Determining direction to a source of correlated signals,using adaptive antenna arrays

Gor'kii State University. Translated from Izvestiya Vysshikh Zavedenii, Radiofizika, Vol. 33, No. 6, pp. 719–725, June, 1990.     

空气中快速运动声源水下声场的波数积分模型

本文采用二维波数积分，对空气中高速运动声源激发的水下声场进行建模。针对二维波数积分计算声场时域解计算量大的问题，提出一种快速计算方法。用本文提出的方法，对深海和浅海情况下，空气中高速运动单频点声源激发的水下声场进行了计算和仿真。计算结果表明：在深海，水下接收信号的幅度和瞬时频率随时间发生变化；接收器深度、接收器与声源运动轨迹的最小距离对接收信号的变化快慢有较大影响，而声源高度的影响较小；在浅海中，接收信号呈现快速的幅度起伏，明显的多普勒频移和大的频率展宽效应。与简正波方法相比，本文方法主要适用于近场计算，而简正波方法适用于远场。另外，当声源频率较高时，二维波数积分方法的计算量将迅速增大。     

Effects of sound source location and direction on acoustic parameters in Japanese churches

In 1965, the Catholic Church liturgy changed to allow priests to face the congregation. Whereas Church tradition, teaching, and participation have been much discussed with respect to priest orientation at Mass, the acoustical changes in this regard have not yet been examined scientifically. To discuss acoustic desired within churches, it is necessary to know the acoustical characteristics appropriate for each phase of the liturgy. In this study, acoustic measurements were taken at various source locations and directions using both old and new liturgies performed in Japanese churches. A directional loudspeaker was used as the source to provide vocal and organ acoustic fields, and impulse responses were measured. Various acoustical parameters such as reverberation time and early decay time were analyzed. The speech transmission index was higher for the new Catholic liturgy, suggesting that the change in liturgy has improved speech intelligibility. Moreover, the interaural cross-correlation coefficient and early lateral energy fraction were higher and lower, respectively, suggesting that the change in liturgy has made the apparent source width smaller.     

Non-line-of-sight sound source localization using matched-field processing

Acoustic diffraction allows sound to travel around opaque objects and therefore may allow beyond-line-of-sight sensing of remote sound sources. This paper reports simulated and experimental results for localizing sound sources based on fully shadowed microphone array measurements. The generic geometry includes a point source, a solid 90° wedge, and a receiving array that lies entirely in the shadow defined by the source location and the wedge. Source localization performance is assessed via matched-field (MF) ambiguity surfaces as a function of receiving array configuration, and received signal-to-noise ratio for the Bartlett and minimum variance distortionless (MVD) MF processors. Here, the sound propagation model is developed from a Green's function integral treatment. A simple 16 element line array of microphones is tested in three mutually orthogonal orientations. The experiments were conducted using an approximate 50-to-1-scaled tabletop model of a blind city-street intersection and produced ambiguity surfaces from source frequencies between 17.5 and 19 kHz that were incoherently summed. The experimental results suggest that a sound source may be localized by the MVD processor when using fully shadowed arrays that have significant aperture parallel to the edge of the wedge. However, this performance is reduced significantly for signal-to-noise ratios below 40 dB.     

On the angular error of intensity vector based direction of arrival estimation in reverberant sound fields

An acoustic vector sensor provides measurements of both the pressure and particle velocity of a sound field in which it is placed. These measurements are vectorial in nature and can be used for the purpose of source localization. A straightforward approach towards determining the direction of arrival (DOA) utilizes the acoustic intensity vector, which is the product of pressure and particle velocity. The accuracy of an intensity vector based DOA estimator in the presence of noise has been analyzed previously. In this paper, the effects of reverberation upon the accuracy of such a DOA estimator are examined. It is shown that particular realizations of reverberation differ from an ideal isotropically diffuse field, and induce an estimation bias which is dependent upon the room impulse responses (RIRs). The limited knowledge available pertaining the RIRs is expressed statistically by employing the diffuse qualities of reverberation to extend Polack's statistical RIR model. Expressions for evaluating the typical bias magnitude as well as its probability distribution are derived.     

Simulation of the vector wave field of a low-frequency sound source in the ocean: Some important features

An approach to the simulation of low frequency vector wave fields in stratified media (mainly in the ocean) is considered. The approach is characterized by an improved stability with respect to dividing the medium into many layers of arbitrary thickness. The model for the sound field of a point source is based on an integral representation of two-dimensional, cylindrically symmetric vector wave fields in inhomogeneous media, so that the contributions of all types of waves are included automatically. The model medium is subdivided into N horizontally homogeneous layers for which 4(N?1) equations are formulated to satisfy the boundary conditions between adjacent layers. The method of the generalized Schmidt matrix is used to obtain the coefficients of the equations; these coefficients are substituted into the expressions (of the Fourier-Bessel integral type) for the local parameters of the field. The latter are calculated according to the numerical procedure, and the results are used to model the distributions of the acoustic pressure and the horizontal and vertical components of the particle velocity in liquid and elastic media. The instability of the calculation procedure may result in a disagreement between the model and the exact solution. However, the disagreement is shown to occur mainly in models containing excessively thick layers. A way for improving the stability of the numerical model is suggested. The simulation results are compared with the exact analytical solution for the simplest example and with the results obtained according to the commonly used generalized matrix procedure (the benchmark problem). The examples of the practical application of the model for investigating more complex seismoacoustic wave fields in the ocean are presented.     

双边相关变换在估计水下噪声源波达方向中的应用

研究双边相关变换高分辨率波束成形去估计水下噪声源波达方向的方法和性能,提出了对水下运动目标进行精确定向的一种高精度定向算法,用分割频带中的能量最大频段的窄带定向结果作为初始角,宽带中的高频分割频带作为处理对象来获得较高的定向精度。仿真数据和实际数据表明,该算法是一种花费时间少、定向精度高、简便实用的宽带定向方法。这种高精度的测向算法为水下两基阵交会测距提供了保证。     

Analysis of errors when using the similar source method to estimate sound fields

The similar source method (SSM) uses an imaginary source distribution to estimate radiated and scattered sound fields. The error analyses in the present study are for possible numerical errors by SSM, solutions of the problem of singularity and non-unique solution by SSM and the effects of possible errors in measurements on the estimation of the sound field by SSM.     

Mathematically trivial control of sound using a parametric beam focusing source

By exploiting a case regarded as trivial, this paper presents global active noise control using a parametric beam focusing source (PBFS). As with a dipole model, one is used for a primary sound source and the other for a control sound source, the control effect for minimizing a total acoustic power depends on the distance between the two. When the distance becomes zero, the total acoustic power becomes null, hence nothing less than a trivial case. Because of the constraints in practice, there exist difficulties in placing a control source close enough to a primary source. However, by projecting a sound beam of a parametric array loudspeaker onto the target sound source (primary source), a virtual sound source may be created on the target sound source, thereby enabling the collocation of the sources. In order to further ensure feasibility of the trivial case, a PBFS is then introduced in an effort to meet the size of the two sources. Reflected sound wave of the PBFS, which is tantamount to the virtual sound source output, aims to suppress the primary sound. Finally, a numerical analysis as well as an experiment is conducted, verifying the validity of the proposed methodology.     

Passive ranging of a moving sound source using three-dimensional measured information

To solve the problem of passive ranging of a moving sound source,a method of estimating the initial distance of target is presented by using the measured inform...     

Tomographic reconstruction of sound fields in air using TV-holography

A vibrating sound source causes periodic pressure variations in the air. The pressure variations lead to corresponding variations in the refractive index of the air which can be measured using full field interferometric techniques like TV-holography. Each measurement maps a cross-section of the integrated sound fields. To obtain a complete map of the volume distribution of sound field, we record cross-sections from different directions. Using tomographical backprojection of these recordings, we reconstruct the amplitude and phase of the sound field in any plane of the volume.     

一种利用分布式传声器阵列的声源三维定位方法

为了提高噪声和混响条件下分布式传声器阵列进行声源定位的性能,提出一种利用空间稀疏性和压缩感知原理的声源三维定位方法。该方法首先通过两次离散余弦变换方式提取出声音信号特征,并用该特征来构建稀疏定位模型,以便能够综合利用语音信号的短时和长时特性,同时降低模型维数;然后利用在线字典学习技术动态调整字典,克服稀疏模型与实际信号之间的失配问题,增强稀疏定位模型的鲁棒性;进而提出一种改进的平滑l₀范数稀疏重构算法来进行声源位置解算,以提高低信噪比条件下的重构精度。仿真结果表明该方法不仅可以实现多目标定位,而且具有较强的抗噪声和抗混响能力.     

Three-dimensional sound source localization using distributed microphone arrays

To improve the performance of sound source localization based on distributed microphone arrays in noisy and reverberant environments,a sound source localization method was proposed.This method exploited the inherent spatial sparsity to convert the localization problem into a sparse recovery problem based on the compressive sensing(CS) theory.In this method two-step discrete cosine transform(DCT)-based feature extraction was utilized to cover both short-time and long-time properties of the signal and reduce the dimensions of the sparse model.Moreover,an online dictionary learning(DL) method was used to dynamically adjust the dictionary for matching the changes of audio signals,and then the sparse solution could better represent location estimations.In addition,we proposed an improved approximate l_0norm minimization algorithm to enhance reconstruction performance for sparse signals in low signal-noise ratio(SNR).The effectiveness of the proposed scheme is demonstrated by simulation results where the locations of multiple sources can be obtained in the noisy and reverberant conditions.     

使用同心多环阵提升声源定位鲁棒性*

房间混响带来的多径失真是影响声源波达方向估计精度的主要因素之一。应用于环形阵列的相干信号子空间方法可以降低相干反射声带来的不利影响。该方法对环形阵进行谐波展开,并利用环谐波域导向矢量的频率无关特性聚焦各频率下的空间相关矩阵。但单环阵列存在展开系数零点,这会导致严重的噪声放大而降低定位的鲁棒性。该文提出了一种环谐波域的最小模同心多环阵,来缓解系数零点处的噪声放大问题。设计了一套麦克风阵列系统,用于评估同心多环阵列定位的鲁棒性。仿真和实验结果均表明:与相同孔径和阵元数的单环阵相比,使用最小模准则设计的同心多环阵列可以显著提升混响场景下的声源定位的稳健性。     

利用等离子体声源测量浅海低频段水声信道特性

浅海水声信道的结构特征及其变化特性是浅海环境水下远程探测和通讯应用的基础.为了有效测量低频水声信道特性,我们在实验中采用了拖曳等离子体声源和垂直阵接收系统。通过实验测量与信道模型仿真输出的比较,对等离子体声源特性,特别是脉冲波形、空间指向性和电声转换效率,以及在信道中的传播特性等进行测量分析。结果表明:实验测量与信道模型仿真符合良好,实验中采用的等离子体声源的发射源级和拖曳姿态稳定,波形一致性好,可以满足走航式连续水下信道测量要求。     

利用残差网络分析普氏蹄蝠耳朵对恒频声源定向的影响

为研究恒频蝙蝠耳朵与空间定位的关系,利用深度学习算法和仿蝙蝠静态双耳接收器,分析蝙蝠耳朵对恒频声源定向的影响。首先根据普氏蹄蝠耳朵模型设计不同双耳夹角和间距的仿生双耳接收器,并从多个空间方位采集声源发射的不同频率的恒频声呐信号,然后提取双耳同步采集信号的时频图并归一化作为输入特征,最后利用残差网络实现声源定向。实验结果表明,静态双耳接收器对恒频声源的定向误差平均值基本保持在3.5°以下,但高于动态单耳接收器的定向误差;定向精度与声源频率及声源所在空间方位有关,声源位于接收器水平方向±30°范围内时,定向精度相对较高;双耳夹角和间距也会影响定向精度,且前者影响较为显著。     

Fourth-order cumulant multiple matrix reconstruction to estimate direction of arrival for coherent sound source under low signal-to-noise ratio

To address the problem of energy diffusion in the subspace of coherent sources and the difficulty of estimating the direction of arrival(DOA) with an under-ranked covariance matrix, an estimation method using a higher-order matrix transformation, i.e. the fourth-order cumulant multiple matrix reconstruction(FOC-MMR), is proposed. Firstly, short-time Fourier transform is performed on the array sound pressure data in frames. Then, the singular value decomposition of the fourth-order cumulant-expan...     

采用四阶累积量多重矩阵重构的低信噪比相干声源波达方向估计

针对相干声源子空间能量扩散且协方差矩阵欠秩难以有效估计波达方向(DOA)的问题,提出了一种采用高阶矩阵变换的估计方法-四阶累积量多重矩阵重构(FOC-MMR)。该方法首先对阵列声压数据分帧进行短时傅里叶变换,然后对四阶累积量扩展的高阶协方差矩阵进行奇异值分解(SVD)得到高阶噪声特征向量,保证该噪声特征向量与扩展后的高阶阵列流形矢量正交匹配,最终实现相干信号的DOA估计。相干单频矩形脉冲信号仿真结果表明,将FOC-MMR方法应用于均匀线阵(ULA, M=4),在信噪比SNR≥-15 dB时,相干信号(θ₁=-20°和θ₂=20°)的均方根误差保持在1.5°以内;在SNR=10 dB时,可正确分辨的两相干信号方位间隔Δθ可以低至5°。在相干脉冲声源实验中,通过混入SNR=5 dB高斯白噪声,验证了FOC-MMR算法在应用于由多个ULA组成的矩形面阵时,其分辨邻近声源和抑制高斯噪声的能力较高。FOC-MMR算法通过对声压阵列数据扩展得到满秩的高阶协方差矩阵,不仅解决了由信号相干造成的噪声和信号特征向量之间能量扩散的问题,还实现了以较高的测向精度和...     

五象限法声源自动跟踪器

介绍由两同心圆组成的等投影面积五象限声－电测量装置的工作原理。当声波与声－电转换器的主要测象限垂直时，其声波在主要测象限内产生的声－电电压最强。根据声－电电压的变化大小来确定声源的方向，实现自动跟踪。