Showing papers on "Acoustic source localization published in 2014"

A geometric approach to sound source localization from time-delay estimates

Abstract: This paper addresses the problem of sound-source localization from time-delay estimates using arbitrarily-shaped non-coplanar microphone arrays. A novel geometric formulation is proposed, together with a thorough algebraic analysis and a global optimization solver. The proposed model is thoroughly described and evaluated. The geometric analysis, stemming from the direct acoustic propagation model, leads to necessary and sufficient conditions for a set of time delays to correspond to a unique position in the source space. Such sets of time delays are referred to as feasible sets. We formally prove that every feasible set corresponds to exactly one position in the source space, whose value can be recovered using a closed-form localization mapping. Therefore we seek for the optimal feasible set of time delays given, as input, the received microphone signals. This time delay estimation problem is naturally cast into a programming task, constrained by the feasibility conditions derived from the geometric analysis. A global branch-and-bound optimization technique is proposed to solve the problem at hand, hence estimating the best set of feasible time delays and, subsequently, localizing the sound source. Extensive experiments with both simulated and real data are reported; we compare our methodology to four state-of-the-art techniques. This comparison shows that the proposed method combined with the branch-and-bound algorithm outperforms existing methods. These in-depth geometric understanding, practical algorithms, and encouraging results, open several opportunities for future work.

SoundCompass: A Distributed MEMS Microphone Array-Based Sensor for Sound Source Localization

Abstract: Sound source localization is a well-researched subject with applications ranging from localizing sniper fire in urban battlefields to cataloging wildlife in rural areas. One critical application is the localization of noise pollution sources in urban environments, due to an increasing body of evidence linking noise pollution to adverse effects on human health. Current noise mapping techniques often fail to accurately identify noise pollution sources, because they rely on the interpolation of a limited number of scattered sound sensors. Aiming to produce accurate noise pollution maps, we developed the SoundCompass, a low-cost sound sensor capable of measuring local noise levels and sound field directionality. Our first prototype is composed of a sensor array of 52 Microelectromechanical systems (MEMS) microphones, an inertial measuring unit and a low-power field-programmable gate array (FPGA). This article presents the SoundCompass's hardware and firmware design together with a data fusion technique that exploits the sensing capabilities of the SoundCompass in a wireless sensor network to localize noise pollution sources. Live tests produced a sound source localization accuracy of a few centimeters in a 25-m2 anechoic chamber, while simulation results accurately located up to five broadband sound sources in a 10,000-m2 open field.

Acoustic contrast, planarity and robustness of sound zone methods using a circular loudspeaker array.

Abstract: Since the mid 1990s, acoustics research has been undertaken relating to the sound zone problem—using loudspeakers to deliver a region of high sound pressure while simultaneously creating an area where the sound is suppressed—in order to facilitate independent listening within the same acoustic enclosure. The published solutions to the sound zone problem are derived from areas such as wave field synthesis and beamforming. However, the properties of such methods differ and performance tends to be compared against similar approaches. In this study, the suitability of energy focusing, energy cancelation, and synthesis approaches for sound zone reproduction is investigated. Anechoic simulations based on two zones surrounded by a circular array show each of the methods to have a characteristic performance, quantified in terms of acoustic contrast, array control effort and target sound field planarity. Regularization is shown to have a significant effect on the array effort and achieved acoustic contrast, particularly when mismatched conditions are considered between calculation of the source weights and their application to the system.

Modeling sound-source localization in sagittal planes for human listeners

Abstract: Monaural spectral features are important for human sound-source localization in sagittal planes, including front-back discrimination and elevation perception. These directional features result from the acoustic filtering of incoming sounds by the listener's morphology and are described by listener-specific head-related transfer functions (HRTFs). This article proposes a probabilistic, functional model of sagittal-plane localization that is based on human listeners' HRTFs. The model approximates spectral auditory processing, accounts for acoustic and non-acoustic listener specificity, allows for predictions beyond the median plane, and directly predicts psychoacoustic measures of localization performance. The predictive power of the listener-specific modeling approach was verified under various experimental conditions: The model predicted effects on localization performance of band limitation, spectral warping, non-individualized HRTFs, spectral resolution, spectral ripples, and high-frequency attenuation in speech. The functionalities of vital model components were evaluated and discussed in detail. Positive spectral gradient extraction, sensorimotor mapping, and binaural weighting of monaural spatial information were addressed in particular. Potential applications of the model include predictions of psychophysical effects, for instance, in the context of virtual acoustics or hearing assistive devices.

Scaling and dimensional analysis of acoustic streaming jets

Abstract: This paper focuses on acoustic streaming free jets. This is to say that progressive acoustic waves are used to generate a steady flow far from any wall. The derivation of the governing equations under the form of a nonlinear hydrodynamics problem coupled with an acoustic propagation problem is made on the basis of a time scale discrimination approach. This approach is preferred to the usually invoked amplitude perturbations expansion since it is consistent with experimental observations of acoustic streaming flows featuring hydrodynamic nonlinearities and turbulence. Experimental results obtained with a plane transducer in water are also presented together with a review of the former experimental investigations using similar configurations. A comparison of the shape of the acoustic field with the shape of the velocity field shows that diffraction is a key ingredient in the problem though it is rarely accounted for in the literature. A scaling analysis is made and leads to two scaling laws for the typical velocity level in acoustic streaming free jets; these are both observed in our setup and in former studies by other teams. We also perform a dimensional analysis of this problem: a set of seven dimensionless groups is required to describe a typical acoustic experiment. We find that a full similarity is usually not possible between two acoustic streaming experiments featuring different fluids. We then choose to relax the similarity with respect to sound attenuation and to focus on the case of a scaled water experiment representing an acoustic streaming application in liquid metals, in particular, in liquid silicon and in liquid sodium. We show that small acoustic powers can yield relatively high Reynolds numbers and velocity levels; this could be a virtue for heat and mass transfer applications, but a drawback for ultrasonic velocimetry.

Spatially resolved acoustic spectroscopy for rapid imaging of material microstructure and grain orientation

Abstract: Measuring the grain structure of aerospace materials is very important to understand their mechanical properties and in-service performance. Spatially resolved acoustic spectroscopy is an acoustic technique utilizing surface acoustic waves to map the grain structure of a material. When combined with measurements in multiple acoustic propagation directions, the grain orientation can be obtained by fitting the velocity surface to a model. The new instrument presented here can take thousands of acoustic velocity measurements per second. The spatial and velocity resolution can be adjusted by simple modification to the system; this is discussed in detail by comparison of theoretical expectations with experimental data.

Sound source localization inspired by the ears of the Ormia ochracea

Abstract: The parasitoid fly Ormia ochracea has the remarkable ability to locate crickets using audible sound. This ability is, in fact, remarkable as the fly's hearing mechanism spans only 1.5 mm which is 50× smaller than the wavelength of sound emitted by the cricket. The hearing mechanism is, for all practical purposes, a point in space with no significant interaural time or level differences to draw from. It has been discovered that evolution has empowered the fly with a hearing mechanism that utilizes multiple vibration modes to amplify interaural time and level differences. Here, we present a fully integrated, man-made mimic of the Ormia's hearing mechanism capable of replicating the remarkable sound localization ability of the special fly. A silicon-micromachined prototype is presented which uses multiple piezoelectric sensing ports to simultaneously transduce two orthogonal vibration modes of the sensing structure, thereby enabling simultaneous measurement of sound pressure and pressure gradient.

Effects of small variations of speed of sound in optoacoustic tomographic imaging

Abstract: Speed of sound difference in the imaged object and surrounding coupling medium may reduce the resolution and overall quality of optoacoustic tomographic reconstructions obtained by assuming a uniform acoustic medium. In this work, the authors investigate the effects of acoustic heterogeneities and discuss potential benefits of accounting for those during the reconstruction procedure.The time shift of optoacoustic signals in an acoustically heterogeneous medium is studied theoretically by comparing different continuous and discrete wave propagation models. A modification of filtered back-projection reconstruction is subsequently implemented by considering a straight acoustic rays model for ultrasound propagation. The results obtained with this reconstruction procedure are compared numerically and experimentally to those obtained assuming a heuristically fitted uniform speed of sound in both full-view and limited-view optoacoustic tomography scenarios.The theoretical analysis showcases that the errors in the time-of-flight of the signals predicted by considering the straight acoustic rays model tend to be generally small. When using this model for reconstructing simulated data, the resulting images accurately represent the theoretical ones. On the other hand, significant deviations in the location of the absorbing structures are found when using a uniform speed of sound assumption. The experimental results obtained with tissue-mimicking phantoms and a mouse postmortem are found to be consistent with the numerical simulations.Accurate analysis of effects of small speed of sound variations demonstrates that accounting for differences in the speed of sound allows improving optoacoustic reconstruction results in realistic imaging scenarios involving acoustic heterogeneities in tissues and surrounding media.

A Steered-Response Power Algorithm Employing Hierarchical Search for Acoustic Source Localization Using Microphone Arrays

Abstract: The localization of a speaker inside a closed environment is often approached by real-time processing of multiple audio signals captured by a set of microphones. One of the leading related methods for sound source localization, the steered-response power (SRP), searches for the point of maximum power over a spatial grid. High-accuracy localization calls for a dense grid and/or many microphones, which tends to impractically increase computational requirements. This paper proposes a new method for sound source localization (called H-SRP), which applies the SRP approach to space regions instead of grid points. This arrangement makes room for the use of a hierarchical search inspired by the branch-and-bound paradigm, which is guaranteed to find the global maximum in anechoic environments and shown experimentally to also work under reverberant conditions. Besides benefiting from the improved robustness of volume-wise search over point-wise search as to reverberation effects, the H-SRP attains high performance with manageable complexity. In particular, an experiment using a 16-microphone array in a typical presentation room yielded localization errors of the order of 7 cm, and for a given fixed complexity, competing methods' errors are two to three times larger.

Underwater acoustic omnidirectional absorber

Abstract: Gradient index media, which are designed by varying local element properties in given geometry, have been utilized to manipulate acoustic waves for a variety of devices. This study presents a cylindrical, two-dimensional acoustic “black hole” design that functions as an omnidirectional absorber for underwater applications. The design features a metamaterial shell that focuses acoustic energy into the shell's core. Multiple scattering theory was used to design layers of rubber cylinders with varying filling fractions to produce a linearly graded sound speed profile through the structure. Measured pressure intensity agreed with predicted results over a range of frequencies within the homogenization limit.

A comparative study of Interaural Time Delay estimation methods.

Abstract: The Interaural Time Delay (ITD) is an important binaural cue for sound source localization. Calculations of ITD values are obtained either from measured time domain Head-Related Impulse Responses (HRIRs) or from their frequency transform Head-Related Transfer Functions (HRTFs). Numerous methods exist in current literature, based on a variety of definitions and assumptions of the nature of the ITD as an acoustic cue. This work presents a thorough comparative study of the degree of variability between some of the most common methods for calculating the ITD from measured data. Thirty-two different calculations or variations are compared for positions on the horizontal plane for the HRTF measured on both a KEMAR mannequin and a rigid sphere. Specifically, the spatial variations of the methods are investigated. Included is a discussion of the primary potential causes of these differences, such as the existence of multiple peaks in the HRIR of the contra-lateral ear for azimuths near the inter-aural axis due to multipath propagation and head/pinnae shadowing.

Bayesian Nonparametrics for Microphone Array Processing

Abstract: Sound source localization and separation from a mixture of sounds are essential functions for computational auditory scene analysis. The main challenges are designing a unified framework for joint optimization and estimating the sound sources under auditory uncertainties such as reverberation or unknown number of sounds. Since sound source localization and separation are mutually dependent, their simultaneous estimation is required for better and more robust performance. A unified model is presented for sound source localization and separation based on Bayesian nonparametrics. Experiments using simulated and recorded audio mixtures show that a method based on this model achieves state-of-the-art sound source separation quality and has more robust performance on the source number estimation under reverberant environments.

Experiments in acoustic source localization using sparse arrays in adverse indoors environments

Abstract: In this paper we experiment with 2-D source localization in smart homes under adverse conditions using sparse distributed microphone arrays. We propose some improvements to deal with problems due to high reverberation, noise and use of a limited number of microphones. These consist of a pre-filtering stage for dereverberation and an iterative procedure that aims to increase accuracy. Experiments carried out in relatively large databases with both simulated and real recordings of sources in various positions indicate that the proposed method exhibits a better performance compared to others under challenging conditions while also being computationally efficient. It is demonstrated that although reverberation degrades localization performance, this degradation can be compensated by identifying the reliable microphone pairs and disposing of the outliers.

A discriminative learning approach to probabilistic acoustic source localization

Abstract: Sound source localization algorithms commonly include assessment of inter-sensor (generalized) correlation functions to obtain direction-of-arrival estimates. Here, we present a classification-based method for source localization that uses discriminative support vector machine-learning of correlation patterns that are indicative of source presence or absence. Subsequent probabilistic modeling generates a map of sound source presence probability in given directions. Being data-driven, the method during training adapts to characteristics of the sensor setup, such as convolution effects in non-free-field situations, and to target signal specific acoustic properties. Experimental evaluation was conducted with algorithm training in anechoic single-talker scenarios and test data from several reverberant multi-talker situations, together with diffuse and real-recorded background noise, respectively. Results demonstrate that the method successfully generalizes from training to test conditions. Improvement over the best of five investigated state-of-the-art angular spectrum-based reference methods was on average about 45% in terms of relative F-measure-related error reduction.

Source and Listener Directivity for Interactive Wave-Based Sound Propagation

Abstract: We present an approach to model dynamic, data-driven source and listener directivity for interactive wave-based sound propagation in virtual environments and computer games. Our directional source representation is expressed as a linear combination of elementary spherical harmonic (SH) sources. In the preprocessing stage, we precompute and encode the propagated sound fields due to each SH source. At runtime, we perform the SH decomposition of the varying source directivity interactively and compute the total sound field at the listener position as a weighted sum of precomputed SH sound fields. We propose a novel plane-wave decomposition approach based on higher-order derivatives of the sound field that enables dynamic HRTF-based listener directivity at runtime. We provide a generic framework to incorporate our source and listener directivity in any offline or online frequency-domain wave-based sound propagation algorithm. We have integrated our sound propagation system in Valve's Source game engine and use it to demonstrate realistic acoustic effects such as sound amplification, diffraction low-passing, scattering, localization, externalization, and spatial sound, generated by wave-based propagation of directional sources and listener in complex scenarios. We also present results from our preliminary user study.

Sound source localization identification accuracy: Bandwidth dependencies

Abstract: Sound source localization accuracy using a sound source identification task was measured in the front, right quarter of the azimuth plane as rms (root-mean-square) error (degrees) for stimulus conditions in which the bandwidth (1/20 to 2 octaves wide) and center frequency (250, 2000, 4000 Hz) of 200-ms noise bursts were varied. Tones of different frequencies (250, 2000, 4000 Hz) were also used. As stimulus bandwidth increases, there is an increase in sound source localization identification accuracy (i.e., rms error decreases). Wideband stimuli (>1 octave wide) produce best sound source localization accuracy (∼6°–7° rms error), and localization accuracy for these wideband noise stimuli does not depend on center frequency. For narrow bandwidths (<1 octave) and tonal stimuli, accuracy does depend on center frequency such that highest accuracy is obtained for low-frequency stimuli (centered on 250 Hz), worse accuracy for mid-frequency stimuli (centered on 2000 Hz), and intermediate accuracy for high-frequency stimuli (centered on 4000 Hz).

Simultaneous-Speaker Voice Activity Detection and Localization Using Mid-Fusion of SVM and HMMs

Abstract: Humans can extract speech signals that they need to understand from a mixture of background noise, interfering sound sources, and reverberation for effective communication. Voice Activity Detection (VAD) and Sound Source Localization (SSL) are the key signal processing components that humans perform by processing sound signals received at both ears, sometimes with the help of visual cues by locating and observing the lip movements of the speaker. Both VAD and SSL serve as the crucial design elements for building applications involving human speech. For example, systems with microphone arrays can benefit from these for robust speech capture in video conferencing applications, or for speaker identification and speech recognition in Human Computer Interfaces (HCIs). The design and implementation of robust VAD and SSL algorithms in practical acoustic environments are still challenging problems, particularly when multiple simultaneous speakers exist in the same audiovisual scene. In this work we propose a multimodal approach that uses Support Vector Machines (SVMs) and Hidden Markov Models (HMMs) for assessing the video and audio modalities through an RGB camera and a microphone array. By analyzing the individual speakers’ spatio-temporal activities and mouth movements, we propose a mid-fusion approach to perform both VAD and SSL for multiple active and inactive speakers. We tested the proposed algorithm in scenarios with up to three simultaneous speakers, showing an average VAD accuracy of 95.06% with an average error of 10.9 cm when estimating the three-dimensional locations of the speakers.

TDOA-based acoustic source localization in the space---range reference frame

Abstract: In this paper we consider a class of geometric methods for acoustic source localization based on range differences (or time differences of arrival), and we offer a new and unifying perspective on such methods based on the adoption of a multidimensional reference frame obtained by adding the range difference coordinate to the spatial coordinates of the source. In this extended coordinate system the working principles of a wide range of source localization methods becomes clear and immediate. The space---range reference frame, however, has a practical purpose as well, as it can be used for gaining insight on why some configurations of microphones lead to better localization performance than others and it suggests methods for improving existing localization techniques. In particular, we derive a closed-form solution of the constrained least-squares localization problem for linear arrays of microphones.

Method and apparatus for determining directions of uncorrelated sound sources in a higher order ambisonics representation of a sound field

Abstract: Higher Order Ambisonics (HOA) represents three-dimensional sound. HOA provides high spatial resolution and facilitates analysing of the sound field with respect to dominant sound sources. The invention aims to identify independent dominant sound sources constituting the sound field, and to track their temporal trajectories. Known applications are searching for all potential candidates for dominant sound source directions by looking at the directional power distribution of the original HOA representation, whereas in the invention all components which are correlated with the signals of previously found sound sources are removed. By such operation the problem of erroneously detecting many instead of only one correct sound source can be avoided in case its contributions to the sound field are highly directionally dispersed.

Use of the swim bladder and lateral line in near-field sound source localization by fish

Abstract: We investigated the roles of the swim bladder and the lateral line system in sound localization behavior by the plainfin midshipman fish ( Porichthys notatus ). Reproductive female midshipman underwent either surgical deflation of the swim bladder or cryoablation of the lateral line and were then tested in a monopolar sound source localization task. Fish with nominally ‘deflated’ swim bladders performed similar to sham-deflated controls; however, post-experiment evaluation of swim bladder deflation revealed that a majority of ‘deflated’ fish (88%, seven of the eight fish) that exhibited positive phonotaxis had partially inflated swim bladders. In total, 95% (21/22) of fish that localized the source had at least partially inflated swim bladders, indicating that pressure reception is likely required for sound source localization. In lateral line experiments, no difference was observed in the proportion of females exhibiting positive phonotaxis with ablated (37%) versus sham-ablated (47%) lateral line systems. These data suggest that the lateral line system is likely not required for sound source localization, although this system may be important for fine-tuning the approach to the sound source. We found that midshipman can solve the 180 deg ambiguity of source direction in the shallow water of our test tank, which is similar to their nesting environment. We also found that the potential directional cues (phase relationship between pressure and particle motion) in shallow water differs from a theoretical free-field. Therefore, the general question of how fish use acoustic pressure cues to solve the 180 deg ambiguity of source direction from the particle motion vector remains unresolved.

Hearing behind walls: localizing sources in the room next door with cosparsity

Abstract: Acoustic source localization is traditionally performed using cues such as interchannel time of arrival and intensity differences to infer the geometric localization of emitting sources with respect to the receiving microphone array. However the presence of obstacles between the sources and the array makes it impossible to rely on the direct path, and more advanced techniques are needed. The huge body of work on sparse recovery suggests an approach where source localization is expressed as a linear inverse problem and the spatial sparsity of the sources is exploited. An inverse problem can be naturally expressed in the recently introduced cosparse framework, exploiting the fact that the acoustic pressure satisfies the homogeneous wave equation except in the few locations of the sources. The resulting optimization problem involves a discretized second derivative analysis operator, which is extremely sparse. In this paper, we demonstrate the performance of the cosparse approach on an extreme source localization problem, where the microphone array is installed in the room next door to the room where the emitting sources are located, somehow hearing behind a wall.

Data-based matched-mode source localization for a moving source.

Abstract: A data-based matched-mode source localization method is proposed in this paper for a moving source, using mode wavenumbers and depth functions estimated directly from the data, without requiring any environmental acoustic information and assuming any propagation model. The method is in theory free of the environmental mismatch problem because the mode replicas are estimated from the same data used to localize the source. Besides the estimation error due to the approximations made in deriving the data-based algorithms, the method has some inherent drawbacks: (1) It uses a smaller number of modes than theoretically possible because some modes are not resolved in the measurements, and (2) the depth search is limited to the depth covered by the receivers. Using simulated data, it is found that the performance degradation due to the afore-mentioned approximation/limitation is marginal compared with the original matched-mode source localization method. The proposed method has a potential to estimate the source range and depth for real data and be free of the environmental mismatch problem, noting that certain aspects of the (estimation) algorithms have previously been tested against data. The key issues are discussed in this paper.

Shallow-water sparsity-cognizant source-location mapping.

Abstract: Using passive sonar for underwater acoustic source localization in a shallow-water environment is challenging due to the complexities of underwater acoustic propagation. Matched-field processing (MFP) exploits both measured and model-predicted acoustic pressures to localize acoustic sources. However, the ambiguity surface obtained through MFP contains artifacts that limit its ability to reveal the location of the acoustic sources. This work introduces a robust scheme for shallow-water source localization that exploits the inherent sparse structure of the localization problem and the use of a model characterizing the acoustic propagation environment. To this end, the underwater acoustic source-localization problem is cast as a sparsity-inducing stochastic optimization problem that is robust to model mismatch. The resulting source-location map (SLM) yields reduced ambiguities and improved resolution, even at low signal-to-noise ratios, when compared to those obtained via classical MFP approaches. An iterative solver based on block-coordinate descent is developed whose computational complexity per iteration is linear with respect to the number of locations considered for the SLM. Numerical tests illustrate the performance of the algorithm.

Sparse sound field representation in recording and reproduction for reducing spatial aliasing artifacts

Abstract: We propose a sound-pressure-to-driving-signal (SP-DS) conversion method for sound field reproduction based on sparse sound field representation. The most important problem in sound field reproduction is how to calculate driving signals of loudspeakers to reproduce desired sound fields. In common recording and reproduction systems, sound pressures at multiple positions obtained in a recording area are only known as the desired sound field; therefore, SP-DS conversion algorithms are necessary. Current SP-DS conversion methods do not take into account sound sources to be reproduced, which results in severe spatial aliasing artifacts. Our proposed method decomposes the received sound pressure distribution based on the generative model of the sound field. Numerical simulation results indicate that the proposed method can achieve higher reproduction accuracy compared to the current methods, especially in higher frequencies above the spatial Nyquist frequency.