront from a virtual point sound source, and a phase controlling section is employed to control the phases of the wavefront.
Abstract: A speaker array apparatus includes a speaker array that emits sounds of a plurality of channels, a beam formation calculating section that performs a calculation for controlling phases of the sounds so that the speaker array emits sound beams in directions set for the respective channels, a sound source localization applying section that performs a calculation for controlling the phases of the sounds emitted from the speaker array so as to form a plurality of virtual point sound sources, and performs a calculation of auditory sensation characteristics at a listening position on a basis of a head-related transfer function, a selecting section that selects one of the beam formation calculating section and the sound source localization applying section, and a phase controlling section that controls the phases of the sounds emitted from the speaker array on a basis of a calculation result of the beam formation calculation section which is selected by the selecting section or applies the auditory sensation characteristics and controls the phase of a wavefront from the virtual point sound source on a basis of a calculation result of the beam formation calculating section which is selected by the selecting section.

PatentDOI
TL;DR: In this article, a method for canceling background noise of a sound source other than a target direction sound source in order to realize highly accurate speech recognition, and a system using the same.
Abstract: Provided is a method for canceling background noise of a sound source other than a target direction sound source in order to realize highly accurate speech recognition, and a system using the same. In terms of directional characteristics of a microphone array, due to a capability of approximating a power distribution of each angle of each of possible various sound source directions by use of a sum of coefficient multiples of a base form angle power distribution of a target sound source measured beforehand by base form angle by using a base form sound, and power distribution of a non-directional background sound by base form, only a component of the target sound source direction is extracted at a noise suppression part. In addition, when the target sound source direction is unknown, at a sound source localization part, a distribution for minimizing the approximate residual is selected from base form angle power distributions of various sound source directions to assume a target sound source direction. Further, maximum likelihood estimation is executed by using voice data of the component of the sound source direction passed through these processes, and a voice model obtained by predetermined modeling of the voice data, and speech recognition is carried out based on an obtained assumption value.

Book ChapterDOI
10 Jan 2008
TL;DR: In this article, Signal Model Localization Approach Taxonomy Indirect Localisation Approaches Direct Localization Approaches Evaluation of Localization Algorithms Conclusions Bibliography contains sections titled: Introduction Signal model localization approach Taxonomy
Abstract: This chapter contains sections titled: Introduction Signal Model Localization Approach Taxonomy Indirect Localization Approaches Direct Localization Approaches Evaluation of Localization Algorithms Conclusions Bibliography

Patent
Ross Cutler1
27 Jun 2008
TL;DR: In this article, sound origination detection through use of infrared detection of satellite microphones, estimation of distance between satellite microphones and base unit utilizing captured audio, and estimation of satellite microphone orientation using captured audio are combined to enhance sound source localization and active speaker detection accuracy.
Abstract: Speakers are identified based on sound origination detection through use of infrared detection of satellite microphones, estimation of distance between satellite microphones and base unit utilizing captured audio, and/or estimation of satellite microphone orientation utilizing captured audio. Multiple sound source localization results are combined to enhance sound source localization and/or active speaker detection accuracy.

Patent
10 Sep 2008
TL;DR: A sound source direction detector comprises FFT analysis sections (103(1) to 103(3) for generating a frequency spectrum in at least one frequency band of acoustic signals for each of the acoustic signals collected by two or more microphones arranged apart from one another, detection sound identifying sections (104(1, to 104(3)) for identifying a time portion of the frequency spectrum of a detection sound, and direction detecting sections (105) for obtaining the difference between the times at which the detection sound reaches the microphones, and outputting it depending on the degree of coincidence between the
Abstract: A sound source direction detector comprises FFT analysis sections (103(1) to 103(3)) for generating a frequency spectrum in at least one frequency band of acoustic signals for each of the acoustic signals collected by two or more microphones arranged apart from one another, detection sound identifying sections (104(1) to 104(3)) for identifying a time portion of the frequency spectrum of a detection sound which obtains a sound source direction from the frequency spectrum in the frequency band, and a direction detecting section (105) for obtaining the difference between the times at which the detection sound reaches the microphones, obtaining the sound source direction from the time difference, the distance between the microphones, and the sound velocity, and outputting it depending on the degree of coincidence between the microphones of the frequency spectrum in the time portion identified by the detection sound identifying sections (104(1) to 104(3)) in a time interval which is the time unit to detect the sound source direction.

Journal ArticleDOI
TL;DR: In this article, the authors present a generalized expression of sound speed variation in terms of a traveltime residual normalized to the vertical component, where the residual traveltimes to any seafloor transponders will have the same value regardless of their depths and slant angles.
Abstract: The GPS/acoustic technique applied to seafloor geodesy intrinsically measures integrated sound speed along a trajectory of an acoustic signal as well as the position of a seafloor transponder array. We present here a generalized expression of sound speed variation in terms of a traveltime residual normalized to the vertical component. With this expression, residual traveltimes to any seafloor transponders will have a same value regardless of their depths and slant angles. This is valid even for the case having horizontal gradient in sound speed structure; the gradient affects only on positioning of a transponder array and not on the estimate of sound speed just beneath the observation point. We monitored temporal variation of this quantity through a GPS/acoustic survey and compared it with in situ expendable bathythermograph (XBT) measurements periodically carried out during the survey. We found that the relative change of the two independent measurements are in good agreement within 5% of the typical amplitude of temporal variation.

Journal ArticleDOI
TL;DR: To eliminate the limitation of the conventional acoustic radiation torque theory, a new theory is established to calculate the radiation torque on any irregularly shaped scatterer in any arbitrary sound field, and it is found that for a semicircular cylinder scattrer in a standing-wave sound field its rotation velocity is normally nonzero and the Radiation torque changes with the spatial attitude.
Abstract: To eliminate the limitation of the conventional acoustic radiation torque theory, which is only applicable to a disklike scatterer in a plane sound field, a new theory is established to calculate the radiation torque on any irregularly shaped scatterer in any arbitrary sound field. First, with the aid of the conservation law of angular momentum, the acoustic radiation torque is expressed as the angular momentum flux through a spherical surface with the center at the scatterer’s centroid. Second, the velocity potential of the scattered field is derived, taking into account the influences of the translational and rotational movements of the scatterer induced by the first order stress of the incident sound field. Finally, a general calculating formula of the acoustic radiation torque is achieved. For a disklike scatterer in a plane sound filed, results from the above formula are well identical with those conventional formulas. By studying the case of a semicircular cylinder scatterer in a standing-wave sound field, it is found that for an irregularly shaped scatterer its rotation velocity is normally nonzero and the radiation torque changes with the spatial attitude.

Journal ArticleDOI
TL;DR: This paper introduces a new method for the estimation of sound source distance and direction using at least three microphone sensors in indoor environments that exploits the existed geometrical relationships of the sensors to form an exact solution to estimating the source position.
Abstract: This paper introduces a new method for the estimation of sound source distance and direction using at least three microphone sensors in indoor environments. Unlike the other methods that normally use approximations in obtaining the time difference between sensors, this method exploits the existed geometrical relationships of the sensors to form an exact solution to estimating the source position. To overcome reverberation, an enhancing pre-process has been used for different sound sources with different spectra, e.g., single frequency, multiple frequencies and different noise shapes. Source direction and distances are estimated from time of sound wave travel and distances of acoustic sensors. Using the method described in this paper a level of 1° accuracy is obtained. Several experimental tests have been undertaken that verify the results. Conclusions and future work are also described.

Journal ArticleDOI
TL;DR: The presented method showed a good performance in comparison with the conventional methods such as time difference of arrival (TDOA) and generalized cross correlation- phase transform (GCC-PHAT) in distant-varying, noise and reverberant environments.
Abstract: This paper is concerned with multiple microphone-based sound source localization in home robot environments. For this purpose, we use the excitation source information to determine the time-delay between each two microphones from speech source when robot's name is called. Furthermore, we present a novel method to estimate the reliable localization angle from the obtained time-delay values. Thus, it can be used as a core technique in conjunction with human-robot interaction that can naturally interact between human and robots in home robot applications. The experimental results on sound localization database revealed that the presented method showed a good performance in comparison with the conventional methods such as time difference of arrival (TDOA) and generalized cross correlation- phase transform (GCC-PHAT) in distant-varying, noise and reverberant environments.

Book ChapterDOI
01 Jan 2008
TL;DR: Fay and Popper as mentioned in this paper used a phase model to detect the direction from which a sound emanates in a minnows (Phoxinus laevis) by using the amplitude, time, or phase difference between the ears.
Abstract: If fish are to behave appropriately with respect to objects and events in their environment they must process an acoustic scene that is often complex (Fay and Popper 2000). A presumptively important part of such behavior is the ability to determine properly the direction from which a sound emanates. Although the question regarding mechanisms for sound-source localization in fishes has been of interest since Karl von Frisch and Sven Dijkgraaf (1935) performed behavioral studies in European minnows (Phoxinus laevis), the mechanisms remain poorly understood, with relatively few biologically plausible models. A localization mechanism that exploits the amplitude, time, or phase difference between the ears as employed by terrestrial vertebrates is not available to fish because the ears are very close together, the speed of sound in water is more than three times faster than in air, and the close impedance match between the fish’s body and water precludes usable diffracted paths (van Bergeijk 1964, 1967; and see Sand and Bleckmann, Chapter 6). Another major difficulty that any model must address is a resolution of the so called “180 ambiguity” that arises because the axis of particle motion associated with a passing sound points both toward and away from the sound source (for review of sound localization by fish see Fay 2005; Sand and Bleckmann, Chapter 6). Current models of directional hearing in fish with mechanisms to resolve the 180 ambiguity include the “phase model” proposed by Schuijf and colleagues (e.g., Chapman and Hawkins 1973; Schuijf 1975; Schuijf and Buwalda 1975) that compares the phase of the pressure and particle motion components of sound or the phase of the direct-path particle motion and the particle motion of sound reflected from surfaces or objects; an “orbital” model by Schellart and de Munck (1987; de Munck and Schellart 1987) in which sound pressure and particle motion together cause the otolith orbits to rotate either clockwise or counterclockwise depending on whether the source is to the left or right; a computational model by Rogers et al. (1988) that also uses both pressure and particle motion; and, a more algorithmic approach pointed out by Kalmijn (1997) by which a fish could make its way to a sound source by

Patent
26 Jan 2008
TL;DR: In this article, a multi-sensor sound source localization (SSL) technique is presented which provides a true maximum likelihood (ML) treatment for microphone arrays having more than one pair of audio sensors.
Abstract: A multi-sensor sound source localization (SSL) technique is presented which provides a true maximum likelihood (ML) treatment for microphone arrays having more than one pair of audio sensors. Generally, this is accomplished by selecting a sound source location that results in a time of propagation from the sound source to the audio sensors of the array, which maximizes a likelihood of simultaneously producing audio sensor output signals inputted from all the sensors in the array. The likelihood includes a unique term that estimates an unknown audio sensor response to the source signal for each of the sensors in the array.

Journal ArticleDOI
TL;DR: Results from simulated dialogues in various conditions favor TDE combination using intersection-based methods over union, and the real-data dialogue results agree with the simulations, showing a 45% RMSE reduction when choosing the intersection over union of TDE functions.
Abstract: The behavior of time delay estimation (TDE) is well understood and therefore attractive to apply in acoustic source localization (ASL). A time delay between microphones maps into a hyperbola. Furthermore, the likelihoods for different time delays are mapped into a set of weighted nonoverlapping hyperbolae in the spatial domain. Combining TDE functions from several microphone pairs results in a spatial likelihood function (SLF) which is a combination of sets of weighted hyperbolae. Traditionally, the maximum SLF point is considered as the source location but is corrupted by reverberation and noise. Particle filters utilize past source information to improve localization performance in such environments. However, uncertainty exists on how to combine the TDE functions. Results from simulated dialogues in various conditions favor TDE combination using intersection-based methods over union. The real-data dialogue results agree with the simulations, showing a 45% RMSE reduction when choosing the intersection over union of TDE functions.

Patent
14 Nov 2008
TL;DR: In this paper, a virtual sound source localization system is proposed, where a receiver operates an operating section to localize a Cch sound source at an approximately center of the loudspeakers, thereby adjusting a sound balance of the speakers.
Abstract: In a virtual sound source localization apparatus, a distance between two loudspeakers and a shortest distance between a line connecting the loudspeakers and a listening position are set beforehand, and a listener operates an operating section to localize a Cch sound source at an approximately center of the loudspeakers, thereby adjusting a sound balance of the loudspeakers. In addition, a controller calculates a difference in distance from the loudspeakers to the listening position, sets a delay amount of delay correctors such that sound emitted from the loudspeakers substantially reaches the listening position simultaneously, and adjusts sound output timing of the loudspeakers. In this way, even though the listening position is changed, the listener can operate the operating section to optimize a virtual surround effect.

Proceedings ArticleDOI
19 May 2008
TL;DR: A robot referee for "rock- paper-scissors (RPS)" sound games; the robot decides the winner from a combination of rock, paper and scissors uttered by two or three people simultaneously without using any visual information is described.
Abstract: This paper describes a robot referee for "rock- paper-scissors (RPS)" sound games; the robot decides the winner from a combination of rock, paper and scissors uttered by two or three people simultaneously without using any visual information. In this referee task, the robot has to cope with speech with low signal-to-noise ratio (SNR) due to a mixture of speeches, robot motor noises, and ambient noises. Our robot referee system, thus, consists of two subsystems - a real-time robot audition subsystem and a dialog subsystem focusing on RPS sound games. The robot audition subsystem can recognize simultaneous speeches by exploiting two key ideas; preprocessing consisting of sound source localization and separation with a microphone array, and system integration based on missing feature theory (MFT). Preprocessing improves the SNR of a target sound signal using geometric source separation with a multi-channel post-filter. MFT uses only reliable acoustic features in speech recognition and masks out unreliable parts caused by interfering sounds and preprocessing. MFT thus provides smooth integration between preprocessing and automatic speech recognition. The dialog subsystem is implemented as a system-initiative dialog system for multiple players based on deterministic finite automata. It first waits for a trigger command to start an RPS sound game, controls the dialog with players in the game, and finally decides the winner of the game. The referee system is constructed for Honda ASIMO with an 8-ch microphone array. In the case with two players, we attained a 70% task completion rate for the games on average.

Patent
02 Sep 2008
TL;DR: In this article, sound correction processing is performed for a sound signal obtained from a sound collecting portion, after determining what is performed as the sound correction process based on an image signal paired with sound signal.
Abstract: Sound correction processing is performed for a sound signal obtained from a sound collecting portion. In particular, sound correction processing is performed after determining what is performed as the sound correction processing based on an image signal paired with a sound signal, the image signal obtained from an imaging portion, the sound signal, control data of the imaging portion, and the like.

Patent
01 Apr 2008
TL;DR: In this paper, a method, medium, and apparatus was proposed to cancel noise from a mixed sound by extracting at least one feature vector indicating an attribute difference between the sound source signals from the noise.
Abstract: A method, medium, and apparatus canceling noise from a mixed sound. The method includes receiving sound source signals including a target sound and noise, extracting at least one feature vector indicating an attribute difference between the sound source signals from the sound source signals, calculating a suppression coefficient considering ratios of noise to the sound source signals based on the at least one extracted feature vector, and canceling the sound source signals corresponding to noise by controlling an intensity of an output signal generated from the sound source signals according to the calculated suppression coefficient. Accordingly, a clear target sound source signal can be obtained.

Patent
22 Sep 2008
TL;DR: In this paper, a sound source direction detection system is presented, which is based on the phase at which the highest of the power levels added by the power level addition section is detected.
Abstract: Disclosed herein is a sound source direction detecting apparatus including: a plurality of microphones configured to collect sounds from a sound source in order to form an audio frame; a frequency decomposition section configured to decompose the audio frame into frequency components; an error range determination section configured to determine the effects of noises collected together with the sounds as an error range relative to phases; a power level dispersion section configured to disperse power levels of the sounds for each of the frequency components decomposed by the frequency decomposition section, on the basis of the error range determined by the error range determination section; a power level addition section configured to add the power levels dispersed by the power level dispersion section; and a sound source direction detection section configured to detect the direction of the sound source based on the phase at which is located the highest of the power levels added by the power level addition section.

Patent
Ross Cutler1
27 Jun 2008
TL;DR: In this article, a region of interest in video image capture for communication purposes is selected based on one or more inputs based on sound source localization, multi-person detection, and active speaker detection using audio and visual cues.
Abstract: Regions of interest in video image capture for communication purposes are selected based on one or more inputs based on sound source localization, multi-person detection, and active speaker detection using audio and/or visual cues. Exposure and/or gain for the selected region are automatically enhanced for improved video quality focusing on people or inanimate objects of interest.

Journal ArticleDOI
Tim Mellow1
TL;DR: Radiation characteristics are calculated for a circular planar sound source in free space with a uniform surface pressure distribution, which can be regarded as a freely suspended membrane with zero mass and stiffness.
Abstract: Radiation characteristics are calculated for a circular planar sound source in free space with a uniform surface pressure distribution, which can be regarded as a freely suspended membrane with zero mass and stiffness. This idealized dipole source is shown to have closed form solutions for its far-field pressure response and radiation admittance. The latter is found to have a simple mathematical relationship with the radiation impedance of a rigid piston in an infinite baffle. Also, a single expansion is derived for the near-field pressure field, which degenerates to a closed form solution on the axis of symmetry. From the normal gradient of the surface pressure, the surface velocity is calculated. The near-field expression is then generalized to an arbitrary surface pressure distribution. It is shown how this can be used as a simplified solution for a rigid disk in free space or a more realistic sound source such as pre-tensioned membrane in free space with non-zero mass and a clamped rim.

Proceedings ArticleDOI
15 Aug 2008
TL;DR: A speaker localization system with six microphones for a humanoid robot called MAHRU of KIST is reported on and a time delay of arrival (TDOA)-based feature matrix with its algorithm based on the minimum sum of absolute errors (MSAE) for sound source localization is proposed.
Abstract: Research on human-robot interaction has recently been getting increasing attention. In the research field of human-robot interaction, speech signal processing in particular is the source of much interest. In this paper, we report on a speaker localization system with six microphones for a humanoid robot called MAHRU of KIST and propose a time delay of arrival (TDOA)-based feature matrix with its algorithm based on the minimum sum of absolute errors (MSAE) for sound source localization. The TDOA-based feature matrix is defined as a simple database matrix calculated from pairs of microphones installed on a humanoid robot. To verify the solid performance of our speaker localization system for a humanoid robot, we present the various experimental results for the speech sources at all directions within 5 m distance and the height divided into three parts.