Adaptive wave field synthesis for active sound field reproduction: experimental results

Sound field reproduction has applications in music reproduction, spatial audio, sound environment reproduction, and experimental acoustics. Sound field reproduction can be used to artificially reproduce the spatial character of natural hearing. The objective is then to reproduce a sound field in a real reproduction environment. Wave field synthesis (WFS) is a known open-loop technology which assumes that the reproduction environment is anechoic. The room response thus reduces the quality of the physical sound field reproduction by WFS. In recent research papers, adaptive wave field synthesis (AWFS) was defined as a potential solution to compensate for these quality reductions from which WFS objective performance suffers. In this paper, AWFS is experimentally investigated as an active sound field reproduction system with a limited number of reproduction error sensors to compensate for the response of the listening environment. Two digital signal processing algorithms for AWFS are used for comparison purposes, one of which is based on independent radiation mode control. AWFS performed propagating sound field reproduction better than WFS in three tested reproduction spaces (hemianechoic chamber, standard laboratory space, and reverberation chamber).     

A wave field synthesis approach to reproduction of spatially correlated sound fields

This article discusses an open-loop wave field synthesis (WFS) approach for the reproduction of spatially correlated sound fields. The main application concerns laboratory reproduction of turbulent boundary layer wall pressure on aircraft fuselages and measurement of their sound transmission loss. The problem configuration involves reconstruction of random sound pressure distributions on a planar reproduction surface using a planar array of reproduction monopoles parallel to the reproduction plane. In this paper, the WFS formulation is extended to sound fields with imposed time and spatial correlation properties (or equivalently imposed cross-spectral density in the frequency and wave number domains). Numerical examples are presented for the reproduction of a propagating plane wave, diffuse acoustic field and wall pressure in subsonic or supersonic turbulent boundary layers. The reproduction accuracy is examined in terms of the size of the source plane and reproduction plane, their separation, and the number of reproduction sources required per acoustic wavelength. While the reproduction approach cannot reconstruct sub-wavelength correlation scales of subsonic turbulent boundary layers, it effectively reconstructs correlation scales larger than the acoustic wavelength, making it appropriate for diffuse acoustic field and supersonic turbulent layers.     

自适应宽带有源消声

实际中存在的噪声,一般都是窄带或有色宽带噪声(简称宽带噪声),而宽带噪声更为普遍.为使宽带自适应有源消声(AANC)得到实际应用,必须保证宽带AANC系统具有良好的稳定性和较高的降噪量.为此,本文对AANC系统稳态性能作了理论分析和数值计算,得到了系统降噪量对噪声带宽、空间声传播通道、自适应滤波器参数等的依赖关系;以自由声场远场AANC为例,从声学角度对AANC系统作了物理解释,从而为改进宽带AANC系统性能提供理论依据.     

An adaptive digital filter for broadband active sound control

The application of digital filtering to active sound control systems has increased their flexibility and reliability thus enabling optimum devices to be produced. A further improvement is possible: the systems can be made adaptive so that optimum performance is maintained. An algorithm is presented for using the signal from a monitoring microphone to adjust the controller's characteristic. The optimum characteristic is obtained quickly due to the rapid convergence of the algorithm. The scheme has been used to control the sound transmitted in a wind tunnel which had a flow with a Mach number of 0·1. The adaptive scheme was able to cope with large variations in the flow velocity and consequent changes in the characteristic required to maintain optimum performance. The attenuation in sound level achieved in the duct came within 2 dB of the theoretical maximum (which is limited by coherence) over more than two octaves of broadband sound.     

An algorithm for designing a broadband active sound control system

Hitherto the filtering techniques used in active sound control systems have been analogue: these are rapidly becoming obsolete because of their lack of stability and versatility. The microprocessor gives digital techniques an edge both on grounds of convenience and economics; moreover the performance of digital filtering is far superior to that of analogue methods. An algorithm is presented which can be used to assess the optimum filter characteristics required for active sound control systems with a single degree of freedom. In the process the system is subjected to three random noise tests which directly yield the characteristics of the filter. The algorithm has been tried out in practical applications and shown to be both quick and convenient to use.     

A demonstration of active control of broadband sound

This paper describes a short experiment undertaken to demonstrate how easily broad-band active control of sound can be used to tackle real industrial problems. Ten decibles of the low frequency sound entering an anechoic chamber through a lobby was blocked by using a single degree of freedom system. The experiment was set up, tested and demonstrated in a day.     

基于声场重建的材料吸声系数测量方法

针对现有方法对材料吸声系数进行现场测量时存在低频测量误差大的问题,本文提出了一种利用扬声器线阵列对材料吸声系数进行现场测量的新方法。该方法使用基于能量比值约束的最小二乘法在待测材料表面进行平面波声场重建并结合双传声器传递函数法对材料的吸声系数进行测量。数值仿真表明在100～1600 Hz频率范围内,新方法在未加约束时能够对材料的吸声系数进行准确测量。在半消声室中利用新方法测量了三聚氰胺泡沫的吸声系数,分析了能量比值约束值对测量结果的影响,并和阻抗管以及其它两种现场测量方法的测量结果进行了对比。结果表明该方法能够对吸声材料在160～1600 Hz频段内的吸声系数进行准确测量,并且相较于现存的现场测量方法,新方法具有更低的测量频率下限。     

混响环境下目标声场稀疏分解多域声场重放方法

对于封闭空间内的声场重放,传统的多点声压匹配方法 (Cov-PM)依据在目标声场测得的声压直接使用最小二乘来计算扬声器权重。然而这种方法要求较多的目标声场采样点以实现足够的精度。对于一类特殊的声场景,也就是目标声场是由少量声源辐射产生时,提出一种对目标声场稀疏分解的方法来进行混响环境下多域声场重放以降低对目标声场采样数量的要求。该文给出基于目标声场等效源稀疏分解多域重放方法(Sparse-ESM)理论推导,通过数值计算以及实验测试两种方式对比所提方法与最小二乘等效源分解方法以及Cov-PM的声场重放性能。数值结果表明,在600 Hz以上的频段,Sparse-ESM方法的重放误差性能提升明显。实验结果也得出了与数值计算相同的结论。同时,还通过数值计算和实验测试两种方法证明了当目标声场声源方向波动时,Sparse-ESM仍然可以保持与其余两类方法相近的声对比度,并且实现较高的亮区重放精度。     

Three dimensional active sound field cancellation

In this paper, a novel method for Three dimensional active sound field cancellation is proposed. The optimum sound cancellations in different situations may be obtained. An array whose elements lie on a set of rings placed on the surface of a scattering object is considered as the device. It is quite possible to cancel the scattered sound field to any arbitrary level over either the whole space or a partial area of interest, as long as the number of the array elements is sufficient.     

采用无耳壳头相关传输函数的虚拟声信号处理

本文提出采用无耳壳、封闭耳道的头相关传输函数（HRTF）进行虚拟声信号处理的方法。理论分析表明，在包含功率均衡的扬声器虚拟声信号处理中，采用无耳壳、封闭耳道的头相关传输函数可有效地减少信号处理函数的谷点，从而减少声重发的音色改变。心理声学的实验结果表明，新处理方法的声像定位与传统的带耳壳、耳道的HRTF处理无显著区别，但音色较后者有所改善。因而新的方法更适合实际的虚拟声信号处理应用。     

Adaptive beamforming for array signal processing in aeroacoustic measurements

Phased microphone arrays have become an important tool in the localization of noise sources for aeroacoustic applications. In most practical aerospace cases the conventional beamforming algorithm of the delay-and-sum type has been adopted. Conventional beamforming cannot take advantage of knowledge of the noise field, and thus has poorer resolution in the presence of noise and interference. Adaptive beamforming has been used for more than three decades to address these issues and has already achieved various degrees of success in areas of communication and sonar. In this work an adaptive beamforming algorithm designed specifically for aeroacoustic applications is discussed and applied to practical experimental data. It shows that the adaptive beamforming method could save significant amounts of post-processing time for a deconvolution method. For example, the adaptive beamforming method is able to reduce the DAMAS computation time by at least 60% for the practical case considered in this work. Therefore, adaptive beamforming can be considered as a promising signal processing method for aeroacoustic measurements.     

Adaptive signal processing for reducing nearby generated flow noise

By the acoustic holography method, it was found that there were multiple nearby noise sources, which are hard to be rejected by using the conventional beamforming technique. Instead of the conventional beamforming, we propose a new adaptive beamforming method to suppress nearby generated flow noise by using sonar sub-arrays.     

宽带波束形成器的自适应综合

提出一种具有频率不变波束图的宽带基阵设计方法。该方法只需给出参考波束图,无须推导权矢量的表达式,便可由计算机自动完成整个设计任务。其设计过程分为3步:(1)用数值计算的方法计算参考频率下的基阵权矢量。(2)利用自适应模拟法得到其它频率处的基阵权矢量。(3)以各频点处的基阵权矢量作为FIR滤波器的频率响应指标设计FIR滤波器。该方法可应用于任意阵型的阵列,且对各阵元的指向性也没有任何限制,因此本文方法具有重要的工程应用价值。     

Down-mixing of multi-channel audio for sound field reproduction based on spatial covariance

This article describes a method for automatic down-mixing multi-channel audio content on the basis of spatial covariance. Such a down-mixing method should be able to convert the signal of the original multi-channel audio system into that for an alternative system with the lesser number of channels, while maintaining the spatial impression of sound. Moreover, it should take into account the listener’s position and transfer function. Wave surface control and convolving the head related transfer function are techniques used in sound field control or reproduction. We consider that the spatial impressions of a sound field, which we perceive through our ears, are reproduced by preserving the relative relationship between observation points even if the wave surface is not completely controlled. Takahashi et al. proposed a new sound field reproduction method that we named “SOund field Reproduction based on sPAtial Covariance” (SORPAC). SORPAC can control the point-to-point covariance in a sound field. We expect that this sound field reproduction method based on spatial covariance can be applied to down-mixing of multi-channel content because SORPAC does not require the listener’s position or transfer function. This article describes SORPAC and its characteristics. We used SORPAC for down-mixing audio content. We confirmed that SORPAC-based down-mixing could accurately reproduce the interaural cross correlation (IACC) in relation to the listener’s position.     

基于声场复现的有源噪声控制支撑技术*

有源噪声控制是一种有广泛用途的低频噪声控制技术,目前已在部分场合取得商业化应用,然而该技术的大规模应用仍然面临诸多困难,声场复现技术为解决其中的关键问题提供了有效手段。该文论述了声场复现的两种实现方式(声场重构和声场再现)的基本原理、国内外现状及其在有源控制技术开发中的应用,尤其是基于声压匹配法的初级声场重构、基于球阵的声场再现以及电声器件布放优化设计。     

Study of demagnetizing field induced by a sound wave

The dynamic demagnetizing factor N _d used to describe perturbation of magnetization of a magnetic fluid by a sound wave has been studied. Experimental N _d values obtained according to the proposed methodology have been compared with the results of a model theory based on the approximating function for bodies of cylindrical shape and oblate ellipsoids of rotation.     

空间声场有源声吸收中的多极源结构

本文从声波基本方程导出了空间声场的有源声吸收能量公式,讨论了单极子,偶极子,三极子等多极次级源对初级声场的有源声吸收及相应的实现条件。在此基础上提出偶极次级源是较简单的次级源结构。在消声室的实验结果表明它具有较好的声吸收性能,为其应用作了初步的可行性研究。     

波场合成与波场分析的有源房间补偿方法

波场合成是一种空间声重放技术,利用扬声器阵列在宽阔的听音区域内重建声场。为消除或者消减听音房间的反射对重建波场的影响,利用测量环绕听音区域的闭合曲线上声压和声压梯度来分析波场,推导了基于圆形阵列进行波域分解的公式,利用多通道逆滤波进行了平面波域的房间补偿,实验结果显示该算法在整个听音区域内都是有效的。与传统补偿方法相比,边界元法所需的测量传声器数目少计算复杂度低,而波域分解具有更充分的波场分析能力,因此是一种更有效的有源房间补偿方法.