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.     

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.     

A miniaturized adaptive microphone array under directional constraint utilizing aggregated microphones

This paper introduces a miniaturized microphone array using the Directionally Constrained Minimization of Power (DCMP) method, which utilizes the transfer functions of microphones located at the same place, namely aggregated microphones. The phased microphone array realizes a noise reduction and direction of arrival (DOA) estimation system according to differences in the arrival time, phase shift, and/or the level of the sound wave for each microphone. Hence it is difficult to miniaturize the microphone array. The objective of our research is to miniaturize the system size using aggregated microphones. In this paper, we first show that the phased microphone array system and the proposed aggregated microphone system can be described within the same framework. We then apply a microphone array under directional constraint to the aggregated microphones and compare the proposed method with the microphone array. We show the directional pattern of the aggregated microphones. We also show the experimental results regarding DOA estimation.     

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.     

数字化声学测量技术 Ⅱ.数字式声强及声功率测量系统

本文介绍在数字声学测量分析系统中，通过双传声器信号互谱密度的计算进行声强及声功率测量的基本原理。该数学分析系统由微计算机，数字信号处理卡和Ａ／Ｄ变换卡组成。在一个数字系统中，通过快速傅里叶变换（ＦＥＴ）进行互谱计算是十分有效的。本文着重介绍了，在声强的测量分析中对声强探头两传声器的固有相位差进行补偿的重要性和补偿方法，这是声强测量的重要环节。     

A technique for measuring the sound of a moving tire

A measurement and analysis technique has been developed to determine the narrow band spectra and the radiation patterns of the sound emitted by a moving tire. The sound is measured by a semicircular array of stationary microphones as the tire passes by the array and is recorded on a multi-channel tape recorder. In the analysis procedure corrections are made for effects associated with a moving sound source, such as the non-stationarity of the signal due to the time-dependent transmission path and the Döppler frequency shifts. In this way the power spectra and the radiation pattern of the sound signal are determined as if the receiver were moving with the tire at a fixed distance. A relationship between the Döppler effect and the frequency resolution associated with the finite Fourier transform is presented. This relation is used as a basis for the Döppler correction procedure.     

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.     

An acousto-optic beamformer

There is a great variety of beamforming techniques that can be used for localization of sound sources. The differences among them usually lie in the array layout or in the specific signal processing algorithm used to compute the beamforming output. Any beamforming system consists of a finite number of transducers, which makes beamforming methods vulnerable to spatial aliasing above a certain frequency. The present work uses the acousto-optic effect, i.e., the interaction between sound and light, to localize sound sources in a plane. The use of a beam of light as the sensing element is equivalent to a continuous line aperture with an infinite number of microphones. This makes the proposed acousto-optic beamformer immune to spatial aliasing. This unique feature is illustrated by means of simulations and experimental results within the entire audible frequency range. For ease of comparison, the study is supplemented with measurements carried out with a line array of microphones.     

Correlation of underwater acoustic signals simultaneously received by omnidirectional and directional means

A comparison of experimental data on the spatial correlation between acoustic signals simultaneously received by an omnidirectional hydrophone and a directional vertical array is carried out. The spatial correlation was measured between the signals received at different distances in a deep ocean. The points of reception were positioned in two convergence zones along the path of sound propagation with a point-to-point distance of about 64 km. Pseudonoise signals were emitted in the frequency range (0.8–2.0) kHz and received by a vertical array, whose beam had a width of ∼2°. Concurrently, multipath signals received with the central hydrophone of the array were recorded. Signals in the first and second convergence zones were received at different times. Nevertheless, in the case of the directional reception, the coefficients of spatial correlation between such signals appeared to be as high as 0.64–0.74 even under the conditions of incomplete resolution of signals in the angle of arrival in the vertical plane. At the same time, in the case of omnidirectional reception, the coefficients of spatial correlation were below 0.32.     

Estimation of multiple sound source directions using artificial robot ears

Estimating the direction of a sound source is an important technique used in various engineering fields, including intelligent robots and surveillance systems. In a household where a user’s voice and noises emitted from electric appliances originate from arbitrary directions in 3-D space, robots need to recognize the directions of multiple sound sources in order to effectively interact with the user.This paper proposes an ear-based estimation (localization) system using two artificial robot ears, each consisting of a spiral-shaped pinna and two microphones, for application in humanoid robots. Four microphones are asymmetrically placed on the left and right sides of the head. The proposed localization algorithm is based on a spatially mapped generalized cross-correlation function which is transformed from the time domain to the space domain by using a measured inter-channel time difference map. For validation of the proposed localization method, two experiments (single- and multiple-source cases) were conducted using male speech. In the case of a single source, with the exception of laterally biased sources, the localization was achieved with an error of less than 10°. In a multiple-source environment, one source was fixed at the front side and the other source changed its direction; from the experimental results, the error rates on the localization of the fixed and varying sources are 0% and 36.9% respectively within an error bound of 15°.     

Design and implementation of a MEMS microphone array system for real-time speech acquisition

Despite many attractive features and the potential for capturing sound in challenging acoustic environments, arrays with a large number of microphones have for a long time been discarded as a practical solution for speech acquisition. This is, among other reasons, due to the high production and computational costs. Only a few realizations of large microphone array systems have been documented, mainly for research and instrumentation use. The advent of MEMS microphones and computationally powerful off-the-shelf hardware has created new possibilities for microphone array development. We investigate a real life application, specifically the case of live sports broadcast, and the requirements that a such application imposes on a microphone array system. We present a system architecture of the first large (300 element circular array with a diameter of 2 m) MEMS microphone array system. In the proposed system, the latest technological advances are utilized to create a user-friendly array control interface. The array’s performance is examined in an anechoic chamber and on a crowded basketball field, and finally compared with existing solutions. The results illustrate the potential of a large MEMS microphone array as part of the technological development in sound acquisition for entertainment and security applications.     

Research on intensity measurement in room impulse field

A new system of sound intensity measurement for impulse field in the room was proposed. This measurement system consists of a repeatable inspiriting sound source and a microphone fixed on a slowly rotating platform, which is equivalent to a circle microphone array composed of many perfectly matched microphones. The test principle was presented and typical application was described. Based upon this system the sound intensity measurement for impulse field in the room was realized. Therefore, not only time but also spatial information of room impulse response can be obtained.     

The diffuse sound field in energetic analysis

Measuring the degree of diffuseness of a sound field is crucial in many modern parametric spatial audio techniques. In these applications, intensity-based diffuseness estimators are particularly convenient, as the sound intensity can also be used to obtain, e.g., the direction of arrival of the sound. This contribution reviews different diffuseness estimators comparing them under the conditions found in practice, i.e., with arrays of noisy microphones and with the expectation operators substituted by finite temporal averages. The estimators show a similar performance, however, each with specific advantages and disadvantages depending on the scenario. Furthermore, the paper derives an estimator and highlights the possibility of using spatial averaging to improve the temporal resolution of the estimates.     

Representation of the sound field of a turbulent vortex ring as a superposition of quadrupoles

Results of an experimental study of the directivity of noise produced by a turbulent vortex ring (Re≈10⁵) are presented. The acoustic measurements were performed in an anechoic chamber by a circular array of microphones with its center lying at the symmetry axis of the ring. A method of a synchronous processing of acoustic signals is proposed. This method allows one to separate different quadrupole components in the measured sound field of a turbulent vortex.     

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.     

基于蒙特卡罗法的实时空间三维声源定位

构建了一个基于四个声音传感器的信号时延采集系统,根据采集系统得到的三个时间差和传感器的响应顺序,提出了一种基于蒙特卡罗法实时空间的三维声源定位算法.该算法通过三维声音定位的非线性方程,构建一个三维模函数,通过寻找空间全局收敛点,并根据公差容限进行变步长搜索,准确快速地计算出声源的位置.     

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。     

水下目标高分辨方位估计特征分解法的C30硬件实验研究

高分辨方位估计特征分解法是一类性能良好的方法，本文从水下阵列信号处理的实际应用出发，对这类方法进行了C30硬件实验研究，所得结果为高分辨技术的实际应用提供了重要依据文中构造了高分辨方位估计的C30硬件实验系统；并实现了高分辨方位估计的Pisarenko法、MUSIC法、Johnson法和最小摸（Mininorm）法；对高分辨实验系统进行了仿真实验和消声水池实验，并测试了各特征分解法的运行时间．结果表明，高分辨C30实验系统表现了良好的方位估计统计性能和实时性能     

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.