Showing papers on "Acoustic source localization published in 1995"

Active control of sound radiation using volume velocity cancellation

Abstract: The active control of sound transmission through a panel has been formulated using a near‐field approach. The effects of minimizing the sound power radiated by the panel and of canceling the net volume velocity of the panel are compared not only in terms of the reduction in sound radiation but also in terms of the change in the space average mean‐squared velocity of the panel and the space average mean‐squared pressure at its surface. Simulations of a thin panel excited by an incident acoustic plane wave and a piezoelectric control actuator show that volume velocity cancellation gives similar reductions in the transmitted sound power to the minimization of sound power radiation up to frequencies at which the size of the plate is about half an acoustic wavelength. The acoustic radiation is analyzed in terms of the radiation modes of the panel which are also used to explain spillover effects. Spillover, which leads to increases in the mean‐squared velocity of the panel and to increases in near‐field pressur...

Direct computation of the sound from a compressible co-rotating vortex pair

Abstract: The far-field sound from corotating vortices is computed by direct computation of the unsteady, compressible Navier-Stokes equations on a computational mesh that extends to two acoustic wavelengths in all directions. The vortices undergo a period of corotation followed by a sudden merger. A 2D version of Moehring's equation is developed and used in conjunction with source terms computed in the simulation to predict the far-field sound. The prediction agrees with the simulation to within 3 percent. Results of far-field pressure fluctuations for an acoustically noncompact case are also presented for which the prediction is 66 percent too high. Results also indicate that the monopole contribution of 'viscous sound' is negligible for this flow.

System and method for measuring acoustic reflectance

Abstract: A system and method of measuring the linear and nonlinear response of an unknown acoustic termination uses a small probe assembly containing a sound source and microphone to determine the reflection function of the unknown acoustic termination. The probe assembly is used with a calibration tube to calculate an electrical signal that will provide a desired acoustic stimulus signal to the acoustic termination. The calibration tube is also used to characterize the signal processing properties of the sound source and microphone, as well as other associated signal processing circuits such as amplifiers, filters, and the like. The calibrated system is subsequently coupled to the unknown acoustic termination to deliver the acoustic stimulus signal. The reflection function is indicative of the power transferred to the unknown acoustic termination. The measurement of the linear transfer characteristic is applicable to any unknown acoustic termination such as a musical instrument or the auditory system. The probe assembly is sized to be positioned directly within the outer portion of the ear and measure the linear characteristics of the ear. The system is further able to measure the nonlinear transfer characteristics of the ear by measuring the linear response at multiple levels of the acoustic stimulus. The system is particularly useful in testing the response of the middle ear and inner ear of humans or other animals.

Investigation of acoustic streaming excited by surface acoustic waves

Abstract: Fluid motion due to high intensity sound is known as acoustic streaming. These flows are generated around obstacles immersed in sound or near oscillating bodies. Boundary streamings occur near the surface of a body in a sound field. In particular, streaming excited by SAW (surface acoustic waves) results in a slender jet. We investigated this streaming theoretically and experimentally. We applied the acoustic streaming theory and SAW transfer theory to the streaming excited by SAW.

Knowing who to listen to in speech recognition: visually guided beamforming

Abstract: With speech recognition systems steadily improving in performance, freedom from head-sets and push-buttons to activate the recognizer is one of the most important issues to achieve user acceptance. Microphone arrays and beamforming can deliver signals that suppress undesired jamming signals but rely on knowledge where the signal is in space. This knowledge is usually derived by identifying the loudest signal source. Knowing who is speaking to whom and where should however not depend on loudness, but on the communication purpose. In this paper, we present acoustic and visual modules that use tracking of the face of a speaker of interest for sound source localization and beamforming for signal extraction. It is shown that in noisy environments a more accurate localization in space can be delivered visually than acoustically. Given a reliable location finder, beamforming substantially improves recognition accuracy.

The nature and technology of acoustic space

Abstract: Introduction. Signal Analysis: Continuous Fourier Transform. Discrete Fourier Transform. Hilbert Transform. Time-Frequency Domain Analysis. Sound Propagation in an Enclosed Space: Reflection at a Boundary. Sound Power Output of a Source. Room Reverberation Theory. Statistics of Random Sound Fields. Transfer Function Statistics. Statistical Phase Analysis (SPA). Modulation Transfer Function of a Reverberant Space. Visualization of the Sound Field in a Room: Sound Intensity and Measurement Method. Intensity Field in a Room. Wave Field Around Sources. Other Types of Sound Intensities. Subjective and Physiological Responses to Sound Fields: Theory of Subjective Preference and the Limitation of Orthogonal Acoustic Factors. Slow Vertex Responses (SVR) Relating to the Subjective Preferences. Auditory Brainstem Responses (ABR) as a Function of the Horizontal Angle to a Listener. Model of the Auditory-Brain System. Individual Preference. Subjective Attributes. Sound Field Control in an Enclosed Space: Passive Control for Sound Power Transmission. Sound Field Reproduction and Sound Image Control. Inverse Filtering for a Reverberant-Sound Path. Active and Adaptive Control of Reverberant Sounds. SoundField Control for Concert Hall Acoustics: Sound Reinforcement for Speech Intelligibility. Sound Source Localization in a Reverberant Space. Sound Field Control for Concert Halls. A Design Study for Concert Halls. Exercises. Appendices. References. Index.

Plasticity in human directional hearing.

Abstract: Interaural time difference (ITD), the main cue for localization of low-frequency sound in azimuth, is widely thought to be evaluated according to Jeffress' model. This theory proposes that each of an array of neurons detects coinciding input from both ears, conducted along axonal delay lines, with the azimuth angle corresponding to the activation of selected neurons. Thus, sound source localization is assumed to depend on axon conduction velocities, a relatively fixed parameter. Clinical experience suggests that directional hearing is adaptable. We investigated if sound localization in azimuth could adapt plastically to altered ITDs. We equipped binaural insert hearing aids with adjustable electronic delay lines. Subjects with normal hearing were required to wear these devices during all waking hours for several days. Localization of an invisible sound source was measured in an anechoic room before and at various intervals after introduction of a constant delay in one ear between 171 and 684 mus. Test sounds were high-pass, low-pass and broad-band noises. Introduction of a delay in one ear lead to an immediate displacement of the perceived sound location towards the opposite side. Within hours of exposure, the displacement was reduced, and further normalization of the perceived localization occurred over several days. After removal of the delays sound localization normalized rapidly. We conclude that ITD alterations can lead to plastic adaptation of directional hearing, which cannot rely exclusively on fixed axon conduction velocities. Our results suggest additional mechanisms for directional hearing on the basis ITD.

A Design Method for Minimizing the Sound Power Radiated from Plates by Adding Optimally Sized, Discrete Masses

Abstract: In this paper, a novel method for minimizing the sound power radiated from a structure is presented. The method involves placing strategically sized masses at specific locations on the structure's surface. The minimization procedure modifies the shapes of the resonant modes of the structure in the frequency range of interest such that they are forced to radiate sound inefficiently. Because of this, they are referred to as weak radiator mode shapes. The method uses an optimization procedure that directly minimizes the radiated sound power from the surface of a plate in an infinite baffle. The procedure can be carried out for a single frequency or over a range of frequencies. Analytical sensitivities of sound power with respect to the design variables are developed and used in the optimization algorithm. Results on various test cases show sound power reductions of 10 dB or more even when several resonances are included in the frequency band. An acoustic intensity probe is used to experimentally verify the results for one test case. The experiment confirms the sound power reductions predicted by the optimization program.

Scattering of sound from a spatially distributed, spherically symmetric source by a sphere

Abstract: General expressions are derived for the scattering of sound from a spatially distributed, spherically symmetric source by rigid and nonrigid spheres. The incident field is determined using a Hankel transform for spherically symmetric variables. The scattered fields are determined by matching the total pressure and normal velocity at the surface of the sphere. Examples of the total and scattered fields are given for a ball source and a spherically symmetric source with a Gaussian spatial distribution. These solutions are compared with results for a point source.

The scattering of sound from a spatially distributed axisymmetric cylindrical source by a circular cylinder

Abstract: General expressions for the scattering of sound from a spatially distributed cylindrical source by a circular cylinder are given The incident field is determined using a Hankel transform The scattered fields are determined by matching the pressure and radial velocity at the surface of the cylinder Examples of total and scattered pressure fields are given for a cylindrical line source and a axisymmetric source with a Gaussian spatial distribution An example is also given for a circular disk source

Apparatus for controlling localization of a sound image

Abstract: The present invention discloses a sound image localization apparatus for localizing a sound image at an arbitrary location in three-dimensional space by adding an attenuation in distance to a digital filter in order to reduce an operation time of convolution and approximating the head related transfer function in a three-dimensional space thereby to control the localization in real time. The sound image localization control apparatus comprises a location sensor for three-dimensional measuring of the direction and location of a listener's head, a microprocessor for correcting sound pressure attenuation in proportion to the distance between a sound source and the head relative to a digital filter that approximates the head related transfer function consistent with the direction of the head, and a convolution processor for convolving the corrected digital filter with the monaural sound source data.

Sound equalization in enclosures using modal reconstruction

Abstract: The equalization of complex sound pressure in an enclosure is investigated in this paper. The sound field in the enclosure is modeled with the sum of a series of modes. This sound field is controlled by multiple sources distributed in the enclosure so that a certain region in the enclosure has a desired complex sound pressure. First, the optimum solution which minimizes the average potential energy of the error pressure over the region is derived, where the error is defined as the difference between the desired sound pressure and the sound pressure caused by the sources. By using the optimum solution, the achievable best performance can be known for the given region and the source distribution. The optimum solution can also be a useful tool for finding an effective source distribution since the performance of the optimum solution depends only on the source distribution. Furthermore, the eigenvalues of the source coupling matrix, which appears in the derivation of the optimum solution, indicates the effectiveness of the source distribution. The effectiveness of source distribution is discussed with nine examples of source distributions in a two‐dimensional enclosure. The multipoint equalization method to approximate the optimum solution is also discussed for the practical use.

The effect of refraction and assumed speeds of sound in tissue and blood on doppler ultrasound blood velocity measurements

Abstract: The combined effect of three assumptions relating to refraction, the speed of sound in tissue and the speed of sound in blood on the accuracy of Doppler ultrasound blood velocity measurements has been investigated. A theoretical relationship giving the net velocity measurement error introduced by these three assumptions has derived using a model in which tissue and blood layers are separated by straight, parallel boundaries. This net error is dependent on the assumed and actual speed of sound in tissue, the assumed speed of sound in blood and the Doppler angle, but is effectively independent of the actual speed of sound in blood. For clinical blood velocity measurements, the net error is estimated to be as much as an 8% overestimation of the actual velocity, higher than previously predicted for any of the factors individually. The relationship also predicts a net velocity measurement error in experimental flow systems and string phantoms which is dependent on the speed of sound in the liquid bath. A water bath at room temperature will give an overestimation of approximately 2%. Experimental investigations using conventional and modified string phantoms and a 5-MHz linear phased array system support these conclusions. The effect of perturbing the layers from their parallel orientation has also been considered theoretically and has provided additional support for the above conclusions. These results may help more accurate Doppler velocity measurements in both experimental and clinical settings.

A note on the use of an approximate formula to predict sound fields above an impedance plane due to a point source

Abstract: The application of an approximate equation to describe the propagation of sound from a point source above an impedance plane [e.g., C. F. Chien and W. W. Soroka, J. Sound Vib. 43, 9–20 (1975)] is discussed. In particular, the correct choice of a complex root is specified. This choice is made by ensuring that the sound pressure be a bounded and continuous function of the geometrical parameters and the surface impedance.

Outdoor sound propagation in the presence of atmospheric turbulence: Experiments and theoretical analysis with the fast field program algorithm

Abstract: Results of experimental measurements of sound propagating over the range from 16 to 250 m at frequencies between 160 and 2000 Hz in a refracting atmosphere are described. Wind velocity and temperature are also measured simultaneously with the acoustical measurements at different heights. The meteorological measurements provide the sound‐speed profile as well as the strength of turbulence up to an elevation of 32 m. The measurements are sampled every 0.25 s, thus providing a continuous record of sound‐speed profiles and sound‐pressure levels over time. The individual sound‐speed profile is used to generate predicted sound levels that are compared with the measured sound levels.

Full field tomographic reconstruction of sound fields using TV holography

Abstract: A vibrating sound source causes periodic variations of the refractive index in the surrounding medium. A light wave passing through the sound field will experience a corresponding variation of its path length which can be measured by interferometric techniques like TV holography. One TV‐holography recording represents the integrated optical pathlength in one direction. The sound source is rotated to record cross sections of the field as seen from different directions. By tomographic backprojection of these recordings, afterward the amplitude and phase of the sound field in any plane of the volume are reconstructed.

Four dimensional acoustical audio system for a homogeneous sound field

Abstract: A four-dimensional acoustical audio system utilizes both spatial and temporal signal processing to provide uniform sound throughout a range of ear levels in an enclosure and combines the output of loudspeakers above and below the desired level to substantially eliminate the perceived direction and location of the sound sources and effectively creates a phantom sound source at the desired level

A second note on the prediction of sound intensity

Abstract: The modal expansion of the sound field in an enclosure often employs rigid-wall modes of the enclosure as base functions. Simple analytical expressions of the modal amplitudes can be obtained if the specific acoustic-impedance ratio of the enclosure boundary is large and the modal coupling among the rigid-wall modes is negligible. It has been demonstrated that the analytical expressions fail to give a correct prediction of sound intensity [J. Pan, J. Acoust. Soc. Am. 93, 1641–1644 (1993)]. Even including the modal coupling, prediction error still exists as a result of the poor convergence inherited in the use of the rigid-wall mode [J. Pan, J. Acoust. Soc. Am. 97, 691–694 (1995)]. This note discusses a new approach of using the extended mode shape functions for the expansion of sound pressure. The new approach significantly improves the accuracy in the prediction of sound pressure and sound intensity in enclosures.

Instrumentation for the Measurement of Sound Speed near the Ocean Surface

Abstract: Air bubbles entrained by breaking waves in the ocean surface layer can dramatically alter the velocity and attenuation of acoustic waves. The development of an effective technique for directly measuring the sound speed near the ocean surface is reported. The method makes use of the travel time of short acoustic pulses between a transmitter and a receiver separated by 40 cm. Phase distortions caused by acoustic reflections from the surface or from nearby buoy structural elements are separated in time from the direct path signal. A DSP-based data processing system was implemented to cross correlate the transmitted and received acoustic pulses and thus yield sound-speed measurements in real time. Perhaps the most significant novelty of the present measurement technique is its ability to make simultaneous measurements of the sound speed at several depths, starting as close as 0.5 m to the surface, at frequencies down to 5 kHz, and at a sample rate of 4 Hz per channel. Furthermore, the technique is di...

Signal source search method and device

Abstract: PROBLEM TO BE SOLVED: To detect the position and characteristics of a signal source without arranging a sensor in an expecting signal source position by analyzing various characteristics containing the space information of sound source from an observation signal obtained with a microphone array arranged in the specified position. SOLUTION: Microphones 10A-10D are arranged in a sound field in the specified arrangement, detect sound propagating in the sound field, convert it into an electric signal, then output the signal to a sampling part 12. These microphones are arranged at the intervals of 1/2 or less the wave length of a signal to be detected. The sampling part 12 conducts sampling in the specified sampling period, and outputs to an FFT operation part 14. Fourier transform of sampling data is conducted in the FFT operation part 14, and high frequency spectrum obtained is outputted to a sound source search operation part 16. The operating part 16 calculates the position of sound source and the strength of the sound source of each frequency component (a frequency spectrum of generated sound) from the frequency spectrum of the signal detected with the microphones 10A-10D, and the calculated results are outputted to a result display part 18.

Use of pitch-azimuth plots in determining the direction of a noise source in water with a vector sound-intensity probe

Abstract: A vector sound‐intensity probe is used to determine the direction of a sound source in water. The probe consists of four pressure sensors in the tetrahedral arrangement, with each component of sound intensity measured simultaneously using the cross‐spectral method. The probe dimensions are at least ten times less than the measured wavelengths. The probe has omnidirectional sensitivity with no 180 deg ambiguity. For plane waves, it is shown that direction finding with the vector probe is based on phase differences. Tests were conducted in a water tank to test the accuracy of direction finding of a sound source, with different types of noise interference. Results are expressed in the form of pitch‐azimuth plots where the data points correspond to different frequencies in the frequency range of the measurements. These plots provide an all around three‐dimensional view of the ambient sound field. Without interference the accuracy of direction finding is within ±2 deg. It is shown that, for an equivalent direc...

Method of and apparatus for controlling noise generated in confined spaces

Abstract: A method of and an apparatus for controlling noise generated in a confined space, being capable of reducing a radiating sound pressure generated from a main noise source to that of an optimal state. The method includes the steps of measuring the radiating sound pressure generated from the noise source, and generating, from an additional sound source, a radiating sound pressure having the same magnitude as the radiating sound pressure generated from the noise source while having a phase 180°-shifted from that of the noise source's radiating sound pressure so that the radiating sound pressures can offset each other when they are mixed. The apparatus includes an additional sound source installed in the confined space, an intensity converter for collecting and measuring sound pressure signals respectively generated from the noise source and the additional sound source, and a microcomputer for applying, to the additional sound source, a control signal for reducing the noise on the basis of the sound pressure signals measured by the intensity converter.

Experiments on the active control of sound radiation using a volume velocity sensor

Abstract: The reduction of sound radiation from vibrating structures is an important problem in acoustics. The sound power radiated from a structure can be reduced by altering the dynamic properties of the structure or by isolating the structure from the source of excitation. This is generally termed passive control. In some applications, notably aircraft, the weight penalty imposed by passive control techniques can be prohibitively large, especially at low frequencies. In these applications active control systems which use secondary control actuators can be used as an alternative technique for reducing the sound power radiation.2''1 It has been shown that the volume velocity of a surface is responsible for the majority of the sound power radiation at low frequencies.3 It has therefore been suggested that the cancellation of volume velocity is an appropriate strategy for reducing the sound power radiation from vibrating surfaces.7 In order to actively cancel the volume velocity of a surface an accurate measure of the volume velocity is required. Designs for volume velocity sensors have been suggested using piezoelectric material etched or cut into specific shapes.4'6"° In the experiments described in this paper such a sensor is tested and used to control the sound power radiation from a rectangular aluminium panel.

Hull-mounted acoustic vector-sensor processing

Abstract: We consider the direction-of-arrival (DOA) estimation problem using arrays of acoustic vector-sensors located at a solid-liquid boundary. We formulate a measurement model and derive an expression for the Cramer-Rao bound (CRB) of a single source. The CRB is used to analyse the case of rigid and pressure-release surfaces in detail and comparison is made with pressure-sensor arrays. In particular, we show that a pressure-sensor array needs to be somewhat less than a quarter wavelength from a pressure-release surface to have comparable performance to an array of velocity sensors mounted at the surface.

Sound velocity profile signal processing system and method for use in sonar systems

Abstract: A method and apparatus for calculating a sound velocity or sound velocity error in an ocean, gulf, sea, bay, littoral region, lake, or river at a first locale using a sensing system (for example, a sonar system) having at least one sensor The technique includes calculating echo time data of a first sounding which is projected in the direction of the first locale at a first angle; calculating first bathymetric data for the first locale using echo time data of the first sounding; calculating echo time data of a second sounding which is projected in the direction of the first locale at a second angle; calculating second bathymetric data for the first locale using echo time data of the second sounding; and calculating sound velocity at the first locale using the first bathymetric data and the echo time data of the first sounding and second bathymetric data and the echo time of the second sounding The sound velocity error and/or sound velocity may be used to further calculate a sound velocity profile, sound velocity error profile, sound velocity gradient, and sound velocity error gradient The signal processing unit employs the sound velocity error, sound velocity, sound velocity profile, sound velocity error profile, sound velocity gradient, and/or sound velocity error gradient to "correct" erroneous bathymetric data caused by changes in the sound velocity The velocity of sound varies at a given location according to temperature, salinity, and pressure or depth

Nonlinear wave propagation in horns and ducts

Abstract: The one‐parameter sound field of finite amplitude is modeled by an acoustic transmission line model and by block‐oriented system models containing dynamic linear and static nonlinear subsystems. The transmission line model is based on conical elements and each element is represented by a linear four‐port and a nonlinear source of volume velocity derived from the nonlinear wave equation in Lagrangian coordinates. The block‐oriented system model with a lattice structure shows the forward and backward propagating sound‐pressure waves separately. A simplified overall model and the derived higher‐order system functions based on the Volterra approach describe the relation between the sound pressure at two points in the sound field. The presented models are the basis for numerical simulations, nonlinear system identification, and signal processing related to the nonlinear sound propagation in horns, ducts, and other acoustical systems with a nearly one‐parameter sound field.