Ambisonics声捡拾与重放音色的双耳听觉模型分析与实验

提出用双耳听觉模型对空间声音色进行分析的普遍方法,并以Ambisonics为例进行了分析。Ambisonics是基于物理声场重构的空间声系统,其最终重构声场误差以及音色改变是由传声器捡拾和重放空间混叠误差共同引起的。采用修正的Moore双耳响度模型计算了Ambisonics重构声场的双耳响度级谱并和目标声场的情况比较,从而定量评价重构声场的音色改变。结果表明,在理想捡拾信号的情况下,无音色改变重放的上限频率和区域大小随Ambisonics的阶数而增加。而对于传声器阵列捡拾的情况,只要阵列的上限频率大于Ambisonics重放的上限频率,在重放的上限频率以下,传声器阵列空间混叠误差对最终重构声场及其感知音色的影响就可以忽略。在此基础上,提出了一种综合考虑捡拾与重放性能的Ambisonics系统优化设计方法。心理声学实验得到了和双耳听觉模型一致的结果,从而也验证了模型分析的有效性。      

Analysis with binaural auditory model and experiment on the timbre of Ambisonics recording and reproduction

A scheme for analyzing the timbre in spatial sound with binaural auditory model is proposed and the Ambisonics is taken as an example for analysis.Ambisonics is a spatial sound system based on physical sound field reconstruction.The errors and timbre colorations in the final reconstructed sound field depend on the spatial aliasing errors on both the recording and reproducing stages of Ambisonics.The binaural loudness level spectra in Ambisonics reconstruction is calculated by using Moore's revised loudness model and then compared with the result of real sound source,so as to evaluate the timbre coloration in Ambisonics quantitatively.The results indicate that,in the case of ideal independent signals,the high-frequency limit and radius of region without perceived timbre coloration increase with the order of Ambisonics.On the other hand,in the case of recording by microphone array,once the high-frequency limit of microphone array exceeds that of sound field reconstruction,array recording influences little on the binaural loudness level spectra and thus timbre in final reconstruction up to the highfrequency limit of reproduction.Based on the binaural auditory model analysis,a scheme for optimizing design of Ambisonics recording and reproduction is also suggested.The subjective experiment yields consistent results with those of binaural model,thus verifies the effectiveness of the model analysis.     

Implementation issues of the nearfield equivalent source imaging microphone array

This paper revisits a nearfield microphone array technique termed nearfield equivalent source imaging (NESI) proposed previously. In particular, various issues concerning the implementation of the NESI algorithm are examined. The NESI can be implemented in both the time domain and the frequency domain. Acoustical variables including sound pressure, particle velocity, active intensity and sound power are calculated by using multichannel inverse filters. Issues concerning sensor deployment are also investigated for the nearfield array. The uniform array outperformed a random array previously optimized for far-field imaging, which contradicts the conventional wisdom in far-field arrays. For applications in which only a patch array with scarce sensors is available, a virtual microphone approach is employed to ameliorate edge effects using extrapolation and to improve imaging resolution using interpolation. To enhance the processing efficiency of the time-domain NESI, an eigensystem realization algorithm (ERA) is developed. Several filtering methods are compared in terms of computational complexity. Significant saving on computations can be achieved using ERA and the frequency-domain NESI, as compared to the traditional method. The NESI technique was also experimentally validated using practical sources including a 125 cc scooter and a wooden box model with a loudspeaker fitted inside. The NESI technique proved effective in identifying broadband and non-stationary sources produced by the sources.     

Real-time near-field acoustic holography for continuously visualizing nonstationary acoustic fields

Near-field acoustic holography (NAH) is an effective tool for visualizing acoustic sources from pressure measurements made in the near-field of sources using a microphone array. The method involving the Fourier transform and some processing in the frequency-wavenumber domain is suitable for the study of stationary acoustic sources, providing an image of the spatial acoustic field for one frequency. When the behavior of acoustic sources fluctuates in time, NAH may not be used. Unlike time domain holography or transient method, the method proposed in the paper needs no transformation in the frequency domain or any assumption about local stationary properties. It is based on a time formulation of forward sound prediction or backward sound radiation in the time-wavenumber domain. The propagation is described by an analytic impulse response used to define a digital filter. The implementation of one filter in forward propagation and its inverse to recover the acoustic field on the source plane implies by simulations that real-time NAH is viable. Since a numerical filter is used rather than a Fourier transform of the time-signal, the emission on a point of the source may be rebuilt continuously and used for other post-processing applications.     

一种用于手持式播放装置的立体声扩展方法

手持式播放装置的一对立体声扬声器相对倾听者的张角窄、距离近,影响立体声重放效果。针对这类应用提出了一种立体声扩展方法,它采用远场和近场头相关传输函数(HRTF)设计听觉传输滤波器并将低频信号滤除。该方法适应了近距离扬声器重放并避免了窄张角扬声器布置带来的信号处理中过分的低频提升问题。该方法不但能改善重放性能,且对计算和存储量要求较低,适合于手持式播放装置的信号处理硬件配置。声像定位实验验证了该方法的效果。     

采用分布式球形阵列的球谐波域声场分离方法

为从测得的混合声场信号中提取出需要的目标声场,提出一种基于分布式球形传声器阵列的声场分离方法。该方法依据声场的球谐波分解,利用阵列各传声器采集到的声场声压信号,获得目标声场与干扰声场的球谐波展开系数,进而估计目标声场。该方法利用声场以不同中心展开的球谐波系数之间的变换关系,直接建立传声器测量声压信号与整体坐标系下声场展开系数的方程,与传统的分布式球形阵列声场分离方法,即先求解局部坐标系下声场展开系数,再变换为整体展开系数的方法,进行比较。分别通过数值仿真和实验说明了提出方法的有效性。结果表明:该方法能够从混合声场中较准确地估计出目标声场,并且在干扰声场能量增大时,保持了较小的声场估计误差,相比于传统方法误差增加更少。     

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。     

水平阵信号压缩感知用于简正波分离

针对水平阵信号简正波分离过程中常规波束形成分辨率低以及warping模态滤波不适用于复杂声信号的问题,提出水平阵信号压缩感知用于简正波分离的方法。利用压缩感知在方位估计中的高分辨特性,通过估计水平阵接收信号在频率方位角上的二维分布,分离得到各阶简正波的方位谱,并逆Fourier变换得到时域波形。仿真孔径1 km、阵元间隔10 m水平阵接收20~200 Hz伪随机声信号和脉冲声信号,所提方法分离出的各阶简正波与理论波形的相关系数在0.97~1.0。对2011年北黄海声学实验中的海底28元水平阵接收的气枪信号,在合成至1 km孔径后使用压缩感知方法分离简正波,其与warping模态滤波分离得的前5阶简正波相关系数在0.82~0.93。仿真与实验都说明了水平阵信号压缩感知简正波分离方法的有效性。      

基于 L-曲线参数优化的均匀声场重建算法

为了得到最小二乘法声场重建问题的稳定解,通常需要引入Tikhonov正则化方法。然而正则化程度取决于正则化参数的选择。针对这一问题,提出了一种基于L-曲线法参数选择的均匀声场重建算法。该算法根据重建误差与扬声器功率计算得到L-曲线,该曲线上曲率最大的点所对应的参数值作为Tikhonov正则化参数的选值。确定正则化参数后可进一步得到扬声器权系数以及重建均匀声场。针对不同正则化参数取值方法,对控制区域进行均匀声场重建以及重建性能仿真。仿真结果及实验表明,L-曲线法实现了重建误差与扬声器驱动信号功率之间的平衡。     

两扬声器虚拟声重放的双耳声压控制与定位性能

在两扬声器虚拟声重放中,通过精确重构双耳声压而产生不同的空间听觉感知。其重放的定位性能应该是由双耳声压控制的代价和稳定性所共同决定的。过去研究主要对双耳声压控制的稳定性进行分析,并以此作为扬声器布置和信号处理的依据。该文研究表明仅对双耳声压的稳定性分析是不足以完全衡量扬声器虚拟声重放的定位性能的。进一步采用虚拟声信号处理滤波器响应平均功率对双耳声压控制的代价进行分析。结果表明,缩窄左右对称扬声器布置的张角或采用非对称扬声器布置会明显增加产生侧向目标虚拟源时的双耳声压控制代价。虚拟源(虚拟声像)定位实验表明,双耳声压控制代价增加会引起虚拟源定位缺陷。实际应用中,为了有效产生侧向虚拟源,应避免采用过窄张角(如立体声偶极)和非对称的扬声器布置。     

利用水平阵模态域波束形成判别声源深度*

针对近水面声源和水下声源的深度判别问题,根据近水面声源难以激发低阶模态的物理现象,研究利用声源波数谱结构和波数位置的不同来分辨近水面声源和水下声源。通过采用MVDR的谱估计方法进行模态域波束形成,补偿水平阵各阵元之间各号简正波的相位差,获得主瓣窄、旁瓣低的声源信号波数谱。波数谱的波数位置与频率呈近似线性关系,水中声速剖面、海底参数、海深都会影响波数谱的具体结构和位置。此外,声源信号的到达角估计误差同样也会影响波数谱主瓣的位置估计。数值仿真结果表明,在浅海负跃层声速剖面条件下,可利用水平阵模态域波束形成判别声源深度,区分近水面声源和水下声源。     

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。      

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.     

The horizontal directivity of noise radiated by a rail and implications for the use of microphone arrays

Structural waves propagating along a railway rail form an extended source of sound radiation. Using an equivalent source model the distribution of this sound in a horizontal plane is investigated and shown to consist mainly of sound propagation at a particular angle to the normal. This direction is determined by the ratio of the wavenumbers in the rail and in air. Due to the extended nature of the rail as a source, the spatial distribution of the sound field in the direction along the track does not lend itself to the use of a simple directivity factor. The consequences for the measurement of noise from the rail using a microphone array are then explored. It is shown that a microphone array focussed normal to the rail does not detect most of the sound radiated by the rail. By turning the focus angle, the sound detected becomes a maximum when this angle corresponds to the angle of propagation of the sound radiation. Measurements on a test track using artificial excitation and measurements during the passage of a train confirm these conclusions.     

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.     

利用宽带声场频率-掠射角干涉结构的深海直达声区目标深度估计方法

针对典型深海环境中宽带声源的深度分辨问题,通过研究深海声场随频率起伏的干涉结构与垂直线阵频域波束输出图中的干涉结构,给出一种直达声区内可区分多水下目标的宽带声源深度估计方法。该方法以近水面目标的射线声场模型为基础,推导出近海面宽带声源接收声场的波束输出表达式,阐明了频域波束输出图中干涉结构与声源深度的对应关系。然后利用改进的傅里叶变换方法将二维频域波束输出图映射到声源深度-掠射角度域,可实现声源深度信息的有效分离。最后开展了深海实验验证,利用垂直阵接收拖曳声源发射的宽带白噪声信号,拖曳声源深度计算结果与实测声源深度基本一致。数值仿真与实验结果均表明该方法可以在多目标复杂环境下准确估计出水下宽带声源的深度。      

A Comparison on the Localization Performance of Static and Dynamic Binaural Ambisonics Reproduction with Different Order

Ambisonics is a series of spatial sound reproduction system based on spatial harmonics decomposition and each order approximation of sound field. Ambisonics signals are originally intended for loudspeakers reproduction. By using head-related transfer functions (HRTFs) filters, binaural Ambisonics converts the Ambisonics signals for static or dynamic headphone reproduction. In present work, the performances of static and dynamic binaural Ambisonics reproduction are evaluated and compared. The mean binaural pressure errors across target source directions are first analyzed. Then a virtual source localization experiment is conducted, and the localization performances are evaluated by analyzing the percentages of front-back and up-down confusion, the mean angle error and discreteness in the localization results. The results indicate that binaural Ambsonics reproduction with insufficiently high order (for example, 5-10 order) is unable to recreate correct high-frequency magnitude spectra in binaural pressures, resulting in degradation in localization for static reproduction. Because dynamic localization cue is included, dynamic binaural Ambisoncis reproduction yields obviously better localization performance than static reproduction with the same order. Even a 3-order dynamic binaural Ambisoncis reproduction exhibits appropriate localizations performance.     

Separation of nonstationary sound fields in the time-wavenumber domain

A number of sound field separation techniques have been proposed for different purposes. However, these techniques just consider the separation of sound fields in the space domain and are restricted to stationary sound fields. When the sound fields are nonstationary, it is also necessary to perform the separation in the time domain. Therefore, on the basis of the propagation principle of sound pressure in the time-wavenumber domain, a nonstationary sound field separation technique with two closely spaced parallel measurement surfaces is proposed. It can separate the nonstationary signals generated by the primary sources in both time and space domains when the disturbing sources exist on the other side of the measurement plane. The signals in time and space domains are separated by using the spatial Fourier transform method and the time domain deconvolution method. A simulation involving two monopoles driven by nonstationary signals demonstrates that the method proposed can remove the influence of disturbing sources in both time and space domains. The feasibility of this method is also demonstrated by an experiment with two loudspeakers located on two sides of measurement planes. Additionally, to comment more objectively on the separation results, some indicators are computed in both the simulation and experiment.     

基于波数域直接求解法的正交加筋板声辐射研究

为了研究正交加筋板的声辐射问题,基于波数域直接求解法,建立了研究正交加筋板声辐射特性的理论模型。先利用傅里叶变换法求解周期结构的声振理论模型,得到波数域中关于结构响应的无限大耦合代数方程组,采用数值方法将其截断成有限项求解,结合稳相法便可快速获得远场辐射声压。该方法对单向和正交加筋板的预测结果与现有文献中的理论结果取得了良好的吻合,验证了理论模型的准确性和可靠性;并进一步通过数值算例研究了作用点位置,加强筋间距及平板厚度对结构声辐射特性的影响。