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.     

The time domain response of some systems for sound reproduction

The perception in a listener of the existence of a “virtual” source of sound at a prescribed spatial position can be produced by ensuring that the acoustic signals at the listener's ears faithfully replicate those that would be produced by a “real” source at the same position. When loudspeakers are used to transmit the signals, it is necessary to pass the signals intended for presentation at the listener's ears through a matrix of filters that provide the inverse of the matrix of transfer functions that relates the loudspeaker input signals to the listener's ear signals. The characteristics of such filter matrices are profoundly influenced by the conditioning of the matrix to be inverted. This filter design problem is reviewed here by representing the loudspeakers as simple point monopole sources the head of the listener as a rigid sphere. The case of a virtual acoustic imaging system that uses two loudspeakers in order to reproduce the signals at the two ears is first described in some detail and previous work is reviewed. It is confirmed that the time domain response of the reproduced field is of long duration at frequencies where the inversion problem is ill-conditioned. The influence of the presence of the listener's head on this time domain behaviour is also evaluated. The principle is then extended to four input-four output reproduction systems and the computational model is used to explain some previous experimental observations. Finally, the conditioning of five input-four output systems is also examined and shown to have some potentially desirable characteristics.     

高精度球面阵聚焦声源定位方法研究

提出了一种高精度高空间分辨率球面阵聚焦声源定位方法——虚拟源法。该方法通过球面阵波束扫描获得实际声源的空间聚焦谱,并假定各扫描点为虚拟声源,将实际声源聚焦谱看作是全体虚拟源共同作用的结果,由此得到各虚拟源对声场的贡献量,从而可实现声源精确定位。仿真研究分析了频率,阵列孔径,声场模态阶数,信噪比等参数对声源定位性能的影响,并与常规算法进行对比。结果显示,该方法不受频率和阵列孔径的限制,避免了空间“混淆”,能够进行高精度高分辨率声源定位,并具有良好的背景噪声抑制能力。      

Localization of multiple sound sources with two microphones

This paper presents a two-microphone technique for localization of multiple sound sources. Its fundamental structure is adopted from a binaural signal-processing scheme employed in biological systems for the localization of sources using interaural time differences (ITD). The two input signals are transformed to the frequency domain and analyzed for coincidences along left/right-channel delay-line pairs. The coincidence information is enhanced by a nonlinear operation followed by a temporal integration. The azimuths of the sound sources are estimated by integrating the coincidence locations across the broadband of frequencies in speech signals (the "direct" method). Further improvement is achieved by using a novel "stencil" filter pattern recognition procedure. This includes coincidences due to phase delays of greater than 2pi, which are generally regarded as ambiguous information. It is demonstrated that the stencil method can greatly enhance localization of lateral sources over the direct method. Also discussed and analyzed are two limitations involved in both methods, namely missed and artifactual sound sources. Anechoic chamber tests as well as computer simulation experiments showed that the signal-processing system generally worked well in detecting the spatial azimuths of four or six simultaneously competing sound sources.     

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。      

浅海复杂环境下等效声速剖面的构建方法*

针对浅海复杂环境声速剖面水平变化情况下的声传播损失预报及目标定位问题,提出了一种基于遗传算法的等效声速剖面重构算法。首先,将声速剖面进行时间和空间上的分解,从而将声速剖面抽象为对声速剖面前三阶正交函数系数的反演;其次,利用遗传算法,以先验声速剖面集为基础,进行参数反演。仿真结果表明,在浅海复杂条件下,传播损失预报受声速剖面及海洋参数的影响,不能直接运用接收或发射位置处的声速剖面进行传播损失预报,否则会对预报结果造成误差。通过构建具有声传播累积效应的等效声速剖面可以提高匹配场定位精度,完成目标定位,且在构建等效声速剖面时,接收位置处即本地声速剖面所占权重较大。     

Three-dimensional acoustic sound field reproduction based on hybrid combination of multiple parametric loudspeakers and electrodynamic subwoofer

Auditory Mixed Reality (MR) systems that reproduce Three-Dimensional (3-D) acoustic sound fields have recently become a research focus because the combination of visual and auditory MR systems can achieve a greater sense of presence than conventional visual MR systems. General auditory MR systems usually use a headphone-based system with a Head-Related Transfer Function (HRTF), which is a major system for reproducing 3-D acoustic sound fields. However, the localization accuracy of sound images with a HRTF depends on the individual. On the other hand, we have already proposed a system for reproducing a 3-D acoustic sound field with parametric loudspeakers instead of headphones. The 3-D acoustic sound field reproduced by this system has achieved a highly accurate localization of sound images. However, one problem is that it is difficult to reproduce lower frequency sounds using parametric loudspeakers, which causes a poorer sound quality. We tried to accomplish a greater sense of presence for 3-D acoustic sound fields based on a hybrid combination of an electrodynamic subwoofer and the parametric loudspeakers by improving the sound quality. Sound images were formed at the target location using the parametric loudspeakers, and a lower frequency sound was compensated for by using the electrodynamic subwoofer. Subjective evaluation experiments were conducted to verify the effectiveness of the proposed system. We confirmed the improved sound quality while maintaining a higher accuracy of sound image localization by using the proposed system. We also confirmed the optimum parameters of the proposed system to achieve a greater sense of presence.     

基于声传播算子的声源定位实验模拟方法研究

声传播算子是一种高效的时域声场计算方法,它能够很方便地计算出给定系统参数下任意时刻任意位置的声场变化情况,本文采用这种方法计算所得的二维房间声场信息进行传声器阵列的声源定位仿真实验。计算结果表明,用该方法获取的阵列数据能有效地应用于阵列信号处理算法中,准确地估计出初始高斯脉冲声源的方向。声传播算子声场计算方法能为传声器阵列声源定位的实验提供方便,使得传声器阵列声源定位算法在不同混响时间的鲁棒性实验研究变得更加简捷。     

Performance of an independent planar virtual sound barrier at the opening of a rectangular enclosure

Planar virtual sound barrier systems have been used successfully to reduce noise radiation through an opening without affecting natural ventilation and lighting. However, the complexity of a fully coupled control system grows at the rate proportional to the square of the number of channels and this make the system implementation become impractical for enclosures with large openings. To reduce the system complexity, this paper proposes an independent planar virtual sound barrier, which is a multi-channel system consisting of many independent single channel active noise control systems. Each single channel system is “independent” in the sense that the control source output of the system is updated only with the signal from its own error sensor. Based on the analytical model of sound radiation through the opening of a rectangular enclosure, the transfer functions from both primary and control sources are calculated first. Then the noise reduction performance, the stability, and the convergence behavior of both fully coupled and independent planar virtual sound barrier systems are investigated. It is found that the independent system with no control output constraint becomes inherently unstable at some frequencies; however its stability can be improved by applying some control output constraint. Reducing the number of channels and the distance between secondary loudspeakers and error microphones can also increase system stability but at the cost of smaller noise reduction. When the system is inherently stable and there is no constraint on control output, the independent system can provide the same noise reduction as the fully coupled one but with faster convergence speed.     

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。      

频率对环绕声声像定位的影响

本文考虑双耳相位差的高级近似，导出了中频情况下适用的具有更普遍意义的平面环绕声声像定位公式。在低频时该式将化为通常的环绕声声像定位公式，而随着声音频率的增加，声像位置将与频率有关。将新的公式用到方型排列和棱型排列的４－４－４环绕声系统，得到了同实验相一致的结果。文中着重指出，声像随频率而变化是导致环绕声重发中侧向声像不稳定的重要在而为今后改进环绕声系统提供了理论基础。     

深海大深度声传播特性及直达声区水下声源距离估计

深海大深度声传播特性对在深海近海底进行水声目标探测和定位具有重要意义。利用一次南海中南部深海不完全声道中的脉冲声传播实验数据,分析了海底附近大深度声传播损失及脉冲多途传播特性,并根据直达波和海底-海面反射波的时延差与收发距离的关系,提出一种利用深海直达声区脉冲多途到达时间进行水下声源距离估计的方法。结果表明:当接收器深度位于南海深海海底附近而声源深度较浅时,直达声区水平宽度可达30 km,传播损失相对影区来说较小,有利于水下声源探测;直达声区的直达波与海底-海面反射波的到达时延差随着收发距离的增大单调减小,可被用于水下声源距离估计。得到水下声源的距离估计结果与实验GPS测量结果较为一致,距离估计均方误差为0.28 km。      

双耳幅值差确定声源方向的神经信息处理机理研究

神经信息系统实质上是定量系统, 应引起足够重视. 关于神经系统的定量研究方面的报道比较少见. 这一问题将会影响进一步的研究, 如双耳声音定向. 双耳定向是定量测量, 用定性分析的方法无法满足要求. 已有的生理实验发现声音输入信号强度与听觉神经的输出频率存在单调递增关系, 所以本文中声音强度的变化被简化成神经脉冲频率的变化. 本文基于圆映射和符号动力学原理, 建立了神经回路定量模型, 模型中对同侧输入回路采用兴奋性耦合, 对侧输入回路采用抑制性耦合, 并考虑神经元间突触连接的量子释放特征, 采用化学耦合模型实现连接, 用耦合系数表示神经元间的耦合程度. 采用Hodgkin-Huxley模型仿真研究听觉神经回路的输入/输出脉冲序列关系. 在已经仿真过的参数范围, 模型在输入信号变化与输出脉冲频率变化间存在单调递增/递减的关系. 对于单输入单输出的神经元, 采用符号动力学方法进行符号化; 对于多输入单输出的神经元, 采用分析各输出脉冲的产生时间, 判断其变化位置, 从神经脉冲序列中得到对应的两耳声音幅值差变化, 以此定位声源. 随着输出脉冲数的增加, 符号序列的长度增加, 符号序列对输入信号变化敏感, 能够得到较高的测量精度. 仿真结果表明这个模型是定量的, 神经脉冲序列能够区分信号的大小.     

Robustness to head misalignment of virtual sound imaging systems

When binaural sound signals are presented with two loudspeakers, the listener's ears are required to be in the relatively small region which is under control of the system. Misalignment of the head results in inaccurate synthesis of the binaural signals. Consequently, directional information associated with the acoustic signals is inaccurately reproduced. When the two loudspeakers are placed close together, the spatial rate of change of the generated sound field is much smaller than that generated by two loudspeakers spaced apart. Therefore, the performance of such a system is expected to be more robust to misalignment of the listener's head. Robustness of performance is investigated here with respect to head displacement in three translational and three rotational directions. A comparison is given between systems consisting of two loudspeakers either placed close together or spaced apart. The extent of effective control with head displacement and the resulting deterioration in directional information is investigated in the temporal and spectral domain by analyzing synthesized binaural signals. Subjective localization experiments are performed for cases in which notable differences in performance are expected from the previous analysis. It is shown that the system comprising two loudspeakers that are close together is very robust to misalignment of the listener's head.     

采用听觉滤波器的宽带MUSIC声源定位方法

在分析了采用短时傅里叶变换的宽带MUSIC声源定位算法(SF-MUSIC)存在问题的基础上,提出了一种采用听觉滤波器的宽带MUSIC声源定位算法(AF-MUSIC)。该算法使用听觉滤波器组对传声器阵列接收到的信号进行不等带宽分解后,在各个频率通道上使用MUSIC算法进行声源定位,并结合子区间频数估计法得出最终定位结果。对算法进行的实验评估表明,在不同声源类型条件下,相比SF-MUSIC算法,AF-MUSIC算法的平均估计误差减少2.5479°,有效地提高了声源波达方向估计的精度。      

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.     

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.      

一种双正交心音小波的构造方法

为了提高小波分析在心音信号处理中的性能, 在分析小波构造理论的基础上, 构造了一种专门用于心音信号处理的小波基. 首先提出一种构造滤波器长度为偶数的紧支撑双正交小波的一般方法; 然后根据心音信号的特点, 讨论心音小波的构造原则和一种基于心音 小波族的心音信号合成模型, 并且在此基础上构造出心音小 波. 为了突出使用心音小波处理心音信号的先进性和实用性, 对心音小波进行了比较全面的理论和数值仿真分析. 实验结果表明, 相比常用的db, bior系列小波, 运用心音小波对心音信号进行处理, 能够获得更好的去噪效果、 更精确的心音分类信息以及更小的重构误差率, 为心音特征提取和身份识别的深入研究提供了一种新方法, 在表征心音个体特征的细节方面具有积极的意义. 本文根据应用对象设计专用小波的方法也为工程应用中小波基的选择提供了一种新途径. 关键词： 双正交小波 心音小波 构造方法 心音合成模型     

Effectiveness of T-shaped acoustic resonators in low-frequency sound transmission control of a finite double-panel partition

Double-panel partitions are widely used for sound insulation purposes. Their insulation efficiency is, however, deteriorated at low frequencies due to the structural and acoustic resonances. To tackle this problem, this paper proposes the use of long T-shaped acoustic resonators in a double-panel partition embedded along the edges. In order to facilitate the design and assess the performance of the structure, a general vibro-acoustic model, characterizing the interaction between the panels, air cavity, and integrated acoustic resonators, is developed. The effectiveness of the technique as well as the optimal locations of the acoustic resonators is examined at various frequencies where the system exhibits different coupling characteristics. The measured optimal locations are also compared with the predicted ones to verify the developed theory. Finally, the performance of the acoustic resonators in broadband sound transmission control is demonstrated.     

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。