采用声场成像的声源自适应降噪方法

针对现有声音定位技术动态适应能力弱的问题,基于由21个MSM261S4030H0组成的麦克风阵列群和声波接收时间差算法实现的声源定位技术,提出声源自适应降噪方法.该方法在声场成像的基础上,通过能根据声源实际情况调整参量的异常值剔除和迭代修正复合式降噪过程,兼顾定位的准确性和实时性,实现8 m内的声源自适应定位.     

基于声矢量传感器的分布式定位系统在水下宽带声源定位中的应用

提出了基于声矢量传感器的分布式浮标网络定位系统,研究了不同应用背景下单个声矢量传感器的测向算法,推导了目标DOA估计的Cramer-Rao界,给出了分层海水介质中多个声矢量传感器的几何定位算法。数值仿真结果表明:(1)系统的定位性能强烈依赖于接收信噪比;(2)该系统适用于单一强声源的定位。     

基于声音的跟踪摄像的初步研究

仿照生物的趋声性,可以实现摄像头追踪拍摄一个不在其视场范围内的声源;依据图像处理技术,可以精确定位在摄像头视场内的声源。将这两种技术结合起来,建立了一套实验系统。从利用声源发出的声音信号传输到两个分开一定距离的声音探测器上的时间差初步定位声源,再利用声源的运动规律使用图像处理技术识别并精确定位声源,通过系统与PC机间的实时通信和回转部件带动声音探测器和摄像头一起做精确灵活的转动,摄像头最终对准了声源。实现了基于声音的跟踪摄像过程。     

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.      

一种分布式双麦克风线阵声源定位方法

分布式麦克风阵列由于比传统的单阵列具有更大的空间孔径,可以获得更好的声源定位性能,因此基于分布式麦克风阵列的声源定位方法成为当前麦克风阵列领域研究的热点之一。本文研究了一种基于分布式双麦克风线阵的声源定位方法,并进行了系统实现,从理论上剖析了该算法的定位精度与单阵列定向误差以及声源位置之间的关系,而且还揭示了声源高度扰动对该算法定位精度的影响。最后,分别通过仿真实验和实际定位系统的测试结果,验证了本文理论分析的正确性。     

基于银膜的自准直光纤法布里-珀罗腔设计及声源定位研究

提出一种基于银膜的自准直外腔式光纤法布里-珀罗(EFPI)干涉仪声传感器阵列,用于中低频声信号的检测以及二维平面声源定位。传感器阵列由三个相同结构的EFPI组成,结构简单、制作简便。银膜的应用和准直器的引入提高了传感器的声压灵敏度,扩大了声源定位范围。实验结果表明：单个传感器声压灵敏度为185 mV/Pa,最小可探测声压为52.7μPa/Hz^1/2@500 Hz。在声源指向性实验中,声源设置在传感器正前方时,其声压灵敏度达到185 mV/Pa,声源设置在传感器侧面(90°,270°)时,其声压灵敏度衰减仅为4.5%。传感器阵列结合到达时间差技术,成功实现高精度的声源定位,系统的理论空间分辨率为0.71 cm,最大定位误差不超过2.8 cm,定位范围为200 cm×200 cm,具有成本低、实用性高等优点。     

基于双传声器对的多声源二维定位跟踪算法

提出一种房间混响声场环境下的多声源二维定位跟踪算法。研究了基于盲源分离的时延估计,以及联合空间分布的多个传声器对的定位算法。用高斯似然函数解决在多源、多维情况下声源定位的时延匹对模糊问题,使之能够用双传声器对实现对多个声源的二维定位,结合粒子滤波算法实现对多个运动声源的跟踪。仿真实验验证了提出算法的有效性。      

波达方向初始化空间混合概率模型的语音增强

提出了波达方向初始化空间混合概率模型的语音增强算法.通过声源定位估计出声源波达方向,再根据此计算相对传递函数,进而构造空间协方差矩阵来初始化空间混合概率模型.论证了相对传递函数在作为模型参数中语音协方差矩阵的主特征向量时,空间混合概率模型对应的概率分布可达到最大值,进而使期望最大化算法在迭代时更易收敛,以得到期望的掩蔽...     

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。     

基于虚拟仪器的简易声源追踪系统设计

为了在大型粮仓中局部出现害虫鸟类时可以快速找出其位置,解决目前声源定位设备成本高、计算量大实时性差的问题,对波达声差（TDOA）定位技术和互相关时延算法进行研究,提出一种改进了广义互相关时延估计算法,设计实现一种基于虚拟仪器的简易声源追踪系统根据害虫害鼠发出声音进行位置追踪进行粮仓管理。详细介绍了系统软硬件设计实现,系统可以实现参数设置、声源坐标的实时获取、文件存储、三维动态和数值显示等功能。实验结果显示,该系统具有实时性好、精度高、计算量小、工作稳定等特点。     

基于遗传算法的传声器阵列位置有源校正研究

传声器阵列的位置误差会导致高精度定位应用场景中算法性能下降,为解决这一问题,该文提出了一种二维平面传声器阵列位置参数的有源校正方法.在校正声源位置已知的情况下,利用各阵元间的到达时间差均方误差之和最小化作为优化目标设计代价函数,根据遗传算法搜索阵列真实位置的全局最优解,从而估计得到传声器阵列的准确位置信息.仿真与半消声...     

单层传声器阵列信号空间重采样的声波分离方法

为了重构非自由声场中目标声源的声场响应,提出单层传声器阵列信号空间重采样的声波分离方法.以球面波函数为基函数,建立由系列球面波函数叠加表达的声场数学模型.基于近场声全息原理,利用单层传声器阵列面上空间重采样形成的两组声压测量信号,求解基函数系数,并重构出传声器阵列两侧声源各自的声场响应,实现声波分离.使用脉动球和振动球共同作用的非自由声场,检验了数学模型以及传声器信号信噪比、传声器阵列形状和面积、声源中心位置、频率等关键参数对声波分离精度的影响,并在全消声室内进行了实验验证.最后,对单层传声器阵列重采样的声波分离方法的实施给出了建议.     

Automatic 3D scanning surface generation for microphone array acoustic imaging

This work presents a new technique for automatically generating the 3D scanning surface for acoustic imaging using microphone arrays. Acoustic images, or maps, of sound coming from spatially distributed sources, may be generated from microphone array data using algorithms such as beamforming. Traditional 2D acoustic maps can contain errors in the near-field if the object being imaged has a 3D shape. It has been shown that using the 3D surface geometry of an object as a scanning surface for beamforming can provide more accurate results. The methods used previously to generate this 3D scanning surface have either required existing CAD (Computer-Aided Design) models of the object being acoustically imaged or have required separate equipment which is generally bulky and expensive. The new method uses one or more cameras in the array, a data projector, and structured light code to automatically generate the 3D scanning surface. This has the advantage that it is inexpensive, can be incorporated as an add-onto existing microphone arrays, has short scan time, and is capable of being extended to imaging dynamic scenes. This technique is tested using beamforming and CLEAN-SC (CLEAN based on spatial Source Coherence) algorithms for a spherical array and an Underbrink multi-arm spiral array. For sound sources located about 1.2 m from the array, the mean position errors obtained are 6 mm. This is a quarter of the diameter of the mini-speakers being used as a sound sources.     

Localization of distinct reflections in rooms using spherical microphone array eigenbeam processing

This paper presents an experimental and comparative study of several spherical microphone array eigenbeam (EB) processing techniques for localization of early reflections in room acoustic environments, which is a relevant research topic in both audio signal processing and room acoustics. This paper focuses on steered beamformer-based and subspace-based localization techniques implemented in the spherical EB domain, including the plane-wave decomposition, eigenbeam delay and sum, eigenbeam minimum variance distortionless response, eigenbeam multiple signal classification (EB-MUSIC), and eigenbeam estimation of signal parameters via rotational invariance techniques (EB-ESPRIT) methods. The directions of arrival of the original sound source and the associated reflection signals in acoustic environments are estimated from acoustic maps of the rooms, which are obtained using a spherical microphone array. The EB-domain-based frequency smoothing and white noise gain control techniques are derived and employed to improve the performance and robustness of reflection localization. The applicability of the presented methods in practice is confirmed by experiments carried out in real rooms.     

Robust tracking of moving sound source using multiple model Kalman filter

In this paper a novel method for tracking an active speaker in a noisy and reverberant environment by means of a spatially distributed microphone array is presented. Firstly, a sound source localization algorithm based on time delays of arrival (TDOA) in microphone pairs provides observed position estimates. Then these remarkably noisy estimates are filtered by a multiple model Kalman filter (MMKF) in order to obtain a smoothed trajectory of the speaker’s movement. Compared with the traditional Kalman filter (KF), simulated results prove the MMKF is more robust and effective in noisy environments.     

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。     

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。     

传声器阵列对高频弱声源的声成像识别方法

研究了传声器阵列对高频弱声源的识别定位方法。该方法根据高频声源的指向性和阵列探测特性等特点,提出了利用信噪比加权方法提高有效阵元对声成像的贡献,根据信噪比的大小对每个阵元添加不同的权值,可以显著提高传声器阵列对高频弱声源的声像清晰度。仿真分析了阵元加权和不加权两种方法对阵列声成像结果的影响,以某型号笔记本电脑电路板噪声为对象进行的实验表明,在阵列测量中充分利用有效阵元信号可以实现对声压级低达10~20dB的微弱噪声源的精确测量。      

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.