Sound reproduction systems using variable-directivity loudspeakers

Sound reproduction systems using omnidirectional loudspeakers produce reflections from room surfaces which interfere with the desired sound field within the array. While active compensation systems can reduce the reverberant level, they require calibration in each room and are processor-intensive. Directional loudspeakers allow the direct to reverberant level to be improved within the array, but still produce a finite exterior field which reflects from the room surfaces. The use of variable-directivity loudspeakers allows the exterior field to be eliminated at low frequencies by implementing the Kirchhoff-Helmholtz integral equation. This paper investigates the performance of variable-directivity arrays in reducing reverberant levels and compares the results with those derived in a previous paper for fixed-directivity arrays. The results presented may have some impact on the design of commercial multi-channel systems for sound reproduction.     

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.     

Interior and exterior sound field control using general two-dimensional first-order sources

Reproduction of a given sound field interior to a circular loudspeaker array without producing an undesirable exterior sound field is an unsolved problem over a broadband of frequencies. At low frequencies, by implementing the Kirchhoff-Helmholtz integral using a circular discrete array of line-source loudspeakers, a sound field can be recreated within the array and produce no exterior sound field, provided that the loudspeakers have azimuthal polar responses with variable first-order responses which are a combination of a two-dimensional (2D) monopole and a radially oriented 2D dipole. This paper examines the performance of circular discrete arrays of line-source loudspeakers which also include a tangential dipole, providing general variable-directivity responses in azimuth. It is shown that at low frequencies, the tangential dipoles are not required, but that near and above the Nyquist frequency, the tangential dipoles can both improve the interior accuracy and reduce the exterior sound field. The additional dipoles extend the useful range of the array by around an octave.     

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.     

Directional perception of distributed sound sources

The perception of spatially distributed sound sources was investigated by conducting two listening experiments in anechoic conditions with 13 loudspeakers evenly distributed in the frontal horizontal plane emitting incoherent noise signals. In the first experiment, widely distributed sound sources with gaps in their distribution emitted pink noise. The results indicated that the exact loudspeaker distribution could not be perceived accurately and that the width of the distribution was perceived to be narrower than it was in reality. Up to three spatially distributed loudspeakers that were simultaneously emitting sound could be individually perceived. In addition, the number of loudspeakers that were indicated as emitting sound was smaller than the actual number. In the second experiment, a reference with 13 loudspeakers and test cases with fewer loudspeakers were presented and their perceived spatial difference was rated. The effect of the noise bandwidth was of particular interest. Noise with different bandwidths centered around 500 and 4000 Hz was used. The results indicated that when the number of loudspeakers was increased from four to seven, the perceived auditory event was very similar to that perceived with 13 loudspeakers at all bandwidths. The perceived differences were larger in wideband noise than in narrow-band noise.     

改进的5.1通路环绕声三扬声器虚拟重发系统

提出一种改进的5．1通路环绕声的三扬声器虚拟重发系统。本文首先指出,现有的两扬声器虚拟重发存在听音区域窄、声像稳定性差等缺陷。通过理论分析和心理声实验证明,采用一对布置在前方±30°的全频带扬声器和一个前方0°的高频中置扬声器组成5．1通路环绕声虚拟重发系统,不但可提高重发声像的稳定性,且扬声器布置实际可行,信号处理相对简单。因此所提出的虚拟系统及其扬声器布置是适合于电视和多媒体计算机的应用。     

虚拟声技术及其应用(上)

本文综述了虚拟声技术及其应用。在简单回顾了有关虚拟声的心理声学基础后，论述和分析了耳机和扬声器重发虚拟声的基本原理和基本的缺陷，总结了近年所提出的虚拟声的改进方法，最后探讨了虚拟声在科学研究、工程技术和消费电子领域的各种应用。     

Sound localization of stereo reproduction with parametric loudspeakers

A parametric loudspeaker radiates an audible signal by the interaction of the primary wave that is amplitude modulated and is known as a super-directivity loudspeaker. The parametric loudspeaker is one of the prominent applications of nonlinear acoustics. So far, the applications have been limited monaural reproduction sound system for public address in museum, station, street etc. In this paper, we investigated sound localization of stereo reproduction using two parametric loudspeakers in comparison with that using two ordinary dynamic loudspeakers. In subjective tests, the binaural information ILD (Interaural Level Difference) or ITD (Interaural Time Delay) was focused on. To investigate the characteristics of sound localization in a wide listening area, three typical listening positions were picked up. Signals were 500 Hz, 1 kHz, 2 kHz and 4 kHz pure tones and pink noise. The used parametric loudspeaker was an equilateral hexagon. The subjective test led to the results that when the parametric loudspeakers were used, the listeners at the three typical listening positions perceived the correct sound localization of not only pure tone but also pink noise and when the ordinary dynamic loudspeakers were used, except for the case of pure tone with ITD, the tendency was almost similar to those using the parametric loudspeakers. The second subjective tests were conducted in order to investigate in details the difference between parametric loudspeakers and ordinary dynamic loudspeakers by increasing the number of subjects. In the case of ITD and 500 Hz using the ordinary dynamic loudspeakers, three types of sound localization were categorized, in which the reversed type was major and the normal and the other types were minor. The ILDs which were measured with a dummy head and were calculated with several formulas were almost the same and indicated the reasons of the reversed typed sound localization and a serious influence of the crosstalk. It was found that in the case of pure tone with ITD, the contradiction between the binaural information ILD and ITD is remarkable, because the directivity of the ordinary dynamic loudspeakers was so dull that the crosstalk components had a serious influence on sound localization. It was determined the parametric loudspeaker could transmit correct binaural information to the listener, because the directivity of the parametric loudspeakers was so sharp that it suppressed the cross talk components.     

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。      

对5通路3/2环绕声系统缺陷的分析

本文利用环绕声重发的声像定位理论，对一种受到重视、很可能将被国际上广泛应用的通用环绕声系统—5通路3／2系统的声像特性进行了分析，并对系统进行了声像定位实验．理论和实验结果表明，按目前的信号馈给方式，系统可在倾听者前方产生稳定而明晰的声像；但在侧向和后方只能产生模糊和不稳定的声像（扬声器方向除外）；将别是在侧向，存在着一个声像的死区，并且对侧向和后方声像来说，听音区域片不宽阔，因而作为一种通用的系统，虽然将其用作伴随图像的重发时，可获得较好的效果；但用作不伴随图像的重发时，系统并不能很好地再观原声场的空间感。由于系统存在这种不容忽视的缺陷，文中着重强调应设法在这方面对系统加以改进     

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。     

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.     

The characteristics of acoustic line array prototypes for scale model experiments

The sound fields created by three different settings of a 5 m long linear loudspeaker array were studied experimentally in the present investigation in an attempt to find out a method to produce a reasonable sound field for scaled down model experiments. The first setting was with all the loudspeakers facing vertically upward, while the second setting was with all the loudspeakers facing the horizontal. The last one resembled the first setting but the loudspeakers were put inside a small partial enclosure whose opening was facing vertically upward. Results show that the last setting can produce a sound field with characteristics close to those of a two-dimensional sound field. The second setting results in a sound directivity pattern relevant to that of the measured A-weighted traffic noise.     

A synthesis approach for reproducing the response of aircraft panels to a turbulent boundary layer excitation

Random wall-pressure fluctuations due to the turbulent boundary layer (TBL) are a feature of the air flow over an aircraft fuselage under cruise conditions, creating undesirable effects such as cabin noise annoyance. In order to test potential solutions to reduce the TBL-induced noise, a cost-efficient alternative to in-flight or wind-tunnel measurements involves the laboratory simulation of the response of aircraft sidewalls to high-speed subsonic TBL excitation. Previously published work has shown that TBL simulation using a near-field array of loudspeakers is only feasible in the low frequency range due to the rapid decay of the spanwise correlation length with frequency. This paper demonstrates through theoretical criteria how the wavenumber filtering capabilities of the radiating panel reduces the number of sources required, thus dramatically enlarging the frequency range over which the response of the TBL-excited panel is accurately reproduced. Experimental synthesis of the panel response to high-speed TBL excitation is found to be feasible over the hydrodynamic coincidence frequency range using a reduced set of near-field loudspeakers driven by optimal signals. Effective methodologies are proposed for an accurate reproduction of the TBL-induced sound power radiated by the panel into a free-field and when coupled to a cavity.     

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.     

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。     

A self-silenced sound beam

Parametric loudspeakers are transmitting two high power ultrasound frequencies. During propagation through the air, nonlinear interaction creates a narrow sound beam at the difference frequency, similar to a light beam from a torch. In this work is added the physical phenomenon of propagation cancellation, leaving a limited region within which the sound can be heard—a 1 meter long cylinder with diameter 8 cm. It is equivalent to a torch which would only illuminate objects within 1 meter. The concept is demonstrated both in simulation and in experiment.     

多通路空间声的前方四扬声器局域Ambisonics信号馈给法*

典型的多通路空间声扬声器布置一般包含水平面左前、右前,高仰角左前上、右前上四个方向的扬声器。 本文提出一种利用该四个扬声器产生前方水平与垂直方向虚拟源的一阶局域Ambisonics 信号馈给法。该信号馈给法是通过对目标和重放声场进行球谐函数展开并取一阶近似得到。采用简化的头部模型和精确的头相关传输函数模型分析表明,一阶局域Ambisonics 信号馈给法可以产生合适的低频听觉定位因素,包括双耳时间差及其随头部转动的动态变化。虚拟源定位实验结果表明,该方法可以在扬声器布置的范围内,甚至在略超出扬声器布置的范围内产生不同方位角和仰角的虚拟源。因而本文的方法可用在多通路空间声重放中产生与图像配合的虚拟源定位效果。     

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.     

Experimental implementation of a low-frequency global sound equalization method based on free field propagation

An experimental implementation of a global sound equalization method in a rectangular room using active control is described in this paper. The main purpose of the work has been to provide experimental evidence that sound can be equalized in a continuous three-dimensional region, the listening zone, which occupies a considerable part of the complete volume of the room. The equalization method, based on the simulation of a progressive plane wave, was implemented in a room with inner dimensions of 2.70 m × 2.74 m × 2.40 m. With this method, the sound was reproduced by a matrix of 4 × 5 loudspeakers in one of the walls. After traveling through the room, the sound wave was absorbed on the opposite wall, which had a similar arrangement of loudspeakers, by means of active control. A set of 40 digital FIR filters was used to modify the original input signal before it was fed to the loudspeakers, one filter for each transducer. The optimal arrangement of the loudspeakers and the maximum frequency that can be equalized is analyzed theoretically in this paper. The presented experimental results show that sound equalization was possible from 10 Hz to approximately 425 Hz in the listening zone. A flat frequency response with deviations within ±5 decibels from the desired value was achieved. A higher demanding performance with deviations within ±1.5 decibels from a flat frequency response was attained in the interval between 20 Hz and 280 Hz. At the same time, the impulse response was quite well approximated to a delayed delta function in the listening zone. Examples of the spatial distribution of the sound field are also shown.