Showing papers on "Acoustic source localization published in 2003"

Robust sound source localization using a microphone array on a mobile robot

Abstract: The hearing sense on a mobile robot is important because it is omnidirectional and it does not require direct line-of-sight with the sound source. Such capabilities can nicely complement vision to help localize a person or an interesting event in the environment. To do so the robot auditory system must be able to work in noisy, unknown and diverse environmental conditions. In this paper, we present a robust sound source localization method in three-dimensional space using an array of 8 microphones. The method is based on time delay of arrival estimation. Results show that a mobile robot can localize in real time different types of sound sources over a range of 3 meters and with a precision of 3/spl deg/.

Particle filtering algorithms for tracking an acoustic source in a reverberant environment

Abstract: Traditional acoustic source localization algorithms attempt to find the current location of the acoustic source using data collected at an array of sensors at the current time only. In the presence of strong multipath, these traditional algorithms often erroneously locate a multipath reflection rather than the true source location. A recently proposed approach that appears promising in overcoming this drawback of traditional algorithms, is a state-space approach using particle filtering. In this paper we formulate a general framework for tracking an acoustic source using particle filters. We discuss four specific algorithms that fit within this framework, and demonstrate their performance using both simulated reverberant data and data recorded in a moderately reverberant office room (with a measured reverberation time of 0.39 s). The results indicate that the proposed family of algorithms are able to accurately track a moving source in a moderately reverberant room.

Sound Detection Mechanisms and Capabilities of Teleost Fishes

Abstract: This chapter, written from the perspective of four authors who have been studying fish bioacoustics for over 120 years (cumulative!), examines the major issues of the field. Each topic is put in some historical perspective, but the chapter emphasizes current thinking about acoustic communication, hearing (including bandwidth, sensitivity, detection of signals in noise, discrimination, and sound source localization), the functions of the ear (both auditory and vestibular, and including the role(s) of the otoliths and sensory hair cells) and their relationships to peripheral structures such as the swim bladder, and the interactions between the ear and the lateral line. Hearing in fishes is not only for acoustic communication and detection of sound-emitting predators and prey but can also play a major role in telling fishes about the acoustic scene at distances well beyond the range of vision. The chapter concludes with the personal views of the authors as to the major challenges and questions for future study. There are still many gaps in our knowledge of fish bioacoustics, including questions on ear function and the significance of interspecific differences in otolith size and shape and hair cell orientation, the role of the lateral line vis-a-vis the ear, the mechanisms of central processing of acoustic (and lateral line) signals, the mechanisms of sound source localization and whether fishes can determine source distance as well as direction, the evolution and functional significance of hearing specializations in taxonomically diverse fish species, and the origins of fish (and vertebrate) hearing and hearing organs.

Coherent acoustic array processing and localization on wireless sensor networks

Abstract: Advances in microelectronics, array processing, and wireless networking have motivated the analysis and design of low-cost integrated sensing, computing, and communicating nodes capable of performing various demanding collaborative space-time processing tasks. In this paper, we consider the problem of coherent acoustic sensor array processing and localization on distributed wireless sensor networks. We first introduce some basic concepts of beamforming and localization for wide-band acoustic sources. A review of various known localization algorithms based on time-delay followed by least-squares estimations as well as the maximum-likelihood method is given. Issues related to practical implementation of coherent array processing, including the need for fine-grain time synchronization, are discussed. Then we describe the implementation of a Linux-based wireless networked acoustic sensor array testbed, utilizing commercially available iPAQs with built-in microphones, codecs, and microprocessors, plus wireless Ethernet cards, to perform acoustic source localization. Various field-measured results using two localization algorithms show the effectiveness of the proposed testbed. An extensive list of references related to this work is also included.

Coherent Acoustic Array Processing and Localization on Wireless Sensor Networks

Abstract: Advances in microelectronics, array processing, and wireless networking have motivated the analysis and design of low-cost integrated sensing, computing, and communicating nodes capable of performing various demanding collaborative space–time processing tasks In this paper, we consider the problem of coherent acoustic sensor array processing and localization on distributed wireless sensor networks We first introduce some basic concepts of beamforming and localization for wide-band acoustic sources A review of various known localization algorithms based on time-delay followed by least-squares estimations as well as the maximum–likelihood method is given Issues related to practical implementation of coherent array processing, including the need for fine-grain time synchronization, are discussed Then we describe the implementation of a Linux-based wireless networked acoustic sensor array testbed, utilizing commercially available iPAQs with built-in microphones, codecs, and microprocessors, plus wireless Ethernet cards, to perform acoustic source localization Various field-measured results using two localization algorithms show the effectiveness of the proposed testbed An extensive list of references related to this work is also included

Adaptive Signal Processing: Applications to Real-World Problems

Abstract: 1 On a Class of Exponentiated Adaptive Algorithms for the Identification of Sparse Impulse Responses.- 2 Adaptive Feedback Cancellation in Hearing Aids.- 3 Single-Channel Acoustic Echo Cancellation.- 4 Multichannel Frequency-Domain Adaptive Filtering with Application to Multichannel Acoustic Echo Cancellation.- 5 Filtering Techniques for Noise Reduction and Speech Enhancement.- 6 Adaptive Beamforming for Audio Signal Acquisition.- 7 Blind Source Separation of Convolutive Mixtures of Speech.- 8 Adaptive Multichannel Time Delay Estimation Based on Blind System Identification for Acoustic Source Localization.- 9 Algorithms for Adaptive Equalization in Wireless Applications.- 10 Adaptive Space-Time Processing for Wireless CDMA.- 11 The IEEE 802.11 System with Multiple Receive Antennas.- 12 Adaptive Estimation of Clock Skew and Different Types of Delay in the Internet Network.

Energy based acoustic source localization

Abstract: A novel source localization approach using acoustic energy measurements from the individual sensors in the sensor field is presented. This new approach is based on the acoustic energy decay model that acoustic energy decays inverse of distance square under the conditions that the sound propagates in the free and homogenous space and the targets are pre-detected to be in a certain region of the sensor field. This new approach is power efficient and needs low communication bandwidth and therefore, is suitable for the source localization in the distributed sensor network system. Maximum Likelihood (ML) estimation with Expectation Maximization (EM) solution and projection solution are proposed to solve this energy based source location (EBL) problem. Cramer-Rao Bound (CRB) is derived and used for the sensor deployment analysis. Experiments and simulations are conducted to evaluate ML algorithm with different solutions and to compare it with the Nonlinear Least Square (NLS) algorithm using energy ratio function that we proposed previously. Results show that energy based acoustic source localization algorithms are accurate and robust.

The Microflown, an acoustic particle velocity sensor

Abstract: The Microflown is an acoustic sensor directly measuring particle velocity instead of sound pressure, which is usually measured by conventional microphones. Since its invention in 1994 it is mostly used for measurement purposes (broadband1D and 3D-sound intensity measurement and acoustic impedance). Possible applications are near and far field sound source localization, in situ acoustic impedance determination and a non contact method to measure structural vibrations (an alternative for a laser vibrometer). Although the Microflown was invented only some years ago, the device is worldwide commercially available, see further www.microflown.com.

Applying scattering theory to robot audition system: robust sound source localization and extraction

Abstract: Robot audition by its own ears (microphones) is essential for natural human-robot communication and interface. Since a microphone is embedded in the head of a robot, the head-related transfer function (HRTF) plays an important role in sound source localization and extraction. Usually, from binaural input, the interaural phase difference (IPD) and interaural intensity difference (IID) are calculated, and then the direction is determined by using IPD and IID with HRTF. The problem of HRTF-based sound source localization is that a HRTF should be measured for each robot in an anechoic chamber, because it depends on the shape of robot's head; HRTF should be interpolated to manipulate a moving talker, because it is available only for discrete azimuth and elevation. To cope with these problems of HRTF, we proposed the auditory epipolar geometry as a continuous function of IPD and IID to dispense with HRTF and have developed a real-time multiple-talker tracking system. This auditory epipolar geometry, however, does not give a good approximation to IID of all range and IPD of peripheral areas. In this paper, the scattering theory in physics is employed to take into consideration the diffraction of sounds around robot's head for better approximation of IID and IPD. The resulting system shows that it is efficient for localization and extraction of sound at higher frequency and from side directions.

Robust adaptive time delay estimation for speaker localization in noisy and reverberant acoustic environments

Abstract: Two adaptive algorithms are presented for robust time delay estimation (TDE) in acoustic environments with a large amount of background noise and reverberation. Recently, an adaptive eigenvalue decomposition (EVD) algorithm has been developed for TDE in highly reverberant acoustic environments. In this paper, we extend the adaptive EVD algorithm to noisy and reverberant acoustic environments, by deriving an adaptive stochastic gradient algorithm for the generalized eigenvalue decomposition (GEVD) or by prewhitening the noisy microphone signals. We have performed simulations using a localized and a diffuse noise source for several SNRs, showing that the time delays can be estimated more accurately using the adaptive GEVD algorithm than using the adaptive EVD algorithm. In addition, we have analyzed the sensitivity of the adaptive GEVD algorithm with respect to the accuracy of the noise correlation matrix estimate, showing that its performance may be quite sensitive, especially for low SNR scenarios.

Integrated design for omni-directional camera and microphone array

Abstract: An omni-directional camera (a 360 degree camera) is proposed with an integrated microphone array. The primary application for such a camera is videoconferencing and meeting recording, and the device is designed to be placed on a meeting room table. The microphone array is in a planar configuration, and the microphones are located as close to the desktop as possible to eliminate sound reflections from the table. The camera is connected to the microphone array base with a thin cylindrical rod, which is acoustically invisible to the microphone array for the frequency range [50-4000] Hz. This provides a direct path from the person talking to all of the microphones in the array, and can therefore be used for sound source localization (determining the location of the talker) and beam-forming (improving the sound quality of the talker by filtering only sound from a particular direction). The camera array is elevated from the table to provide a near frontal viewpoint of the meeting participants.

Acoustic Source Localization and Beamforming: Theory and Practice

Abstract: We consider the theoretical and practical aspects of locating acoustic sources using an array of microphones. A maximum-likelihood (ML) direct localization is obtained when the sound source is near the array, while in the far-field case, we demonstrate the localization via the cross bearing from several widely separated arrays. In the case of multiple sources, an alternating projection procedure is applied to determine the ML estimate of the DOAs from the observed data. The ML estimator is shown to be effective in locating sound sources of various types, for example, vehicle, music, and even white noise. From the theoretical Cramer-Rao bound analysis, we find that better source location estimates can be obtained for high-frequency signals than low-frequency signals. In addition, large range estimation error results when the source signal is unknown, but such unknown parameter does not have much impact on angle estimation. Much experimentally measured acoustic data was used to verify the proposed algorithms.

Highly directional acoustic receivers.

Abstract: The theoretical directivity of a single combined acoustic receiver, a device that can measure many quantities of an acoustic field at a collocated point, is presented here. The formulation is developed using a Taylor series expansion of acoustic pressure about the origin of a Cartesian coordinate system. For example, the quantities measured by a second-order combined receiver, denoted a dyadic sensor, are acoustic pressure, the three orthogonal components of acoustic particle velocity, and the nine spatial gradients of the velocity vector. The power series expansion, which can be of any order, is cast into an expression that defines the directivity of a single receiving element. It is shown that a single highly directional dyadic sensor can have a directivity index of up to 9.5 dB. However, there is a price to pay with highly directive sensors; these sensors can be significantly more sensitive to nonacoustic noise sources.

Sound source location and quantification using arrays of vector probes

Abstract: Method and apparatus for locating and quantifying sound sources using an array of acoustic vector probes ( 200 ). Signals received at the probes are converted to digital form and fed into a digital signal processor ( 400 ) which computes the sound pressure and the sound-intensity vector at each probe. The set of sound-intensity vectors measured by the array provides a set of directions to a sound source ( 100 ) whose approximate spatial coordinates are determined using a least-squares triangulation formula. The sound-intensity vectors also determine sound-power flow from the source. In addition sound pressure measured by the probes can be phased to form a sensitivity beam ( 250 ) for scanning a source. Sound-intensity measurements made concurrently can be used to determine the spatial coordinates of the part being scanned and the sound power radiated by that part. Results are displayed on a computer screen or other device ( 500 ) permitting an operator to interact with and control the apparatus. Additional related features and methods are disclosed.

Robot recognizes three simultaneous speech by active audition

Abstract: Robots should listen to and recognize speeches with their own ears under noisy environments and simultaneous speeches to attain smooth communications with people in a real world. This paper presents three simultaneous speech recognition based on active audition which integrates audition with motion. Our robot audition system consists of three modules - a real-time human tracking system, an active direction-pass filter (ADPF) and a speech recognition system using multiple acoustic models. The real-time human tracking realizes robust and accurate sound source localization and tracking by audio-visual integration. The performance of localization shows that the resolution of the center of the robot is much higher than that of the peripheral. We call this phenomenon "auditory fovea" because it is similar to visual fovea (high resolution in the center of the human eye). Active motions such as being directed at the sound source improve localization because of making the best use if the auditory fovea. The ADPF realizes accurate and fast sound separation by using a pair of microphones. The ADPF separates sounds originating from the specified direction obtained by the real-time human tracking system. Because the performance of separation depends on the accuracy of localization, the extraction of sound from the front direction is more accurate than that of sound from the periphery. This means that the pass range of ADPF should be narrower in the front direction than in periphery. In other words, such active pass range control improves sound separation. The separated speech is recognized by the speech recognition using multiple acoustic models that integrates multiple results to output the result with the maximum likelihood. Active motions such as being directed at a sound source improve speech recognition because it realizes not only improvement of sound extraction but also easier integration of the results using face ID by face recognition. The robot audition system improved by active audition is implemented on an upper-torso humanoid. The system attains localization, separation and recognition of three simultaneous speeches and the results proves the efficiency of active audition.

Estimation of the number of sound sources using support vector machines and its application to sound source separation

Abstract: A method of estimating the number of sound sources in a reverberant sound field is proposed in this paper. It is known that the eigenvalue distribution of the spatial correlation matrix calculated from a multiple microphone input reflects information on the number of sources. However, in a reverberant sound field, the feature of the number of sources in the eigenvalue distribution is degraded by the room reverberation. In this paper, support vector machines is applied to classify the eigenvalue distributions which are not clearly separable. The proposed method is then applied to the source separation system and is evaluated via automatic speech recognition.

Voice recognition apparatus, voice recognition apparatus and program thereof

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 voice 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 voice recognition is carried out based on an obtained assumption value.

New direct approaches to robust sound source localization

Abstract: When more than two microphones are used, the traditional time-delay-of-arrival (TDOA) based sound source localization (SSL) approach involves two steps. The first step computes TDOA for each microphone pair, and the second step combines these estimates. This two-step process discards relevant information in the first step, thus degrading the SSL accuracy and robustness. Although less used, one-step processes do exist. In this paper, we review these processes, create a unified framework, and introduce two new one-step algorithms. We compare our proposed approaches against existing 1and 2-step approaches and demonstrate significantly better SSL performance.

Non-linear effects in a 200-kHz sound beam and the consequences for target-strength measurement

Abstract: When estimating target strength (TS), sound pressure in the beam where the target is located has to be measured accurately. Sound pressure is normally calculated from the source level, and transmission loss is based on geometric spreading and absorption loss. Additional losses caused by non-linear acoustic propagation may be important, especially in the case of high-power, high-frequency, and highly directive sources. ‘Non-linear loss’ from the fundamental frequency is due to energy from the fundamental harmonic being transferred into higher harmonics. This loss affects the beam pattern in ways that will depend on both power and range, since the non-linear loss depends, in turn, on sound pressure. We present the results of sound-pressure measurements and simulations from a 200-kHz transducer with a beam width of 7 � . Sound pressure was measured at different ranges and power levels using a broadband hydrophone to detect some of the higher harmonic frequencies that occur in non-linear acoustic propagation. The TS of a solid copper sphere was measured using a standard echosounder with no correction for non-linear loss. Our study illustrates that this can be significant and that a corresponding correction needs to be considered when estimating TS.

Sound source search system

Abstract: It is possible to simultaneously identify the sound coming direction from a sound source in all directions and estimate the sound intensity of the sound source. A plurality of microphones (11) are arranged on the surface of a baffle (10) of a shape such as a sphere and polyhedron so that sound from all directions are acquired. A calculation device (40) calculates the amplitude characteristic and the phase characteristic of acoustic signals acquired by the microphones (11). The signal information and information on sound field analysis around the baffle are integrated and calculation to emphasize a sound coming from a particular direction is performed for all the directions so as to identify the sound coming direction from a sound source. According to these calculation results and the distance input by an input device (70), it is possible to estimate the sound intensity of the sound source at a plurality of portions generated at the sound source or boundary surface.

Active control of acoustic reflection, absorption, and transmission using thin panel speakers

Abstract: This paper explores the development of thin panels that can be controlled electronically so as to provide surfaces with desired reflection coefficients. Such panels can be used as either perfect reflectors or absorbers. They can also be designed to be transmission blockers that block the propagation of sound. The development of the control system is based on the use of wave separation algorithms that separate incident sound from reflected sound. In order to obtain a desired reflection coefficient, the reflected sound is controlled to appropriate levels. The incident sound is used as an acoustic reference for feedforward control and has the important property of being isolated from the action of the control system speaker. In order to use a panel as a transmission blocker, the acoustic pressure behind the panel is driven to zero. The use of the incident signal as a reference again plays a key role in successfully reducing broadband transmission of sound. The panels themselves are constructed using poster board and small rare-earth actuators. Detailed experimental results are presented showing the efficacy of the algorithms in achieving real-time control of reflection or transmission. The panels are able to effectively block transmission of broadband sound. Practical applications for these panels include enclosures for noisy machinery, noise-absorbing wallpaper, the development of sound walls, and the development of noise-blocking glass windows.

Wideband RELAX and Wideband CLEAN for Aeroacoustic Imaging

Abstract: Microphone arrays can be used for acoustic source localization and characterization in wind tunnel testing. In this paper, the wideband RELAX (WB-RELAX) and the wideband CLEAN (WB-CLEAN) algorithms are presented for aeroacoustic imaging using an acoustic array. WB-RELAX is a parametric approach that can be used efficiently for point source imaging without the sidelobe problems suffered by the delay-and-sum beamforming approaches. WB-CLEAN does not have sidelobe problems either, but it behaves more like a nonparametric approach and can be used for both point source and distributed source imaging. Moreover, neither of the algorithms suffers from the severe performance degradations encountered by the adaptive beamforming methods when the number of snapshots is small and/or the sources are highly correlated or coherent with each other. A two-step optimization procedure is used to implement the WB-RELAX and WB-CLEAN algorithms efficiently. The performance of WB-RELAX and WB-CLEAN is demonstrated by applying them to measured data obtained at the NASA Langley Quiet Flow Facility using a small aperture directional array (SADA). Somewhat surprisingly, using these approaches, not only were the parameters of the dominant source accurately determined, but a highly correlated multipath of the dominant source was also discovered.

A 3d real time rendering engine for binaural sound reproduction

Abstract: A method of computationally efficient 3D sound reproduction via headphones is presented using a virtual Ambisonic approach. Previous studies have shown that incorporating head tracking as well as room simulation is important to improve sound source localization capabilities. The simulation of virtual acoustic space requires to filter the stimuli with head related transfer functions (HRTFs). In time-varying systems this yields the problem of high quality interpolation between different HRTFs. The proposed model states that encoding signals into Ambisonic domain results in time-invariant HRTF filters. The proposed system is implemented on a usual notebook using Pure Data (PD), a graphically based open source real time computer music software.

Adaptive Multichannel Time Delay Estimation Based on Blind System Identification for Acoustic Source Localization

Abstract: Time delay estimation is a difficult problem in a reverberant environment and the traditional generalized cross-correlation (GCC) methods do not perform well. Recently, a blind channel identification-based adaptive approach called the eigenvalue decomposition (ED) algorithm has been proposed to deal with room reverberation more effectively. The ED algorithm focuses on a system with two channels whose impulse responses can be blindly identified only if they do not share common zeros. The assumption often breaks down for acoustic channels whose impulse responses are long. In this chapter, the blind channel identification-based time delay estimation approach is generalized to multiple (more than 2) channel systems and a normalized multichannel frequency-domain LMS algorithm is proposed. The proposed algorithm is more practical since it is less likely for all channels to share a common zero when more channels are available. It is shown by using the data recorded in the Varechoic chamber at Bell Labs that the proposed method performs better than the ED and GCC algorithms.

Design and experiments of bio-mimicry sound source localization sensor with gimbal-supported circular diaphragm

Abstract: We propose a novel structure for sound source localization by mimicking an auditory organ of a parasitoid fly. It is based on lateral gradients of sound pressure from a non-frontal source. The gimbal-supported circular diaphragm is very suitable for detecting the small gradients and miniaturizing into a micro sensor. We describe the design method and FEM analysis of the diaphragm structure. We show fabrication of two types of sensors, a scaled-up type and a surface-micromachined type, and the experimental results.

System and process for sound source localization using microphone array beamsteering

Abstract: A system and process for sound source localization (SSL) utilizing beamsteering is presented. The present invention provides for improved beamsteering with less drain on system resources while providing accurate, real time results. To accomplish this, the present SSL system and process rejects as much as possible extraneous audio frames and analyzes only those frames exhibiting a well defined sound source. In addition, the number of beams is reduced as much as possible to save on processing time, but a full scan of the working volume is still made with the beams. And finally, interpolation is used to increase the precision of the technique.