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.

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.

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.

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.

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.

Target localization and parameters estimation by sonar system with explosions as underwater sound sources

Underwater target localization and parameters(azimuth and range) estimation by the method of utilizing explosions as underwater sound sources are described in this paper.The narrow beam reverberation model of the target echo signal is researched to estimate the target azimuth in reverberation background.Estimation errors of target azimuth and range are studied and proved to approximately meet Gauss distribution.Then the variance formula of target range error is deduced.Simulation experiments are applied to research the target range error and its standard deviation,and a series of measures to improve the estimation accuracy of target range are proposed.It is confirmed by the data processing results of simulations and lake experiments that the proposed method can accurately locate underwater target at a long distance on the condition of a certain underwater explosion range error.

Research on acoustic holography method for identification of sound sources

This paper presents the method named acoustic holography which can be used to identify noise sources. A new formula of holography reconstruction is obtained, based on the Kirchhoff integral formula. Some simulating tests are carried out using the new formula. The comparison with other reconstruction formulas proves that the new formula is more effective. By using acoustic holography method, some interesting results about the noise of a vehicle are shown. The results proves that acoustic holography is an effective method for the identification of the complex noise sources.

Distributed sound source localization algorithm with sound velocity calibration in windy environments

A new sound source localization method with sound speed compensation is proposed to reduce the wind influence on the performance of conventional TDOA （Time Difference of Arrival） algorithms. First, the sound speed is described as a set of functions of the unknown source location, to approximate the acoustic velocity field distribution in the wind field. Then, they are introduced into the TDOA algorithm, to construct nonlinear equations. Finally, the particle swarm optimization algorithm is used to estimate the source location. The simulation results show that the proposed algorithm can significantly improve the localization accuracy for different wind velocities, source locations and test area sizes. The experimental results show that the proposed method can reduce localization errors to about 40% of the original error in a four nodes localization system.

A head-related transfer function model for fast synthesizing multiple virtual sound sources

A head-related transfer function （HRTF） model for fast and real-time synthesizing multiple virtual sound sources is proposed. A head-related impulse response （HRIR, time- domain version of HRTF） is first decomposed by a two-level wavelet packet and then represented by a model composed of subband filters and reconstruction filters. The coefficients of the subband filters are the zero interpolation of the wavelet coefficients of the HRIR. The coefficients of the reconstruction filters can be calculated from the wavelet function. The model is simplified by applying a threshold method to reduce the wavelet coefficients. The calculated results indicate that for a model with 30 wavelet coefficients, the error of reconstructed HRIR is about 1%. And the result of a psychoacoustic test shows that a model with 35 wavelet coefficients is perceptually indistinguishable from the original HRIR. When multiple virtual sound sources are synthesized simultaneously, the computational cost of the proposed model is much less than the traditional HRTF filters.

Design and validation on a multiple sound source fast-measurement system of near-field head-related transfer functions

Near-field head-related transfer functions （HRTFs） are essential to scientific re- searches of binaural hearing and practical applications of virtual auditory display. High ef- ficiency, accuracy and repeatability are required in a near-field HRTF measurement. Hence, there is no reference which intents on solving the measuring difficulties of near-field HRTF for human subjects. In present work, an efficient near-field HRTF measurement system based on computer control is designed and implemented, and a fast calibration method for the system is proposed to first solve the measurement of near-field HRTF for human subjects. The efficiency of measurement is enhanced by a comprehensive design on the acoustic, electronic and mechanical parts of the system. And the accuracy and repeatability of the measurement are greatly im- proved by carefully calibrating the positions of sound source, subject and binaural microphones. This system is suitable for near-field HRTF measurement at various source distances within 1.0 m, for both human subject and artificial head. The time costs of HRTF measurement at a single sound source distance and full directions has been reduced to less than 20 minutes. The measurement results indicate that the accuracy of the system satisfies the actual requirements. The system is applicable to scientific research and can be used to establish an individualized near-field HRTF database for human subjects.

Numerical analysis of 3-D sound propagation in lined ducts with circumferential mode sound sources at inlet

Numerical analysis of three-dimensional sound propagation in soft-soft or soft-hard circular ducts with circumferential and axial modes of sound sources at the inlet has been carried out. In this paper , the numerical method and the samples are offered and the effects of circumferential and axial modes on numerical results are discussed in detail .

A study on the random array generation method for moving sound source identification

Based on the problem that the generating method of random array structure is inefficient, a method is proposed to generate the random target arrays by using coaxial circu- lar array in the polar coordinates in the premise that the array angular resolution of source identification is guaranteed. According to the principle of moving sound source identification, this work deduces the basic non-equidistance coaxial circular rings＇ radius, and generates target random arrays which were suitable for moving sound source identification through array partitioning, condition filtering in the polar coordinates and simulation evaluation. Finally, numerical simulation and moving car sound source identification test have been done. The analytical results show that using this method to generate random array is effective. Compared with the traditional regular arrays, the target random array has more accurate moving sound source identification performance.

THE COMPUTATION OF PARTIAL COHERENCE FUNCTION APPLIED FOR SOUND SOURCE IDENTIFICATION

In this paper, a new method is applied to get the computation formula of partial coherence function. The main attention is paid to the computation formula of the partial coherence function with three and four signals. The advantages of the method discussed in the paper are clear in physical meaning and easy to compute at the end of the paper,the application of the method to the identification of an air compressor noise source is presented and the results are satisfactory.

Application of wavelet scale correlation filtering and its improved algorithm in signal processing with a spark sound source

It is seriously interfered by ship noise when analyzing and extracting broadband spark sound source signal. In the energy concentrated domain which is below 5 kHz, the traditional scale correlation filtering algorithm, which is based on adjacent-scale correlation, has limited anti-interference ability due to the low signal-to-noise ratio （SNR） and similar Lipschitz exponent characteristic of each other. However, because different frequency bands of the broadband electric spark signal have different noise interferences, the filtering algorithm based on adjacent-scale correlation is adapted to high SNR and small-scale high-frequency wavelet coefficients filtering; the filtering algorithm based on cross-scale correlation is adapted to low SNR and large-scale low-frequency wavelet coefficients filtering, and the threshold coefficient selection method had been corrected in the algorithm. It is shown that the filtering algorithm has a good filtering effect and extracts the broadband spark sound source signal effectively; it is applicable to broadband underwater acoustic signM processing in the presence of narrow-band strong interference background noise.

On electro-acoustic characteristics of a marine broadband sparker for seismic exploration

The construction of major marine infrastructure projects and the exploration and development of deep-sea mineral resources require fine imaging of seabed strata and structures.The highresolution marine seismic exploration based on a high broadband sparker source is an important approach to reveal seabed stratum and reservoir structure,and identify geohazard.To optimize the performance of sparker seismic source,we investigated the electro-acoustic characteristics of spark discharge under conditions of different charging voltages and electrode numbers.Results show that the sound source level increased with the increase of the charging voltage,whereas the main frequency decreased when the charging voltage increases.In addition,it was found that the charging capacitance had more obvious influence on the main frequency than the sound source level did.Although the load energy decreased with increasing electrode number,the sound source level still increased but the main frequency decreased.Meanwhile,the primary to bubble(P/B)ratio increased with the increase of the electrode number.To gain a deeper insight into the electro-acoustic characteristics,we investigate the relationship between sound source level and power peak,from which a good correlation was observed.A more practical statistical analysis on the rise rate of current was processed,and a perfect logarithmic function was derived.Furthermore,we found that the main frequency was most possibly subjected to the electrical energy,especially the charging energy per electrode.The results indicate that the charging energy per electrode less than 10 J could increase the main frequency to above 300 Hz.At last,the main frequency could be reduced to 20 Hz when the charging energy of a single-electrode discharge was enhanced to over 4 kJ.This study shall be helpful in developing a sparker seismic source and improving the performance for marine engineering exploration and geohazard assessment.

Step-by-Step Numerical Prediction of Aerodynamic Noise Generated by High Speed Trains

In this paper,the unsteady flow around a high-speed train is numerically simulated by detached eddy simulation method(DES),and the far-field noise is predicted using the Ffowcs Williams-Hawkings(FW-H)acoustic model.The reliability of the numerical calculation is verified by wind tunnel experiments.The superposition relationship between the far-field radiated noise of the local aerodynamic noise sources of the high-speed train and the whole noise source is analyzed.Since the aerodynamic noise of high-speed trains is derived from its different components,a stepwise calculation method is proposed to predict the aerodynamic noise of high-speed trains.The results show that the local noise sources of high-speed trains and the whole noise source conform to the principle of sound source energy superposition.Using the head,middle and tail cars of the high-speed train as noise sources,different numerical models are established to obtain the far-field radiated noise of each aerodynamic noise source.The far-field total noise of high-speed trains is predicted using sound source superposition.A step-by-step calculation of each local aerodynamic noise source is used to obtain the superimposed value of the far-field noise.This is consistent with the far-field noise of the whole train model’s aerodynamic noise.The averaged sound pressure level of the far-field longitudinal noise measurement points differs by 1.92 dBA.The step-by-step numerical prediction method of aerodynamic noise of high-speed trains can provide a reference for the numerical prediction of aerodynamic noise generated by long marshalling high-speed trains.

A Complete Analysis of Clarity(C50)Using I-SIMPA to Maintain Ideal Conditions in an Acoustic Chamber

In any closed environment considered,it can be seen that the acoustic parameters are inherently not constant over the entire area considered.In a closed environment,it is ideally preferred to maintain the acoustic parameters as constant so that there exists better quality of sound leading to better auditory perception with respect to the audience.Practically,some of the acoustic parameters like reverberation time and clarity do not strictly pertain to the pattern obtained theoretically.In this paper,simulations are carried out using I-SIMPA under different values of Sound Transmission Class(STC),source position,distribution and the chamber dimensions and provides an insight into the behaviour of these acoustic parameters and the appropriate values that have to be infused into the system to build the chamber.The acoustic environment is modelled keeping an actual closed room in mind and testing is done with respect to different values of surface absorption coefficients(practically indicated using Sound Transmission Class)and dimensions to determine ideal conditions.