The effect of right versus left long-term single-sided deafness on sound source localization

Purpose:To analyze the effect of right versus left long-term single-sided deafness(SSD)on sound source localization(SSL),discuss the necessity of intervention and treatment for SSD patients,and analyze the therapeutic effect of long-term unilateral cochlear implantation(UCI)from the perspective of SSL.Methods:This study included 25 patients with SSD,11 patients with UCI,and 30 participants with normal hearing(NH).Their SSL ability was tested by obtaining their average root mean square(RMS)error values of SSL test.Results:The results showed that the RMS error value of SSD,UCI and NH groups were 52.26±20.25◦,69.84±12.14◦and 4.27±2.66◦,respectively.The ability of SSL was better in the SSD-L group than that in the SSD-R group,and no significant difference existed in the SSD-R and the UCI group.Conclusion:When bilateral deafness patients select unilateral treatment,right-side cochlear implantation may be more beneficial in terms of SSL,which means that the central auditory cortex in long-term SSD patients is affected differently based on which side their deafness occurs.

Piecewise Acoustic Source Imaging with Unknown Speed of Sound Using a Level-Set Method

We investigate the following inverse problem:starting from the acoustic wave equation,reconstruct a piecewise constant passive acoustic source from a single boundary temporal measurement without knowing the speed of sound.When the amplitudes of the source are known a priori,we prove a unique determination result of the shape and propose a level set algorithm to reconstruct the singularities.When the singularities of the source are known a priori,we show unique determination of the source amplitudes and propose a least-squares fitting algorithm to recover the source amplitudes.The analysis bridges the low-frequency source inversion problem and the inverse problem of gravimetry.The proposed algorithms are validated and quantitatively evaluated with numerical experiments in 2D and 3D.

Sound Source Localisation for a High-Speed Train and Its Transfer Path to Interior Noise

Noise is one of the key issues in the operation of high-speed railways, with sound source localisation and its transfer path as the two major aspects. This study investigates both the exterior and interior sound source distribution of a high-speed train and presents a method for performing the contribution analysis of airborne sound with regard to the interior noise. First, both exterior and interior sound source locations of the high-speed train are identified through in-situ measurements. Second, the sound source contribution for di erent regions of the train and the relationships between the exterior and interior noises are analysed. Third, a method for conducting the contribution analysis of airborne sound with regard to the interior noise of the high-speed train is described. Lastly, a case study on the sidewall area is carried out, and the contribution of airborne sound to the interior noise of this area is obtained. The results show that, when the high-speed train runs at 310 km/h, dominant exterior sound sources are located in the bogie and pantograph regions, while main interior sound sources are located at the sidewall and roof. The interior noise, the bogie area noise and the sound source at the middle of the coach exhibit very similar rates of increase with increasing train speed. For the selected sidewall area, structure-borne sound dominates in most of the 1/3 octave bands.

Numerical Calculation of Seafloor Synthetic Seismograms Caused by Low Frequency Point Sound Source

Elastic wave on seafloor caused by low frequency noise radiated from ship is called ship seismic wave which can be used to identify ship target. In order to analyze the wave components and the propagating properties of ship seismic wave, the numerical calculation of synthetic seismograms on seafloor aroused by a low frequency point sound source is carried out using a wave number integration technique combined with inverse Fourier transform. According to the numerical example of hard seafloor, the time series of seismic wave on seafloor are mostly composed of interface waves and normal mode waves. Each normal mode wave has a well defined low cut-off frequency, while the interface wave doesn't have. The frequency dispersion of normal mode wave is obvious when frequency is lower than 100Hz, while the interface wave is dispersive only in the infra-sound frequency range. The time series of seismic wave is dominated by the interface wave when the source frequency is less than the minimal cut-off frequency of normal mode wave.

Identification of diesel front sound source based on continuous wavelet transform

Acoustic signals from diesel engines contain useful information but also include considerable noise components To extract information for condition monitoring purposes, continuous wavelet transform (CWT) is used for the characterization of engine acoustics. This paper first reviews CWT characteristics represented by short duration transient signals. Wavelet selection and CWT are then implemented and wavelet transform is used to analyze the major sources of the engine front's exterior radiation sound. The research provides a reliable basis for engineering practice to reduce vehicle sound level. Comparison of the identification results of the measured acoustic signals with the identification results of the measured surface vibration showed good agreement.

A Particle Filter Based Compressive Sensing Method for Tracking Moving Wideband Sound Sources

Tracking moving wideband sound sources is one of the most challenging issues in the acoustic array signal processing which is based on the direction of arrival(DOA) estimation. Compressive sensing(CS) is a recent theory exploring the signal sparsity representation, which has been proved to be superior for the DOA estimation. However, the spatial aliasing and the offset at endfire are the main obstacles for CS applied in the wideband DOA estimation. We propose a particle filter based compressive sensing method for tracking moving wideband sound sources. First, the initial DOA estimates are obtained by wideband CS algorithms. Then, the real sources are approximated by a set of particles with different weights assigned. The kernel density estimator is used as the likelihood function of particle filter. We present the results for both uniform and random linear array. Simulation results show that the spatial aliasing is disappeared and the offset at endfire is reduced. We show that the proposed method can achieve satisfactory tracking performance regardless of using uniform or random linear array.

Sound Source Localization Based on SRP-PHAT Spatial Spectrum and Deep Neural Network

Microphone array-based sound source localization(SSL)is a challenging task in adverse acoustic scenarios.To address this,a novel SSL algorithm based on deep neural network(DNN)using steered response power-phase transform(SRP-PHAT)spatial spectrum as input feature is presented in this paper.Since the SRP-PHAT spatial power spectrum contains spatial location information,it is adopted as the input feature for sound source localization.DNN is exploited to extract the efficient location information from SRP-PHAT spatial power spectrum due to its advantage on extracting high-level features.SRP-PHAT at each steering position within a frame is arranged into a vector,which is treated as DNN input.A DNN model which can map the SRP-PHAT spatial spectrum to the azimuth of sound source is learned from the training signals.The azimuth of sound source is estimated through trained DNN model from the testing signals.Experiment results demonstrate that the proposed algorithm significantly improves localization performance whether the training and testing condition setup are the same or not,and is more robust to noise and reverberation.

A FAST SEARCH METHOD OF STEERED RESPONSE POWER WITH SMALL-APERTURE MICROPHONE ARRAY FOR SOUND SOURCE LOCALIZATION

The Steered Response Power(SRP)method works well for sound source localization in noisy and reverberant environment.However,the large computation complexity limits its practical application.In this paper,a fast SRP search method is proposed to reduce the computational complexity using small-aperture microphone array.The proposed method inspired by the SRP spatial spectrum includes two steps:first,the proposed method estimates the azimuth of the sound source roughly and determines whether the sound source is in far field or near field;then,different fine searching operations are performed according to the sound source being in far field or near field.Experiments both in simulation environments and real environments have been performed to compare the localization accuracy and computation complexity of the proposed method with those of the conventional SRP-PHAT algorithm.The results show that,the proposed method has a comparative accuracy with the conventional SRP algorithm,and achieves a reduction of 93.62%in computation complexity compared to the conventional SRP algorithm.

Microphone Array-Based Sound Source Localization Using Convolutional Residual Network

Microphone array-based sound source localization(SSL)is widely used in a variety of occasions such as video conferencing,robotic hearing,speech enhancement,speech recognition and so on.The traditional SSL methods cannot achieve satisfactory performance in adverse noisy and reverberant environments.In order to improve localization performance,a novel SSL algorithm using convolutional residual network(CRN)is proposed in this paper.The spatial features including time difference of arrivals(TDOAs)between microphone pairs and steered response power-phase transform(SRPPHAT)spatial spectrum are extracted in each Gammatone sub-band.The spatial features of different sub-bands with a frame are combine into a feature matrix as the input of CRN.The proposed algorithm employ CRN to fuse the spatial features.Since the CRN introduces the residual structure on the basis of the convolutional network,it reduce the difficulty of training procedure and accelerate the convergence of the model.A CRN model is learned from the training data in various reverberation and noise environments to establish the mapping regularity between the input feature and the sound azimuth.Through simulation verification,compared with the methods using traditional deep neural network,the proposed algorithm can achieve a better localization performance in SSL task,and provide better generalization capacity to untrained noise and reverberation.

Method for measuring the low-frequency sound power from a complex sound source based on sound-field correction in a non-anechoic tank

Similar to air reverberation chambers, non-anechoic water tanks are important acoustic measurement devices that can be used to measure the sound power radiated from complex underwater sound sources using diffusion field theory. However,the problem of the poor applicability of low-frequency measurements in these tanks has not yet been solved. Therefore,we propose a low-frequency acoustic measurement method based on sound-field correction(SFC) in an enclosed space that effectively solves the problem of measuring the sound power from complex sound sources below the Schroeder cutoff frequency in a non-anechoic tank. Using normal mode theory, the transfer relationship between the mean-square sound pressure in an underwater enclosed space and the free-field sound power of the sound source is established, and this is regarded as a correction term for the sound field between this enclosed space and the free field. This correction term can be obtained based on previous measurements of a known sound source. This term can then be used to correct the mean-square sound pressure excited by any sound source to be tested in this enclosed space and equivalently obtain its free-field sound power. Experiments were carried out in a non-anechoic water tank(9.0 m × 3.1 m × 1.7 m) to confirm the validity of the SFC method. Through measurements with a spherical sound source(whose free-field radiation characteristics are known),the correction term of the sound field between this water tank and the free field was obtained. On this basis, the sound power radiated from a cylindrical shell model under the action of mechanical excitation was measured. The measurement results were found to have a maximum deviation of 2.9 d B from the free-field results. These results show that the SFC method has good applicability in the frequency band above the first-order resonant frequency in a non-anechoic tank. This greatly expands the potential low-frequency applications of non-anechoic tanks.

An Improved Time-Domain Inverse Technique for Localization and Quantifcation of Rotating Sound Sources

The time-domain inverse technique based on the time-domain rotating equivalent source method has been proposed to localize and quantify rotating sound sources. However, this technique encounters two problems to be addressed: one is the time-consuming process of solving the transcendental equation at each time step, and the other is the difculty of controlling the instability problem due to the time-varying transfer matrix. In view of that, an improved technique is proposed in this paper to resolve these two problems. In the improved technique, a de-Dopplerization method in the time-domain rotating reference frame is frst applied to eliminate the Doppler efect caused by the source rotation in the measured pressure signals, and then the restored pressure signals without the Doppler efect are used as the inputs of the time-domain stationary equivalent source method to locate and quantify sound sources. Compared with the original technique, the improved technique can avoid solving the transcendental equation at each time step, and facilitate the treatment of the instability problem because the transfer matrix does not change with time. Numerical simulation and experimental results show that the improved technique can eliminate the Doppler efect efectively, and then localize and quantify the rotating nonstationary or broadband sources accurately. The results also demonstrate that the improved technique can guarantee a more stable reconstruction and compute more efciently than the original one.

SOUND SOURCE LOCALIZATION OF DIGITAL HEARING AIDS USING WAVELET BASED MULTIVARIATE STATISTICAL METHOD

The letter proposed a sound source localization method of digital hearing aids using wavelet based multivariate statistics with the Generalized Cross Correlation (GCC) algorithm. Haar wavelet is used to decompose GCC sequences and extract four wavelet characteristics. And then, Hotelling T2 statistical method is used to fuse the four wavelet characteristics. The statistical value is used to judge the number of sound sources and obtain corresponding time delay estimation which is used to localize the position of sound source. The experimental results show that the proposed method has better robustness in an environment with severe noise and reverberation. Meanwhile, the complexity of al-gorithm is moderate, which is available for sound source localization of hearing aids.

Simulating Underwater Plasma Sound Sources to Evaluate Focusing Performance and Analyze Errors

Focused underwater plasma sound sources are being applied in more and more fields. Focusing performance is one of the most important factors determining transmission distance and peak values of the pulsed sound waves. The sound source’s components and focusing mechanism were all analyzed. A model was built in 3D Max and wave strength was measured on the simulation platform. Error analysis was fully integrated into the model so that effects on sound focusing performance of processing-errors and installation-errors could be studied. Based on what was practical, ways to limit the errors were proposed. The results of the error analysis should guide the design, machining, placement, debugging and application of underwater plasma sound sources.

Measurement of Aerodynamic Sound Source around a Circular Cylinder by Particle Image Velocimetry

The purpose of this paper is to propose a sound localization method as an alternative of the time-resolved particle image velocimetry (PIV) system for detecting the aerodynamic sound source of a circular cylinder in a uniform flow. The sound source intensity of a circular cylinder in a uniform flow is evaluated by measuring the time-derivative of instantaneous velocity field in the flow field using a pair of planar PIV system. It allows the visualization of the sound source intensity distribution, which is the time-derivative of the vector product of vorticity and velocity. The experimental results indicate that the aerodynamic sound is generated from the separation point and the velocity fluctuation in the separating shear layer from the circular cylinder. These results agree qualitatively with the previous findings from experiment and numerical simulation, which supports the validity of the present experimental method for evaluating the sound source intensity distribution.

Investigation on Major Noise Sources of Vehicle Diesel Engine Based on Sound Intensity Measurement

In order to reduce the noise and vibration of the diesel engine,it is crucial to exactly identify the engine noise source character.Based on "two-microphone" method,the sound intensity measurement of a vehicle four-stroke diesel engine was carried out in a hemi-anechoic chamber.Then the sound intensity contour maps were obtained from the measurement results and the main noise components of different frequencies on all the measurement surfaces were picked out to construct contour maps.By analysizing the relationship between the characteristics of contour maps and the space distribution of the engine compartment,the major sources of the exterior radiation noise of the diesel engine were identified.The results provided a creditable basis for improving the noise performance of the engine in the next phase.

Sound Source Measurement of a Semi-Circular Cylinder in a Uniform Flow by Particle Image Velocimetry

In this paper, the measurement of an aerodynamic sound source for a semi-circular cylinder in a uniform flow is described using Particle Image Velocimetry (PIV). This experimental technique is based on vortex sound theory, where the time derivative of vorticity is evaluated with the aid of two sets of standard PIV systems. The experimental results indicate that the sound source for the semi-circular cylinder is located around the shear layer near the edge of the semi-circular cylinder. The sound source intensity and the area are reduced in the semi-circular cylinder compared with those of a circular cylinder. This result indicates that the aerodynamic sound of the semi- circular cylinder is smaller than that of the circular cylinder, which supports the microphone measurement result.

Development of a System for Location Finding of Low-Level Sound Source by Using Human Acoustic System with Stochastic Resonance

The purpose of this study is to develop a system that enables location finding of a small sound. The location finding of a small sound has some difficulties such as high computational costs or disturbances from the ambient noises and reflected waves. The proposed system is composed of a biologically-inspired system which uses a hearing mechanism based on the human ear and a mechanism for perceiving weak signals that uses stochastic resonance. The location finding mechanism in the proposed system is based on the time-lag detecting architecture. On the other hand, the stochastic resonance mechanism can pick up the small sound source in the ambient noises. Using this proposed system, we implemented the location finding of small sounds through numerical simulations and hardware experiments. Good results were obtained for the small sound source location finding.

An Analytic Model of Subminiature Auditory Sensation System for Sound Source Localization

It is reported that some types of insects have a remarkable ability to detect the direction of an incident sound even though its acoustic sensory organs are in very close proximity each other. Maybe the ears are jointed by a cuticular structure with which the separated motions can be coupled mechanically and thus be magnified. In this paper, a detailed model is setup to describe the principle of this type of localization using a mechanical coupled structure. The transfer functions and the responses of the model in terms of time and frequency are analyzed to describe the mechanism of its ability of directional hearing. This analytical model provides a method to design the experimental model for the predetermined incident sound pressure, and the analysis of this model shows that this structure have the ability to determine the direction of the incident stimulus.

An Intelligent Robot based on Sound Source Localization and Ultrasound Distance Detection

In both industrial and research areas of electronic engineering,Sound Source Localization for robot control has always been an interesting subject to be further studied.Under some dangerous situation,especially when a special driver is required to implement a particular task,the device should be able to combine robotics control technology with Sound Source Localization,and take actions according to the different response patterns.In this research project,a multifunc-tional model driver,named "Mobile Island",has been designed and built up by integrating the Emulator 8051 micro-controller,Intel 8255 interfaces,some components and other necessary devices.The intelligent Mobile Island imple-mented by C language programs can operate under three control modes.In the sound control Mode 1,the model driver can detect and track a target by Sound Source Localization and then turn and move toward the destination.In the keypad control Mode 2,it can be controlled by a manual keypad.In the free run Mode 3,Mobile Island can move and turn by itself.When finding an object in front,it will turn away before moving forward again,so that it can avoid crashing on the obstacle.

Modeling the Underwater Acoustic Field Excited by an Airborne Rapidly Moving Source Using Wavenumber Integration

It is complicated to model the acoustic field in stratified ocean for airborne aircraft,due to high speed of the source and air-to-water sound transmission.To our knowledge,there are very few papers in the open literature dealing with this complicated problem;but,in our opinion,they all require great amount of computation.We now propose a different method that requires much less computation.We improve the wavenumber integration method to model the received temporal signal for a moving source in stratified ocean and sum up in a concise form the core of our paper as follows:(A) Eq.(11) can be calculated by means of fast Chirp Z transform and the signals at all N time points are generated simultaneously;(B) direct numerical evaluation of the wavenumber integral in Eq.(4) produces large numerical errors;so it is necessary to shift the integration slightly below the real axis;(C) we compare the computation cost of direct calculation method with that of our fast calculation method;from the results presented in table 1,we can see that the fast calculation method consumes much less computation time,particularly for long duration signals;(D) for an airborne rapidly moving source,we compute the Doppler-shifted signals in shallow water and analyze their short-time Fourier transform;from Fig.1b,we can see that the received signals have multiple frequency components for a tonal source due to source motion and that each component corresponds to an arrival path.