前方空间环绕声的四扬声器虚拟重放

考虑头部转动带来的动态因素对听觉垂直定位的贡献,提出了前方空间环绕声的四扬声器虚拟重放方法。4个扬声器分别布置在水平面左前、右前以及高仰角的左前上、右前上方向,并采用听觉传输信号处理的方法将多通路空间环绕声信号转换为4个扬声器的重放信号。以9.1通路空间环绕声虚拟重放为例,采用头相关传输函数对双耳声压及其包含的定位因素进行分析表明,该方法可以产生正确的双耳时间差及其随头部转动的变化,从而产生合适的侧向定位双耳因素和垂直定位的动态因素。而心理声学实验结果表明,该方法可以重放稳定的前方空间的水平和垂直虚拟源。因此,四扬声器布置结合听觉传输处理足以重放前方空间环绕声的垂直定位信息,实现多通路空间环绕声的向下混合与简化。      

Least squares approach in wavenumber domain for sound field recording and reproduction using multiple parallel linear arrays

A novel signal processing method is proposed for sound field recording and reproduction using multiple parallel linear microphone and loudspeaker arrays. In sound field recording and reproduction, the problem is how to calculate the transfer filters that transform the signals recorded by microphones into the driving signals of the loudspeakers. The proposed method is based on the spatial Fourier transform in the horizontal angle combined with the least squares (LS) approach in the elevation angle. In the proposed method, the signals recorded by each linear microphone array and those that drive each loudspeaker array are decomposed into the wavenumber domain by the spatial Fourier transform in the horizontal direction. The transfer filters are then calculated by the LS approach in the wavenumber domain. As a result, the size of the matrix of each transfer function in the wavenumber domain is much smaller than that of the conventional LS approach in the temporal frequency domain (LSTF), and well-conditioned stable transfer filters can be obtained with low computational cost without regularization. Computer simulation results show that the proposed method reconstructed a sound field around the control points as accurately as the conventional LSTF.     

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 priori mesh grading for the numerical calculation of the head-related transfer functions

Head-related transfer functions (HRTFs) describe the directional filtering of the incoming sound caused by the morphology of a listener’s head and pinnae. When an accurate model of a listener’s morphology exists, HRTFs can be calculated numerically with the boundary element method (BEM). However, the general recommendation to model the head and pinnae with at least six elements per wavelength renders the BEM as a time-consuming procedure when calculating HRTFs for the full audible frequency range. In this study, a mesh preprocessing algorithm is proposed, viz., a priori mesh grading, which reduces the computational costs in the HRTF calculation process significantly. The mesh grading algorithm deliberately violates the recommendation of at least six elements per wavelength in certain regions of the head and pinnae and varies the size of elements gradually according to an a priori defined grading function. The evaluation of the algorithm involved HRTFs calculated for various geometric objects including meshes of three human listeners and various grading functions. The numerical accuracy and the predicted sound-localization performance of calculated HRTFs were analyzed. A-priori mesh grading appeared to be suitable for the numerical calculation of HRTFs in the full audible frequency range and outperformed uniform meshes in terms of numerical errors, perception based predictions of sound-localization performance, and computational costs.     

Sound-field reproduction in-room using optimal control techniques: simulations in the frequency domain

This paper describes the simulations and results obtained when applying optimal control to progressive sound-field reproduction (mainly for audio applications) over an area using multiple monopole loudspeakers. The model simulates a reproduction system that operates either in free field or in a closed space approaching a typical listening room, and is based on optimal control in the frequency domain. This rather simple approach is chosen for the purpose of physical investigation, especially in terms of sensing microphones and reproduction loudspeakers configurations. Other issues of interest concern the comparison with wave-field synthesis and the control mechanisms. The results suggest that in-room reproduction of sound field using active control can be achieved with a residual normalized squared error significantly lower than open-loop wave-field synthesis in the same situation. Active reproduction techniques have the advantage of automatically compensating for the room's natural dynamics. For the considered cases, the simulations show that optimal control results are not sensitive (in terms of reproduction error) to wall absorption in the reproduction room. A special surrounding configuration of sensors is introduced for a sensor-free listening area in free field.     

门控卷积神经网络多通路声学回声消除算法*

多通路声重放系统能够增强听者的现实感与空间感,但在免提通信条件下,其不可避免会受到噪声和回声干扰,严重影响通信质量。针对上述问题,本文提出了一种基于门控卷积循环神经网络的多通路声学回声消除和噪声抑制方法。该方法以传声器接收信号和重放声道的压缩复数谱为网络输入,以近端语音的压缩复数谱为网络的输出目标,直接从传声器拾取信号中恢复近端纯净语音,无需对声重放信号进行去相关处理,解决了传统自适应滤波方法中存在的非唯一解问题,同时保证了多通路声重放质量。仿真和真实声学环境实验均表明本文所提出的方法可显著消除多通路声重放系统的噪声和回声,在语音质量和回声返回衰减增益方面均优于传统算法。     

Sound field monitoring by tomographic electronic speckle pattern interferometry

Time-averaging electronic speckle pattern interferometry (ESPI) allows to record the phase modulation of light that has propagated through a sound field. When such data are collected from different projection directions the three-dimensional spatial distribution of amplitude and phase of the sound are obtained by tomographic back projection. The performance of such a setup increases with the number of projection directions, the number of effective resolution elements in the detector, and the number of recordings taken in averaging. These efforts, however, compete with the need for acceptable recording and processing times. Recent improvements in time-averaging ESPI enable even demanding applications in sound field monitoring. This is demonstrated in the design of a 38.5 kHz ultrasound source composed of a large number of individual piezoelectric transducer elements and intended to generate highly directive audio sound by nonlinear mixing in air (parametric array). The success of this method relies essentially on a non-intrusive control of the spatial homogeneity of the ultrasound field. Tomographic ESPI data have guided in a delicate alignment of the transducer elements yielding the expected narrowing of the angular radiation of the audio sound.     

A probabilistic method for analysis of sound localization performance

We propose a novel probabilistic method for quantitative analysis of the sound localization performance. The analysis is based on the two kinds of probability distributions estimated from a single dataset containing listening test results for sound localization. The quantitative parameters of the von Mises probability distributions provide meaningful interpretations on the localization performance. The mean direction represents the listener’s perceptual bias, and the shape parameters and the circular variance provide information on how much the responses are concentrated about the mean direction. The front-back confusion can be determined more systematically by the proposed method than the conventional one, especially for the responses near the boundary of front-back confusion region based on the conventional criterion. The proposed method can be easily extended to analyze the up-down and left-right confusions. To investigate the feasibility of the proposed method, the already published dataset originally obtained by Iida et al. was analyzed using the proposed probabilistic method. The results showed that the proposed method can provide meaningful and reasonable interpretations on the localization performance.     

Influence of the Number of Loudspeakers on the Timbre in Horizontal and Mixed-Order Ambisoncis Reproduction

Ambisonics is a series of flexible spatial sound reproduction systems based on spatial harmonics decomposition of sound field. Traditional horizontal and spatial Ambisonics reconstruct horizontal and spatial sound field with certain order of spatial harmonics, respectively. Both the Shannon-Nyquist spatial sampling frequency limit for accurately reconstructing sound field and the complexity of system increase with the increasing order of Ambisonics. Based on the fact that the horizontal localization resolution of human hearing is higher than vertical resolution, mixed-order Ambisonics (MOA) reconstructs horizontal sound field with higher order spatial harmonics, while reconstructs vertical sound field with lower order spatial harmonics, and thereby reaches a compromise between the perceptual performance and the complexity of system. For a given order horizontal Ambisoncis or MOA reproduction, the number of horizontal loudspeakers is flexible, providing that it exceeds some low limit. By using Moore’s revised loudness model, the present work analyzes the influence of the number of horizontal loudspeakers on timbre both in horizontal Ambisonics and MOA reproduction. The binaural loudness level spectra (BLLS) of Ambisoncis reproduction are calculated and then compared with those of target sound field. The results indicate that below the Shannon-Nyquist limit of spatial sampling, increasing the number of horizontal loudspeakers influence little on BLLS then timbre. Above the limit, however, the BLLS for Ambisoncis reproduction deviate from those of target sound field. The extent of deviation depends on both the direction of target sound field and the number of loudspeakers. Increasing the number of horizontal loudspeakers may increase the change of BLLS then timbre in some cases, but reduce the change in some other cases. For MOA, the influence of the number of horizontal loudspeakers on BLLS and timbre reduces when virtual source departs from horizontal plane to the high or low elevation. The subjective evaluation experiment also validates the analysis.     

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.     

A basic study on the sound field analysis of periodic structures by boundary integral equation method

This study deals with the development of the approximate method to analyze the sound field around equally spaced finite obstacles, using the periodic boundary condition. First, on the assumption that the equally spaced finite obstacles are the periodically arranged obstacles, the sound field is analyzed by boundary integral equation method with a Green’s function which satisfies the periodic boundary condition. Furthermore, by comparing these results and the exact solution by using the fundamental solution as Green’s function, the validity of the approximate method is also investigated. Next, in order to evaluate the applicability of the approximate method, the simple formula using some parameters, i.e., the frequency, the period, and the number of obstacles, etc., is proposed. The results of the sound field analysis applied the formula are presented.     

浅海波导环境失配下的协方差矩阵拟合稳健匹配场声源功率估计

实际浅海波导中环境噪声为相干噪声,最小方差匹配场声源功率估计方法能在相干噪声背景下准确估计声源辐射功率,但该方法受环境不确定性影响较大;此外,由于最小方差匹配场声源功率估计方法使用信号幅度作为中间量估计声源功率,信号幅度估计误差会二次放大并传递到声源功率估计结果中。本文提出一种协方差矩阵拟合稳健最小方差匹配场声源功率估计方法,该方法引入信道传递函数不确定集,结合协方差矩阵拟合思想将声源功率估计问题建模为在信道传递函数不确定集约束下对函数取极值的问题,使用Lagrange乘子法求解该问题得到信道传递函数估计值和声源辐射功率估计值。环境失配影响声源辐射功率估计性能的根本原因在于信道传递函数偏差较大,协方差矩阵拟合稳健匹配场声源功率估计方法有效减小了环境失配时信道传递函数的偏差,从而显著提升环境失配稳健性。此外,该方法使用权值直接估计声源功率,无需使用信号幅度作为中间量,避免了估计误差的传递。仿真验证了协方差矩阵拟合稳健匹配场声源功率估计方法的环境失配稳健性。     

双扬声器近场声源重放实验研究

该文针对电子器件散热用的一款变速轴流风扇的气动噪声及其降噪方法进行实验研究。首先利用风扇旋转轴等高平面内圆周分布的传感器阵列测量风扇不同转速下远场噪声分布,总声压级与转速的对数关系验证散热风扇主要气动噪声属于偶极源噪声,频谱分析显示离散单音噪声为主要噪声影响因素。基于管道声学理论的管道模态截止方法,研究进出风口安装圆形短管对风扇气动噪声的影响,实验结果显示不同位置、不同长度的短管对风扇远场噪声影响不同。额定转速下,在进风口安装2 cm管道可以使远场1 m处平均总声压级下降4.1 dB(A),降噪效果显著。模态测量结果显示,此种情况下对应离散单音处的风扇主要模态幅值大大降低,风扇离散单音噪声降低从而噪声总声压级大幅减小。该方法为散热风扇降噪提供了一种新的途径。     

Inter-channel de-correlation for perceptual audio coding

Sound reproduction is evolving towards multi-channel systems with a growing number of channels. Consequently, high quality multi-channel codecs are required. Last generation perceptual audio codecs, represented by MPEG advanced audio coder (AAC), can efficiently code typical surround multi-channel material but may benefit from a previous block of inter-channel de-correlation, for certain audio recordings. These recordings present high correlation between non-symmetric channels, or correspond to loudspeaker arrangements different from surround, such as linear loudspeaker arrays. In order to valuate the advantages of this new coding approach, a multi-channel perceptual quality measure is developed.     

Beamforming synthesis of binaural responses from computer simulations of acoustic spaces

Auditorium designs can be evaluated prior to construction by numerical modeling of the design. High-accuracy numerical modeling produces the sound pressure on a rectangular grid, and subjective assessment of the design requires auralization of the sampled sound field at a desired listener position. This paper investigates the production of binaural outputs from the sound pressure at a selected number of grid points by using a least squares beam forming approach. Low-frequency axisymmetric emulations are derived by assuming a solid sphere model of the head, and a spherical array of 640 microphones is used to emulate ten measured head-related transfer function (HRTF) data sets from the CIPIC database for half the audio bandwidth. The spherical array can produce high-accuracy band-limited emulation of any human subject's measured HRTFs for a fixed listener position by using individual sets of beam forming impulse responses.     

ACTIVE CONTROL OF ENVIRONMENTAL NOISE, VII: PERFORMANCE OF MULTI-FREQUENCY, MULTI-CHANNEL, FREEFIELD SOUND CANCELLING SYSTEMS

The paper considers the performance of multi-frequency, multi-channel, free field sound cancelling systems for the reduction of discrete frequency and periodic noise. The approach uses a method of directional active noise control. Large sound reductions from these systems have been made possible through: (a) synthetically generating the cancelling sound and synchronizing with the primary source; (b) automatic alignment of all stability regions of the control system and; (c) avoiding instability produced by these multi-channel systems.     

Analytic and numeric Green’s functions for a two-dimensional electron gas in an orthogonal magnetic field

We have derived closed analytic expressions for the Green’s function of an electron in a two-dimensional electron gas threaded by a uniform perpendicular magnetic field, also in the presence of a uniform electric field and of a parabolic spatial confinement. A workable and powerful numerical procedure for the calculation of the Green’s functions for a large infinitely extended quantum wire is considered exploiting a lattice model for the wire, the tight-binding representation for the corresponding matrix Green’s function, and the Peierls phase factor in the Hamiltonian hopping matrix element to account for the magnetic field. The numerical evaluation of the Green’s function has been performed by means of the decimation-renormalization method, and quite satisfactorily compared with the analytic results worked out in this paper. As an example of the versatility of the numerical and analytic tools here presented, the peculiar semilocal character of the magnetic Green’s function is studied in detail because of its basic importance in determining magneto-transport properties in mesoscopic systems.     

Experiments on the active control of transformer noise

This paper describes an experiment which demonstrates how a useful degree of active noise control can be achieved with ordinary sound amplification and reproduction equipment. A loudspeaker positioned next to a large pair of noisy electricity transformers was made to mimic their noise but in antiphase. The aim of the study was to investigate the degree to which the “antisound” would cancel a disturbing noise heard in a nearby office. Some 20 decibels of control was achieved very easily for the 100 Hz component of the noise but the higher frequency sound could only be controlled in localized patches. The experiment suggests that sounds of discrete frequencies of less than 100 Hz are relatively easily controlled with unsophisticated audio equipment, but that useful control of higher frequency elements is much more difficult.     

Optimal secondary source position in exterior spherical acoustical holophony

Exterior spherical acoustical holophony is a branch of spatial audio reproduction that deals with the rendering of a given free-field radiation pattern (the primary field) by using a compact spherical loudspeaker array (the secondary source). More precisely, the primary field is known on a spherical surface surrounding the primary and secondary sources and, since the acoustic fields are described in spherical coordinates, they are naturally subjected to spherical harmonic analysis. Besides, the inverse problem of deriving optimal driving signals from a known primary field is ill-posed because the secondary source cannot radiate high-order spherical harmonics efficiently, especially in the low-frequency range. As a consequence, a standard least-squares solution will overload the transducers if the primary field contains such harmonics. Here, this is avoided by discarding the strongly decaying spherical waves, which are identified through inspection of the radiation efficiency curves of the secondary source. However, such an unavoidable regularization procedure increases the least-squares error, which also depends on the position of the secondary source.This paper deals with the above-mentioned questions in the context of far-field directivity reproduction at low and medium frequencies. In particular, an optimal secondary source position is sought, which leads to the lowest reproduction error in the least-squares sense without overloading the transducers. In order to address this issue, a regularization quality factor is introduced to evaluate the amount of regularization required. It is shown that the optimal position improves significantly the holophonic reconstruction and maximizes the regularization quality factor (minimizes the amount of regularization), which is the main general contribution of this paper. Therefore, this factor can also be used as a cost function to obtain the optimal secondary source position.