Model experiments on acoustic tomography of the nonlinear parameter

It is shown experimentally that the scattering of sound by sound can be observed outside the region of interaction between primary waves and that it carries information on the spatial distribution of the scatterer’s nonlinear characteristics. Based on this effect, a simple method for reconstructing the distribution of the nonlinear parameter by using complex-modulated broadband primary signals and a small number of transmit and receive transducers is implemented. The possibility of reconstructing a two-dimensional distribution of a nonlinear parameter inhomogeneity with the use of only three transducers (two transmitters and one receiver) without additional scanning in the aspect angle is demonstrated. This simple scheme loses information on the low-frequency components of the scatterer’s spatial spectrum and reconstructs the distribution of only the spatial variations of the nonlinear parameter.     

Localization of multiple sound sources with two microphones

This paper presents a two-microphone technique for localization of multiple sound sources. Its fundamental structure is adopted from a binaural signal-processing scheme employed in biological systems for the localization of sources using interaural time differences (ITD). The two input signals are transformed to the frequency domain and analyzed for coincidences along left/right-channel delay-line pairs. The coincidence information is enhanced by a nonlinear operation followed by a temporal integration. The azimuths of the sound sources are estimated by integrating the coincidence locations across the broadband of frequencies in speech signals (the "direct" method). Further improvement is achieved by using a novel "stencil" filter pattern recognition procedure. This includes coincidences due to phase delays of greater than 2pi, which are generally regarded as ambiguous information. It is demonstrated that the stencil method can greatly enhance localization of lateral sources over the direct method. Also discussed and analyzed are two limitations involved in both methods, namely missed and artifactual sound sources. Anechoic chamber tests as well as computer simulation experiments showed that the signal-processing system generally worked well in detecting the spatial azimuths of four or six simultaneously competing sound sources.     

Experimental studies of the characteristic features of vector receivers in application to ocean tomography

Experimental data on the angular structure of the sound field on the shelf of the Sea of Japan are discussed. The data are obtained by using the vector-phase processing of complex signals in the form of M-sequences with a central frequency of 2500 Hz. With the parameters of the water bulk being monitored, an unambiguous relation is established between the variability of the temperature regime on a fixed acoustic path and the arrival times and angles. It is shown that the use of vector receivers in ocean tomography provides an additional independent parameter of the pulse response of the underwater waveguide and leads to an increase in the efficiency of solving the problem of reconstructing the hydrophysical fields from the data of acoustic sensing.     

Interference structure of low-frequency reverberation signals in shallow water

Using numerical simulation, an analysis was conducted of the interference structure of a bottomscattered sound field generated by a wideband point source in shallow water under winter and summer conditions. The scattered signals were received from the place where the source was located and were subjected to Fourier transform with a sliding window. The paper demonstrates the possibility of estimating the waveguide invariant for backscattered signals when processing the sound intensity distributions in wide frequency and distance ranges up to the scattering area. A technique is proposed for reconstructing the twodimensional field of internal waves using variations of the interference pattern of reverberation signals. The influence of wind surface waves on the degree of interference band contrast is illustrated.     

Least squares approach in wavenumber domain for sound field recording and reproduction using multiple parallel linear arrays

A novel signal processing method is proposed for sound field recording and reproduction using multiple parallel linear microphone and loudspeaker arrays. In sound field recording and reproduction, the problem is how to calculate the transfer filters that transform the signals recorded by microphones into the driving signals of the loudspeakers. The proposed method is based on the spatial Fourier transform in the horizontal angle combined with the least squares (LS) approach in the elevation angle. In the proposed method, the signals recorded by each linear microphone array and those that drive each loudspeaker array are decomposed into the wavenumber domain by the spatial Fourier transform in the horizontal direction. The transfer filters are then calculated by the LS approach in the wavenumber domain. As a result, the size of the matrix of each transfer function in the wavenumber domain is much smaller than that of the conventional LS approach in the temporal frequency domain (LSTF), and well-conditioned stable transfer filters can be obtained with low computational cost without regularization. Computer simulation results show that the proposed method reconstructed a sound field around the control points as accurately as the conventional LSTF.     

浅海单矢量传感器宽带声源三维定位仿真

针对水平不变的浅海波导中单矢量传感器对低频宽带声源的三维被动定位问题,首先利用平均声强器估计声源方位；其次,通过分离简正模的声压和水平振速分量联合处理获得运动声源相对速度,进一步基于垂直声能流中简正模相干项特征频率不变性以及长时间窗口中多快拍信号的统一处理,建立WARPING变换频谱作为代价函数的搜索处理器,估计该段信号的初始距离,进而获得各时刻声源距离,所提出的方法避免了对拷贝声场和引导声源的依赖；最后,利用多阶简正模相干项与非相干项能量模基处理方法,当声场中存在三阶以上简正模时,可对声源深度进行匹配估计。仿真分析表明,单个矢量传感器能够给出声源的方位、距离及深度估计结果。     

Directional reception of broadband signals by means of ultrasonic transducer arrays

This paper presents a new approach to the theory and technique of ultrasonic transducer arrays, which makes it possible to obtain their directional beam patterns for receiving broadband acoustical signals.The general formulae were derived for the beam pattern of the array, and as an example the beam patterns of the end-fire array have been computed and plotted, for signals of different correlation function.For experimental verification of the theory the simple processor solution is proposed. This technique is based on spatial/temporal signal processing for beamforming of the directional response of an end-fire array.     

Surrogate analysis for detecting nonlinear dynamics in normal vowels.

Normal vowels are known to have irregularities in the pitch-to-pitch variation which is quite important for speech signals to be perceived as natural human sound. Such pitch-to-pitch variation of vowels is studied in the light of nonlinear dynamics. For the analysis, five normal vowels recorded from three male and two female subjects are exploited, where the vowel signals are shown to have normal levels of the pitch-to-pitch variation. First, by the false nearest-neighbor analysis, nonlinear dynamics of the vowels are shown to be well analyzed by using a relatively low-dimensional reconstructing dimension of 4 < or = d < or = 7. Then, we further studied nonlinear dynamics of the vowels by spike-and-wave surrogate analysis. The results imply that there exists nonlinear dynamical correlation between one pitch-waveform pattern to another in the vowel signals. On the basis of the analysis results, applicability of the nonlinear prediction technique to vowel synthesis is discussed.     

用于浅海有源声呐目标深度估计的匹配相位处理EI北大核心CSCD

针对浅海波导中有源声呐目标深度估计问题,提出了通过单个声源发射低频宽带信号,垂直双接收水听器接收目标回波,利用宽带目标回波比值的相位特征进行匹配相位处理的方法,不仅消除了目标散射特性对有源声呐目标深度估计的影响,而且仅需计算单程的信道传递函数,运算量小,有利于实时处理的实现需求。首先通过理论推导,定义了一个隐含目标深度、且与目标散射特性无关的声场特征——单程传播向量宽带干涉结构,在Pekeris波导条件下仿真分析了简正波阶数与单程传播向量宽带干涉结构的幅度和相位的关系,发现选取较多的简正波阶数贡献的单程传播向量宽带干涉结构进行匹配相位处理可以提高目标深度估计的性能。进一步仿真分析表明,在发射信号选取一定的时间长度和带宽的条件下,利用全部阶简正波且信噪比大于-10 dB时,方法的深度估计误差在5 m之内。最后分析了信号时间长度和处理带宽对有源声呐目标深度估计性能的影响,以及海底声速、海深和声速剖面失配时方法的鲁棒性。     

Experimental study of mode selection in shallow-water sea

The possibilities of matching low-frequency underwater sound pulses to the parameters of an oceanic waveguide are considered. The objective is to optimize the system of few-mode tomographic observation in a shallow-water sea. Experimental data are analyzed for two methods of selecting the low-frequency fewmode pulses propagating in a shallow-water sea. The first method excites probe pulses by vertically elongated arrays, with spatial filtering after vertical or horizontal arrays receive the pulses. The second method is based on exciting broadband signals with linear frequency modulation by a single transmitter. The selection of the few-mode signal is performed by time strobing the signals at the output of the matched filter after a horizontal array receives the pulses. The distance between the sound sources and receiving systems varied from 10 to 300 km.     

Quantum ratchet driven by broadband perturbation

Motion of an atomic ensemble trapped into a deep optical lattice is considered. We propose a novel approach to construct an atomic ratchet by superimposing two additional lattices whose amplitudes are small and subjected to broadband modulation. The broadband modulation is modeled by means of harmonic noise. Directed atomic transport occurs with the properly chosen phase shift between the signals modulating the amplitudes of the additional lattices. It is shown that efficiency of the ratchet depends non-monotonously on the parameter determining the spectral width of modulating signals.     

Spatial-temporal correlation of a diffuse sound field

The spatial correlation function of the sound in a diffuse field is a quantity widely used in many reverbrant room acoustic applications. Although results for the spatial and temporal correlation for pure-tone and band-limited diffuse fields have already been developed, these have not been generalized for other signal types. This work presents a generalized derivation of the diffuse field spatial-temporal correlation which can be used for stationary random signals with given power spectral density. It is shown that the spatial-temporal correlation depends entirely on the temporal correlation of the signal exiting the diffuse field, or alternatively on its power spectral density. A simulation using the plane wave model is presented for tonal and broadband diffuse sound fields.     

On the possibility of using acoustic reverberation for remote sensing of ocean dynamics

The two-point correlation function of diffuse noise fields produced by distributed random sound sources carries useful information on the medium of sound propagation. Such information can be used for performing passive acoustic tomography of the ocean. In a number of cases that are important for practice, the noise field in the ocean is predominated by contributions of individual point sources. Here, a theoretical study is presented on the possibility of determining the sound speed and current velocity in the water column by the correlation processing of reverberation signals measured by two vertical receiving arrays. In other words, we study the possibility of replacing the diffuse noise produced by a great number of delta-correlated sources by waves generated by a localized source and scattered at the rough surface and bottom of the ocean for sensing the medium. The correlation function of scattered waves is calculated by using the method of small perturbations. It is shown that the correlation processing of the scattered waves offers an opportunity of measuring the acoustic nonreciprocity and reconstructing the field of sound speed in the fluid, without using any acoustiLc transceivers.     

The nonlinear propagation of acoustic waves in a viscoelastic medium containing cylindrical micropores

Based on an equivalent medium approach, this paper presents a model describing the nonlinear propagation of acoustic waves in a viscoelastic medium containing cylindrical micropores. The influences of pores' nonlinear oscillations on sound attenuation, sound dispersion and an equivalent acoustic nonlinearity parameter are discussed. The calculated results show that the attenuation increases with an increasing volume fraction of micropores. The peak of sound velocity and attenuation occurs at the resonant frequency of the micropores while the peak of the equivalent acoustic nonlinearity parameter occurs at the half of the resonant frequency of the micropores. Furthermore, multiple scattering has been taken into account, which leads to a modification to the effective wave number in the equivalent medium approach. We find that these linear and nonlinear acoustic parameters need to be corrected when the volume fraction of micropores is larger than 0.1%.     

深远海线性内波条件下声场时间相关性的空间分布

深海声场特定的干涉结构导致其时间相关性的空间起伏,研究这种空间特性可以为水声信号的探测与处理提供重要参考。利用抛物方程声场仿真模型,联合Monte-Carlo数值方法计算分析了深远海线性内波条件下声场时间相关性的空间分布特性。与现有的研究相比,给出了时间相关性的距离和深度起伏特征。结果表明,当接收达到一定距离,声场时间相关性的空间分布具有与声场干涉条纹类似的结构,声场干涉越强,时间相关性越好。此外,声源频率和声速标准差的变化会引起时间相关性空间分布规律的改变,且会聚区传播模式下的改变强于深海声道传播模式。      

A spatial correlation method for reconstructing blood flow velocity vector in the nonlinear tomography

A method for reconstructing the total blood flow velocity vector distribution is proposed. Experimental data are obtained with the help of the second-order nonlinearity parameter tomography scheme that employs wideband coded signals and a small number of transducers. The data are processed using the selective matched filtering technique in combination with spatial correlation of fragments of the speckle structure that appears in the process of reconstruction of the spatial nonlinearity parameter distribution. Results of numerical simulations are presented.     

球形传声器阵列下基于主导声源检测MUSIC群时延算法的多声源定位*

针对混响环境中,多径效应、散射、衍射等原因导致声源定位失败或分辨能力不足的现象,提出一种基于主导声源检测MUSIC群时延的邻近多声源定位方法。该方法采用球形传声器阵列,相比平面阵列可以捕获3D声场信息,利用球谐域下信号的频率分量与角度分量解耦的优势,从而可直接利用频率平滑技术处理宽带语声信号而不需要构造聚焦矩阵,并在球谐域下通过设置阈值对一组时频段进行主导声源检测,从而选择出包含直达声的一组时频块来构造MUSIC群时延空间谱。上述举措在提升波达方向估计在高混响环境下定位鲁棒性的同时,也提高了多个邻近声源的分辨能力。仿真实验结果表明,所提出的主导声源检测MUSIC群时延算法,在高混响和低信噪比条件下,仍具有更好的定位精度与更优的邻近多声源分辨效果。     

Broadband sound generation by confined pulsating jets in a mechanical model of the human larynx

Experiments were performed to study the production of broadband sound in confined pulsating jets through orifices with a time-varying area. The goal was to better understand broadband sound generation at the human glottis during voicing. The broadband component was extracted from measured sound signals by the elimination of the periodic component through ensemble averaging. Comparisons were made between the probability density functions of the broadband sound in pulsating jets and of comparable stationary jets. The results indicate that the quasi-steady approximation may be valid for the broadband component when the turbulence is well established and the turbulence kinetic energy is comparatively large. A wavelet analysis of the broadband sound showed that random sound production was modulated at the driving frequency. Two distinct sound production peaks were observed during one cycle, presumably associated firstly with jet formation and secondly with flow deceleration during orifice closing. Most high-frequency sound was produced during the closing phase. Deviations from quasi-steady behavior were observed. As the driving frequency increased, sound production during the opening phase was reduced, possibly due to the shorter time available for turbulence to develop. These results may be useful for better quality voice synthesis.     

Modal analysis of broadband acoustic receptions at 3515-km range in the North Pacific using short-time Fourier techniques

In 1995-1996 the Acoustic Thermometry of Ocean Climate (ATOC) experiment provided an opportunity to study long-range broadband transmissions over a series of months using mode-resolving vertical arrays. A 75-Hz source off the California coast transmitted broadband pulses to receiving arrays in the North Pacific, located at ranges of 3515 and 5171 km. This paper develops a short-time Fourier transform (STFT) processor for estimating the signals propagating in the lowest modes of the ocean waveguide and applies it to analyze data from the ATOC experiment. The STFT provides a convenient framework for examining processing issues associated with broadband signals. In particular, this paper discusses the required frequency resolution for mode estimation, analyzes the broadband performance of two standard modal beamforming algorithms, and explores the time/frequency tradeoffs inherent in broadband mode processing. Short-time Fourier analysis of the ATOC receptions at 3515 km reveals a complicated arrival structure in modes 1-10. This structure is characterized by frequency-selective fading and a high degree of temporal variability. At this range the first ten modes have equal average powers, and the magnitude-squared coherence between the modes is effectively zero. The coherence times of the peaks in the STFT mode estimates are on the order of 5.5 min. An analysis of mean arrival times yields modal dispersion curves and indicates that there are statistically significant shifts in travel time over 5 months of ATOC transmissions.     

一种双正交心音小波的构造方法

为了提高小波分析在心音信号处理中的性能, 在分析小波构造理论的基础上, 构造了一种专门用于心音信号处理的小波基. 首先提出一种构造滤波器长度为偶数的紧支撑双正交小波的一般方法; 然后根据心音信号的特点, 讨论心音小波的构造原则和一种基于心音 小波族的心音信号合成模型, 并且在此基础上构造出心音小 波. 为了突出使用心音小波处理心音信号的先进性和实用性, 对心音小波进行了比较全面的理论和数值仿真分析. 实验结果表明, 相比常用的db, bior系列小波, 运用心音小波对心音信号进行处理, 能够获得更好的去噪效果、 更精确的心音分类信息以及更小的重构误差率, 为心音特征提取和身份识别的深入研究提供了一种新方法, 在表征心音个体特征的细节方面具有积极的意义. 本文根据应用对象设计专用小波的方法也为工程应用中小波基的选择提供了一种新途径. 关键词： 双正交小波 心音小波 构造方法 心音合成模型