Distributed sound source localization algorithm with sound velocity calibration in windy environments

A new sound source localization method with sound speed compensation is proposed to reduce the wind influence on the performance of conventional TDOA(Time Difference of Arrival) algorithms. First, the sound speed is described as a set of functions of the unknown source location, to approximate the acoustic velocity field distribution in the wind field. Then,they are introduced into the TDOA algorithm, to construct nonlinear equations. Finally, the particle swarm optimization algorithm is used to estimate the source location. The simulation results show that the proposed algorithm can significantly improve the localization accuracy for different wind velocities, source locations and test area sizes. The experimental results show that the proposed method can reduce localization errors to about 40% of the original error in a four nodes localization system.     

虚拟接收目标定位技术中引导声源对目标声源的信噪比补偿

目标声源的低信噪比是目标定位技术中的瓶颈。在浅海近似均匀环境中,利用虚拟接收方法对声源测距的目标定位方法相比于传统的匹配场定位方法可以避免对环境参数和声场模型的依赖,同时省去生成拷贝场时的大量计算。通过数值模拟和实验数据处理,比较了在虚拟接收方法声源测距中不同信噪比的引导声源对目标声源测距的结果。提高引导声源的信噪比,可以一定程度上提高于涉条纹清晰度,从而提高测距准确度。提高引导声源的信噪比可作为目标声源的信噪比补偿是利用虚拟接收方法的目标定位技术的又一个优点。      

Localization dominance in the median-sagittal plane: effect of stimulus duration

Localization dominance is an aspect of the precedence effect (PE) in which the leading source dominates the perceived location of a simulated echo (lagging source). It is known to be robust in the horizontal/azimuthal dimension, where binaural cues dominate localization. However, little is known about localization dominance in conditions that minimize binaural cues, and most models of precedence treat the phenomena as "belonging" to the binaural system. Here, localization dominance in the median-sagittal plane was studied where binaural cues are greatly reduced, and monaural spectral/level cues are thought to be the primary cues used for localization. Lead-lag pairs of noise bursts were presented from locations spaced in 15 degrees increments in the frontal, median-sagittal plane, with a 2-ms delay in their onsets, for source durations of 1, 10, 25, and 50-ms. Intermixed with these trials were single-speaker trials, in which lead and lag were summed and presented from one speaker. Listeners identified the speaker that was nearest to the perceived source location. With single-speaker stimuli, localization improves as signal duration is increased. Furthermore, evidence of elevation compression was found with a dependence on duration. With lead-lag pairs, localization dominance occurs in the median plane, and becomes more robust with increased signal duration. These results suggest that accurate localization of a co-located lead-lag pair is necessary for localization dominance to occur when the lag is spatially separated from the lead.     

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。      

高斯射线束法在深海声源定位中的应用

在深海匹配场(Matched Field Processing-MFP)声源定位中,拷贝场模型的计算精度和速度对声源的定位效果和定位效率有着直接的影响。针对这一问题,本文数值仿真研究了深海匹配场定位中采用高斯射线束理论作为拷贝场计算模型的可行性。高斯射线束理论具有计算速度快,物理意义清晰,并且适合并行化处理等优点。文中利用不同水文环境、不同频域范围的大量仿真实验来证明该模型的适用性。结果表明,高斯射线束模型在深远海的声源定位中有很好的定位精度,并且在较低频段同样有效,可以作为深远海声源定位的拷贝场计算模型。     

Improved steered response power method for sound source localization based on principal eigenvector

Sound source localization is essential in many microphone arrays application, ranging from teleconferencing systems to artificial perception in a reverberant noisy environment. The steered response power (SRP) using the phase transform (SRP-PHAT) source localization algorithm has been proved robust, however, the performance of the SRP-PHAT algorithm degrades in highly reverberant noisy environment. Though the SRP-based maximum likelihood localizers are more robust than SRP-PHAT, they have the drawback of requiring noise variance to be estimated in a silent room. This paper presents an improved SRP-PHAT algorithm based on principal eigenvector. Sound source location is estimated from the principal eigenvector computed from the frequency-domain correlation matrix. Using both simulated and real data, we show that the proposed algorithm achieves higher source localization accuracy compared to the SRP-PHAT algorithm.     

水下声学滑翔机用于东印度洋声传播特性分析及声源估计

常规实验方法无法同步获取深海大尺度声学和水文数据,水下滑翔机可作为同步观测平台解决该问题.首先利用在东印度洋北部海域水下滑翔机同步获取的声传播和水文实验数据,分析了水下滑翔机的自噪声谱级和实验海区声传播特性,然后推算并修正了滑翔机水下运动轨迹,利用第一影区水下滑翔机接收声传播信号的脉冲多途到达时间差对声源进行测距与定深。潜标接收噪声与滑翔机自噪声谱级对比表明,水下滑翔机在海洋中无动力运动时的系统自噪声接近于潜标观测的海洋环境噪声。滑翔机实测的声传播损失与模型计算结果吻合较好,第一影区水下声源测距定深结果与实际位置较为一致,测距与定深的相对误差均小于5%。利用加载水听器的水下滑翔机可以实现水文环境数据与声学信号的同步观测,对深海声传播特性测量及定位算法研究具有重要意义。      

基于简正波分解的不同阵列匹配场定位性能分析

针对利用不同阵列对浅海环境中水下目标的定位问题,基于简正波分解方法,对组合阵的目标声源定位性能进行了研究,着力解决在实际实验环境下定位性能不够高的问题,并降低实验设备布放难度.在浅海环境下,基于匹配场理论的声接收阵可实现目标的定位,但定位性能受阵形、阵元数目等影响.通过研究不同声接收阵的简正波分解矩阵,可以有效辨别不同阵形定位性能的优劣.仿真实验表明,当某一子阵简正波分解效果较差时,会降低组合阵的定位性能.基于实际实验的需求,在对短垂直阵和组合阵性能的研究中发现,由于水平阵对接收声场的定位模糊度函数中的旁瓣有抑制效果,从而造成模糊度函数表面上旁瓣较低,定位目标的主旁瓣比有所提升的现象.仿真实验表明,不同组合阵形的定位准确度均在90%以上,基于实际应用的考虑,组合阵无疑是对定位性能和实验复杂度的折中选择.     

Application of focusing arrays to the problems of acoustic brightness thermometry

Theoretical and experimental studies on the localization of heated objects by the methods of acoustic brightness thermometry are carried out. It is demonstrated that, in the case of using a single focusing array, the spatial localization of heated objects depends on the size of the source. One-and two-dimensional tomography of a real heated source is performed by an acoustic thermal tomograph with a focusing array. The results agree well with the data calculated according to the suggested model. The applicability of correlation focusing acoustic brightness thermometry to the localization of a heated source is investigated both theoretically and experimentally. It is demonstrated that a considerable increase in the spatial resolution of the method leads to a significant loss in sensitivity.     

分布式压缩感知麦克风阵列多声源方位估计*

麦克风阵列已被广泛应用于音/视频会议等人机交互领域中时,多声源应用场景对声源方位估计性能提出了更高的要求。压缩感知(CS)声源定位算法将声源定位问题转化为信号的稀疏重构问题,相比传统的定位算法如相位变换加权（SRP-PHAT）和时延累加定位（DS）能够获得较高的定位性能,但多声源的存在一定程度上降低了稀疏程度,影响了CS重构性能。考虑到传统的CS定位算法并未利用多个连续语音帧之间声源空间向量的共同稀疏性,提出采用分布式压缩感知（DCS）理论以改善多声源的稀疏恢复估计的性能。仿真和实验结果表明,相比于传统定位算法和CS-OMP算法,DCS-SOMP算法在不同信噪比和不同声源强度的环境中,对多声源的方位估计都具有更好的定位性能和定位稳健性。     

风场环境中声速修正的分布式声源定位算法

为减小声速误差对定位精度的影响,提出了一种基于声速修正的分布式声源定位方法。首先,将声速表示为未知声源位置的函数,逼近风场中的声速场分布,然后将其代入TDOA (Time Differences of Arrival)算法中,构建非线性超定方程组,最后采用粒子群优化算法求解声源位置。对不同风速、不同声源位置及不同测试区域进行仿真,结果表明:修正后的定位精度比修正前有明显提高,尤其对于大范围并且声源靠近测试区域边缘位置的定位系统,改善更加明显;4个节点的定位系统实验结果表明,修正后的定位误差可降至修正前的4l%,该方法能更好的应用于风场中的定位系统。      

格林函数解卷积处理的阵不变量浅海无源定位

水下声源无源定位是声呐技术重要的研究方向.针对水下声源无源定位问题,本文提出了一种基于格林函数解卷积处理的阵不变量无源定位方法.该方法使用盲解卷积算法从水平阵接收信号中提取时域格林函数,然后采用空域解卷积方法处理得到的时域格林函数,获得波束时间偏移,从波束时间偏移中计算得到阵不变量,解算目标距离,从而实现声源定位.区别...     

Boundary element model for simulating sound propagation and source localization within the lungs

An acoustic boundary element (BE) model is used to simulate sound propagation in the lung parenchyma. It is computationally validated and then compared with experimental studies on lung phantom models. Parametric studies quantify the effect of different model parameters on the resulting acoustic field within the lung phantoms. The BE model is then coupled with a source localization algorithm to predict the position of an acoustic source within the phantom. Experimental studies validate the BE-based source localization algorithm and show that the same algorithm does not perform as well if the BE simulation is replaced with a free field assumption that neglects reflections and standing wave patterns created within the finite-size lung phantom. The BE model and source localization procedure are then applied to actual lung geometry taken from the National Library of Medicine's Visible Human Project. These numerical studies are in agreement with the studies on simpler geometry in that use of a BE model in place of the free field assumption alters the predicted acoustic field and source localization results. This work is relevant to the development of advanced auscultatory techniques that utilize multiple noninvasive sensors to construct acoustic images of sound generation and transmission to identify pathologies.     

Low latency localization of multiple sound sources in reverberant environments

Sound source localization algorithms determine the physical position of a sound source in respect to a listener. For practical applications, a localization algorithm design has to take into account real world conditions like multiple active sources, reverberation, and noise. The application can impose additional constraints on the algorithm, e.g., a requirement for low latency. This work defines the most important constraints for practical applications, introduces an algorithm, which tries to fulfill all requirements as good as possible, and compares it to state-of-the-art sound source localization approaches.     

利用宽带引导声源重构拷贝声场的目标声源定位

为了避开传统的匹配场目标定位技术对环境先验知识的依赖性,提出了在均匀浅海环境中只知道少量环境参数的情况下,利用垂直接收阵和不同距离上的两枚宽带引导声源重构声场对目标声源进行定位的一种方法。这种方法主要基于简正波估计和声场重构两种关键技术,同时省去了匹配场定位技术中大量的拷贝声场计算。数值仿真主要采用线性Bartlett匹配处理器分析了目标定位效果,在信噪比高于10 d B的情况下,定位效果良好。     

Measurement and modeling of the acoustic field near an underwater vehicle and implications for acoustic source localization

The performance of traditional techniques of passive localization in ocean acoustics such as time-of-arrival (phase differences) and amplitude ratios measured by multiple receivers may be degraded when the receivers are placed on an underwater vehicle due to effects of scattering. However, knowledge of the interference pattern caused by scattering provides a potential enhancement to traditional source localization techniques. Results based on a study using data from a multi-element receiving array mounted on the inner shroud of an autonomous underwater vehicle show that scattering causes the localization ambiguities (side lobes) to decrease in overall level and to move closer to the true source location, thereby improving localization performance, for signals in the frequency band 2-8 kHz. These measurements are compared with numerical modeling results from a two-dimensional time domain finite difference scheme for scattering from two fluid-loaded cylindrical shells. Measured and numerically modeled results are presented for multiple source aspect angles and frequencies. Matched field processing techniques quantify the source localization capabilities for both measurements and numerical modeling output.     

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°.     

海洋声学环境下水中声源的时延估计法定位精度分析

水中声源的定位精度受到海洋声学环境的重要影响。结合海上试验的实际应用,分析了水下观测平台采用时延估计法对声源的定位精度问题。根据理论分析,计算了时延估计误差、海洋中声速不均匀、平台非稳性、及声传播起伏等因素引起的俯仰角和方位角误差。利用误差传递公式,获得了上述因素引起的不同平台深度下,不同距离声源的定位误差。比较了采用平面阵与立体阵、是否补偿声线弯曲效应等条件下定位误差的变化,并通过海上试验结果进行了部分验证。研究结果表明,海洋声速不均匀对定位误差的贡献最大。采用立体阵代替平面阵、测量海洋声速剖面并补偿声线弯曲引起的定位误差,在1000m距离上可使定位相对误差从最大30%降低到约10%,有效提高了较远距离上的定位精度。研究结果对于采取措施提高水中声源的定位精度有指导意义。      

The effect of multimicrophone noise reduction systems on sound source localization by users of binaural hearing aids

This paper evaluates the influence of three multimicrophone noise reduction algorithms on the ability to localize sound sources. Two recently developed noise reduction techniques for binaural hearing aids were evaluated, namely, the binaural multichannel Wiener filter (MWF) and the binaural multichannel Wiener filter with partial noise estimate (MWF-N), together with a dual-monaural adaptive directional microphone (ADM), which is a widely used noise reduction approach in commercial hearing aids. The influence of the different algorithms on perceived sound source localization and their noise reduction performance was evaluated. It is shown that noise reduction algorithms can have a large influence on localization and that (a) the ADM only preserves localization in the forward direction over azimuths where limited or no noise reduction is obtained; (b) the MWF preserves localization of the target speech component but may distort localization of the noise component. The latter is dependent on signal-to-noise ratio and masking effects; (c) the MWF-N enables correct localization of both the speech and the noise components; (d) the statistical Wiener filter approach introduces a better combination of sound source localization and noise reduction performance than the ADM approach.     

Matched localization of a horizontal array in shallow sea

I.IntroductionInsha1lowsea,passivesource1ocalizationisadifficu1t,yctinterestingprob1emthathasre-ceivcdagrcatdca1ofattcnti0ninthelastfCwycars.l'-'1Theconventiona1sourccloca1izationistOmatchthercceivcdacousticdatawiththesoundficldprcdictcdbythepropagationmodcl.Thistechniqueiscallcdmatchcd-ficldproccssing(MFP).Itiscommonlyacceptcdthatinshal-low-watercnvironmcnLssoundficldcanbcwclldcscribcdbyanormal-modemodel.Accord-ingtothismodcl,thcacousticprcssurecanbecxprcsscdasalincarcombinationofthenorma…