Acoustic positioning using multiple microphone arrays

Passive acoustic techniques are presented to solve the localization problem of a sound source in three-dimensional space using off-the-shelf hardware. Multiple microphone arrays are employed, which operate independently, in estimating the direction of arrival of sound, or, equivalently, a direction vector from the array's geometric center towards the source. Direction vectors and array centers are communicated to a central processor, where the source is localized by finding the intersection of the direction lines defined by the direction vectors and the associated array centers. The performance of the method in the air is demonstrated experimentally and compared with a state-of-the-art method that requires centralized digitization of the signals from the microphones of all the arrays.     

Active stereo sound localization

Estimating the direction of arrival of sound in three-dimensional space is typically performed by generalized time-delay processing on a set of signals from a fixed array of omnidirectional microphones. This requires specialized multichannel A/D hardware, and careful arrangement of the microphones into an array. This work is motivated by the desire to instead only use standard two-channel audio A/D hardware and portable equipment. To estimate direction of arrival of persistent sound, the position of the microphones is made variable by mounting them on one or more computer-controlled pan-and-tilt units. In this paper, we describe the signal processing and control algorithm of a device with two omnidirectional microphones on a fixed baseline and two rotational degrees of freedom. Experimental results with real data are reported with both impulsive and speech sounds in an untreated, normally reverberant indoor environment.     

运动声源快速定位的声达时差法

针对声达时差法只能用于非运动声源定位的问题,本文提出一种运动声源快速定位方法。该方法以声达时差为基本定位原理,基于声源计算位置对多普勒效应进行解耦并进行声信号多普勒效应修正,根据三角定位方法构建声传播空间矩阵,以声源位置偏差度为目标基于单纯形优化搜索算法进行声源位置快速逼近,实现了对匀速直线运动的单声源的定位追踪,提高定位实时性。该方法将声达时差法拓展到运动声源的定位,同时解决了消除多普勒效应带来的计算过程复杂、运算量大的问题,仅用4个传声器就可实现运动声源的快速定位,突破了传统运动声源识别中对大传声器阵列的依赖。仿真实验和实车运动声源识别实验结果证明了该方法的有效性,本研究为短时发声运动声源的识别提供了一种简便、高效的方法。      

Blind extraction and localization of sound sources using point sources based approaches

This paper presents theoretical models for blind sound source localization and separation of the signals emitted by arbitrary point sources in free space. Source localizations are achieved by a model based approach that accounts for the spherical spreading of an acoustic wave and utilizes an iterative triangulation, based on the signals measured by a three-dimensional microphone array. Once source locations are determined, the source signals are separated by using the point source separation (PSS) method, which is valid for all types of signals, including harmonic, continuous, transient, random, narrowband and broadband. General solutions for signals separation are presented. Theoretically, PSS can reconstruct the individual source signals exactly. This is because it employs the free-space Green's function, which defines the exact correlation among individual sources and measurement microphones. To validate PSS, numerical simulations are carried out and results are compared with those obtained by FastICA (Independent Component Analysis) code. The impacts of various parameters such as the microphone configuration, type of source signals, signal to noise ratio, number of microphones and source localization errors on the quality of signals separation by using PSS and FastICA are examined. The advantages and disadvantages of PSS and FastICA are compared and discussed.     

Determining the direction to a sound source in air using vector sound-intensity probes

This paper is concerned with probes for measuring vector sound intensity in air using the minimum number of sound-pressure sensors. The probes consist of an arrangement of four small microphones at the vertices of a regular tetrahedron. They are connected to a digital signal processor, which determines the sound-intensity vector, using the cross-spectral formulation based on finite-difference approximations. Determining the direction of a sound source is an obvious application. To do this accurately the probes should be omnidirectional. This implies that the microphones in the probe have to be omnidirectional and to have the same response. Results in the paper show that the direction of a sound source can be determined with an accuracy of a few degrees. Two types of probes are described. One measures the sound-intensity vector in three-dimensional space. The other measures the vector in a half space such as would occur above the ground or in front of a wall.     

封闭空间声场重构的多层等效源法

对于封闭空间内的多途反射声,传统的等效声源法将其等效为距离边界一定距离的单层等效声源体进行声场重构,然而等效源与边界的距离选取依据不确定。因此,为获得等效声源配置的最优距离,在等效声源法(ESM)的基础上构建多层等效声源,提出一种适用于封闭环境声场重构的多层等效声源法(MESM),并依据等效声源的空间分布的稀疏性来获得等效声源强度信息。首先给出多层等效源法的理论依据,其次通过数值计算以及实验测试两种方式对比验证了所提方法。数值结果表明:MESM相比于ESM可在600 Hz以上频段获得低5~10 dB左右的重构误差,但是200 Hz以下的低频重构误差会增加5 dB左右。实验结果表明:MESM可比ESM获得更低的重构误差。文章最后基于数值计算研究了所提方法的主要影响因素。研究表明:虽然MESM会比ESM耗费2倍的计算时间,但在整体频率范围内,MESM可在ESM基础上提升600 Hz以上的重构性能。另外,等效声源的层数和层内数目的改变不会影响声场重构性能,而当传声器数目较多、阵列位置随机、空间边界的吸声系数不是很大时,MESM可获得比ESM更低重构误差,特别是600 Hz以上的中频段区间。      

Source localization in complex listening situations: selection of binaural cues based on interaural coherence

In everyday complex listening situations, sound emanating from several different sources arrives at the ears of a listener both directly from the sources and as reflections from arbitrary directions. For localization of the active sources, the auditory system needs to determine the direction of each source, while ignoring the reflections and superposition effects of concurrently arriving sound. A modeling mechanism with these desired properties is proposed. Interaural time difference (ITD) and interaural level difference (ILD) cues are only considered at time instants when only the direct sound of a single source has non-negligible energy in the critical band and, thus, when the evoked ITD and ILD represent the direction of that source. It is shown how to identify such time instants as a function of the interaural coherence (IC). The source directions suggested by the selected ITD and ILD cues are shown to imply the results of a number of published psychophysical studies related to source localization in the presence of distracters, as well as in precedence effect conditions.     

Source characterization of a subsonic jet by using near-field acoustical holography

In the present study, patch near-field acoustical holography was used in conjunction with a multireference, cross-spectral sound pressure measurement to visualize the sound field emitted by a subsonic jet and to predict its farfield radiation pattern. A strategy for microphone array design is described that accounts for the low spatial coherence of aeroacoustic sources and for microphone self-noise resulting from entrained flow near the jet. In the experiments, a 0.8-cm-diameter burner was used to produce a subsonic, turbulent jet with a Mach number of 0.26. Six fixed, linear arrays holding eight reference microphones apiece were disposed circumferentially around the jet, and a circular array holding sixteen, equally spaced field microphones was traversed along the jet axis to measure the sound field on a 30-cm-diameter cylindrical surface enclosing the jet. The results revealed that the jet could be modeled as a combination of eleven uncorrelated dipole-, quadrupole-, and octupole-like sources, and the contribution of each source type to the total radiated sound power could be identified. Both the total sound field reconstructed in a three-dimensional space and the farfield radiation directivity obtained by using the latter model were successfully validated by comparisons to directly measured results.     

传声器阵列特征值滤波去噪方法

作为二阶统计量的互谱矩阵(CSM)是声学成像算法的核心输入量.为增强传声器阵列的去噪表现,研究了互谱矩阵特征值滤波的机理,并提出了两种新型的特征值滤波方法的设计准则:(1)声源互谱矩阵的Stein无偏风险估计(SURE收缩),即基于SURE准则的特征值软阈值收缩;(2)进一步提高声源互谱矩阵EYM(Eckart-You...     

Effects of in-the-ear microphone directionality on sound direction identification

As advanced signal processing algorithms have been proposed to enhance hearing protective device (HPD) performance, it is important to determine how directional microphones might affect the localization ability of users and whether they might cause safety hazards. The effect of in-the-ear microphone directivity was assessed by measuring sound source identification of speech in the horizontal plane. Recordings of speech in quiet and in noise were made with Knowles Electronic Manikin for Acoustic Research wearing bilateral in-the-ear hearing aids with microphones having adjustable directivity (omnidirectional, cardioid, hypercardioid, supercardioid). Signals were generated from 16 locations in a circular array. Sound direction identification performance of eight normal hearing listeners and eight hearing-impaired listeners revealed that directional microphones did not degrade localization performance and actually reduced the front-back and lateral localization errors made when listening through omnidirectional microphones. The summed rms speech level for the signals entering the two ears appear to serve as a cue for making front-back discriminations when using directional microphones in the experimental setting. The results of this study show that the use of matched directional microphones when worn bilaterally do not have a negative effect on the ability to localize speech in the horizontal plane and may thus be useful in HPD design.     

Blind separation of sound sources from the principle of least spatial entropy

The aim of the paper is to offer a method for separating incoherent and compact sound sources which may overlap in both the space and frequency domains. This is found of interest in acoustical applications involving the identification and ranking of sound sources stemming from different physical origins. The principle proceeds in two steps, the first one being reminiscent to source reconstruction (e.g. as in near-field acoustical holography) and the second one to blind source separation. Specifically, the source mixture is first expanded into a linear combination of spatial basis functions whose coefficients are set by backpropagating the pressures measured by an array of microphones to the source domain. This leads to a formulation similar, but no identical, to blind source separation. In the second step, these coefficients are blindly separated into uncorrelated latent variables, assigned to incoherent “virtual sources”. These are shown to be defined up to an arbitrary rotation. A unique set of sound sources is finally recovered by searching for that rotation (by conjugate gradient descent in the Stiefel manifold of unitary matrices) which maximizes their spatial compactness, as measured either by their spatial variance or their spatial entropy. This results in the proposal of two separation criteria coined “least spatial variance” and “least spatial entropy”, respectively. The same concept of spatial entropy, which is central to the paper, is also exploited in defining a new criterion, the entropic L-curve, dedicated to determining the number of active sound sources. The idea consists in considering the number of sources that achieves the best compromise between a low spatial entropy (as expected from compact sources) and a low statistical entropy (as expected from a low residual error). The proposed methodology is validated on both laboratory experiments and numerical data, and illustrated on an industrial example concerned with the ranking of sound sources on a Diesel engine. At the same time, its robustness to the estimated number of active sources is demonstrated.     

三通管路管壁声传递损失测试方法

针对汽车进气系统三通管路的特点,提出了多通管路的管壁传递损失测试方法。并以某车型的双涡轮增压发动机进气三通管道为例,采用该方法评价其用塑料代替铝后的声学性能,主要以声传递损失来评价涡轮增压器噪声通过三通连接管路管壁的辐射和透射特性。测试过程中,三通管道的两个连接涡轮增压器端口分别用声源两次发声,靠近进气歧管端口采用两种不同反射末端,然后在每段管路布置两个压力场扬声器进行测试,并基于平面波分离入射波和反射波,同时在三通管道外用声功率半球面十点分布法自由场扬声器测试,经过3次测量来计算管道管壁的声传递损失。由于声传递损失是管道本身特性决定,所以该测试方法能够准确找出塑料件和金属件在不同频率的声学特性差异。而后,在声传递损失测试结果的基础上,结合近场声全息方法和波束形成原理进行声源识别,可知该三通管路材质改为塑料后主要噪声来自焊缝薄弱处的中高频透射声和管壁结构的低频辐射声。     

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。     

Beamforming with a circular microphone array for localization of environmental noise sources

It is often enough to localize environmental sources of noise from different directions in a plane. This can be accomplished with a circular microphone array, which can be designed to have practically the same resolution over 360°. The microphones can be suspended in free space or they can be mounted on a solid cylinder. This investigation examines and compares two techniques based on such arrays, the classical delay-and-sum beamforming and an alternative method called circular harmonics beamforming. The latter is based on decomposing the sound field into a series of circular harmonics. The performance of the two signal processing techniques is examined using computer simulations, and the results are validated experimentally.     

吸顶式传声器阵列阵元坐标标定方法

提出了一种吸顶式传声器阵列阵元坐标的标定方法。针对在混响声场中,时延估计算法性能严重下降从而导致在标定传声器阵元坐标时产生较大误差的问题,提出了利用脉冲声源作为标定声源,并且截取脉冲源直达声的方法来抑制混响声场的影响,提高传声器阵元坐标标定的精度。建立了阵元坐标标定的误差分析模型,并以白噪声和脉冲声源作为标定声源进行数据仿真和对比分析。仿真结果表明,使用脉冲声源作为标定声源能有效地抑制混响声场的影响,获得传声器阵列阵元的准确坐标。同时,在封闭的房间内建立起孔径为3.5 m、64阵元的螺旋状吸顶传声器阵列进行了实验研究,实验结果验证了本文提出方法的有效性。      

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.     

Localization of a sound source in a noisy environment by hyperbolic curves in quefrency domain

Time Difference of Arrivals (TDOAs) of sound waves between microphones have to do with source localization. How well a sound source can be localized depends on how precisely the TDOAs are estimated. Although many ways to estimate TDOA have been proposed, noise always prevents us from finding exact time differences more or less in practice. Cross correlation has been the most prevalent way to estimate time difference, and various cross correlations robust to noise have also been developed. Nevertheless, much remains to be done for exact TDOA estimation under noisy environments. A novel way to show time delays in quefrency domain by removing noise has been proposed, which is called Minimum Variance Cepstrum (MVC). In particular, it is practically desirable to visualize source position with as few number of sensors as possible. Once TDOAs are obtained precisely, it is enough to show the source position in a 2-D plane using hyperbolic curves with only three sensors. In this work, the MVC is adopted to accurately estimate TDOAs under noise, and a way to localize an acoustic source by intersecting hyperbolic curves using the TDOAs between three microphones is proposed. Numerical simulations on TDOA estimation and source localization with white Gaussian noise demonstrated that the proposed method worked well under the noisy environment, and we compared the results with those of other old but well-established cross correlation estimators. In addition, experiments to detect a leaking point on a pipe successfully showed where the leak sound was generated.     

Three-dimensional sound localization from a compact non-coplanar array of microphones using tree-based learning

One of the various human sensory capabilities is to identify the direction of perceived sounds. The goal of this work is to study sound source localization in three dimensions using some of the most important cues the human uses. In an attempt to satisfy the requirements of portability and miniaturization in robotics, this approach employs a compact sensor structure that can be placed on a mobile platform. The objective is to estimate the relative sound source position in three-dimensional space without imposing excessive restrictions on its spatio-temporal characteristics and the environment structure. Two types of features are considered, interaural time and level differences. Their relative effectiveness for localization is studied, as well as a practical way of using these complementary parameters. A two-stage procedure was used. In the training stage, sound samples are produced from points with known coordinates and then are stored. In the recognition stage, unknown sounds are processed by the trained system to estimate the 3D location of the sound source. Results from the experiments showed under +/-3 degrees in average angular error and less than +/-20% in average radial distance error.     

Front-back confusion resolution in three-dimensional sound localization using databases built with a dummy head

Sound localization plays an important role in everyday life. It helps us to separate sounds coming from different sources and thus to acquire acoustic information. This paper describes an algorithm for localizing the position of a sound source, as recorded by dummy head microphones. The recorded signals are considered to be basic, random signals within an imaginary round room. The goal of this research is to localize random signals produced from different positions using information about basic signals. The method used is based on the identification of similarities between basic and random signals. It includes an interaural time difference comparison at the beginning, and continues with further analysis of the differences in signal spectrums. One of the main issues arising in sound localization is the problem of front-back confusion, and this paper shows how it was resolved by the use of reference signals.     

A directional acoustic array using silicon micromachined piezoresistive microphones

The need for noise source localization and characterization has driven the development of advanced sound field measurement techniques using microphone arrays. Unfortunately, the cost and complexity of these systems currently limit their widespread use. Directional acoustic arrays are commonly used in wind tunnel studies of aeroacoustic sources and may consist of hundreds of condenser microphones. A microelectromechanical system (MEMS)-based directional acoustic array system is presented to demonstrate key technologies to reduce the cost, increase the mobility, and improve the data processing efficiency versus conventional systems. The system uses 16 hybrid-packaged MEMS silicon piezoresistive microphones that are mounted to a printed circuit board. In addition, a high-speed signal processing system was employed to generate the array response in near real time. Dynamic calibrations of the microphone sensor modules indicate an average sensitivity of 831 microV/Pa with matched magnitude (+/-0.6 dB) and phase (+/-1 degree) responses between devices. The array system was characterized in an anechoic chamber using a monopole source as a function of frequency, sound pressure level, and source location. The performance of the MEMS-based array is comparable to conventional array systems and also benefits from significant cost savings.